draft-ietf-rtcweb-jsep-26.txt   rfc8829.txt 
Network Working Group J. Uberti Internet Engineering Task Force (IETF) J. Uberti
Internet-Draft Google Request for Comments: 8829 Google
Intended status: Standards Track C. Jennings Category: Standards Track C. Jennings
Expires: August 31, 2019 Cisco ISSN: 2070-1721 Cisco
E. Rescorla, Ed. E. Rescorla, Ed.
Mozilla Mozilla
February 27, 2019 January 2021
JavaScript Session Establishment Protocol JavaScript Session Establishment Protocol (JSEP)
draft-ietf-rtcweb-jsep-26
Abstract Abstract
This document describes the mechanisms for allowing a JavaScript This document describes the mechanisms for allowing a JavaScript
application to control the signaling plane of a multimedia session application to control the signaling plane of a multimedia session
via the interface specified in the W3C RTCPeerConnection API, and via the interface specified in the W3C RTCPeerConnection API and
discusses how this relates to existing signaling protocols. discusses how this relates to existing signaling protocols.
Status of This Memo Status of This Memo
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Table of Contents Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 4 1. Introduction
1.1. General Design of JSEP . . . . . . . . . . . . . . . . . 4 1.1. General Design of JSEP
1.2. Other Approaches Considered . . . . . . . . . . . . . . . 6 1.2. Other Approaches Considered
2. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 6 1.3. Contradiction regarding bundle-only "m=" sections
3. Semantics and Syntax . . . . . . . . . . . . . . . . . . . . 7 2. Terminology
3.1. Signaling Model . . . . . . . . . . . . . . . . . . . . . 7 3. Semantics and Syntax
3.2. Session Descriptions and State Machine . . . . . . . . . 7 3.1. Signaling Model
3.3. Session Description Format . . . . . . . . . . . . . . . 11 3.2. Session Descriptions and State Machine
3.4. Session Description Control . . . . . . . . . . . . . . . 11 3.3. Session Description Format
3.4.1. RtpTransceivers . . . . . . . . . . . . . . . . . . . 11 3.4. Session Description Control
3.4.2. RtpSenders . . . . . . . . . . . . . . . . . . . . . 12 3.4.1. RtpTransceivers
3.4.3. RtpReceivers . . . . . . . . . . . . . . . . . . . . 12 3.4.2. RtpSenders
3.5. ICE . . . . . . . . . . . . . . . . . . . . . . . . . . . 12 3.4.3. RtpReceivers
3.5.1. ICE Gathering Overview . . . . . . . . . . . . . . . 12 3.5. ICE
3.5.2. ICE Candidate Trickling . . . . . . . . . . . . . . . 13 3.5.1. ICE Gathering Overview
3.5.2.1. ICE Candidate Format . . . . . . . . . . . . . . 13 3.5.2. ICE Candidate Trickling
3.5.3. ICE Candidate Policy . . . . . . . . . . . . . . . . 14 3.5.2.1. ICE Candidate Format
3.5.4. ICE Candidate Pool . . . . . . . . . . . . . . . . . 15 3.5.3. ICE Candidate Policy
3.5.5. ICE Versions . . . . . . . . . . . . . . . . . . . . 16 3.5.4. ICE Candidate Pool
3.6. Video Size Negotiation . . . . . . . . . . . . . . . . . 16 3.5.5. ICE Versions
3.6.1. Creating an imageattr Attribute . . . . . . . . . . . 16 3.6. Video Size Negotiation
3.6.2. Interpreting imageattr Attributes . . . . . . . . . . 17 3.6.1. Creating an imageattr Attribute
3.7. Simulcast . . . . . . . . . . . . . . . . . . . . . . . . 19 3.6.2. Interpreting imageattr Attributes
3.8. Interactions With Forking . . . . . . . . . . . . . . . . 20 3.7. Simulcast
3.8.1. Sequential Forking . . . . . . . . . . . . . . . . . 20 3.8. Interactions with Forking
3.8.2. Parallel Forking . . . . . . . . . . . . . . . . . . 21 3.8.1. Sequential Forking
4. Interface . . . . . . . . . . . . . . . . . . . . . . . . . . 22 3.8.2. Parallel Forking
4.1. PeerConnection . . . . . . . . . . . . . . . . . . . . . 22 4. Interface
4.1.1. Constructor . . . . . . . . . . . . . . . . . . . . . 22 4.1. PeerConnection
4.1.2. addTrack . . . . . . . . . . . . . . . . . . . . . . 24 4.1.1. Constructor
4.1.3. removeTrack . . . . . . . . . . . . . . . . . . . . . 24 4.1.2. addTrack
4.1.4. addTransceiver . . . . . . . . . . . . . . . . . . . 25 4.1.3. removeTrack
4.1.5. createDataChannel . . . . . . . . . . . . . . . . . . 25 4.1.4. addTransceiver
4.1.6. createOffer . . . . . . . . . . . . . . . . . . . . . 25 4.1.5. onaddtrack Event
4.1.7. createAnswer . . . . . . . . . . . . . . . . . . . . 26 4.1.6. createDataChannel
4.1.8. SessionDescriptionType . . . . . . . . . . . . . . . 27 4.1.7. ondatachannel Event
4.1.8.1. Use of Provisional Answers . . . . . . . . . . . 28 4.1.8. createOffer
4.1.8.2. Rollback . . . . . . . . . . . . . . . . . . . . 28 4.1.9. createAnswer
4.1.9. setLocalDescription . . . . . . . . . . . . . . . . . 29 4.1.10. SessionDescriptionType
4.1.10. setRemoteDescription . . . . . . . . . . . . . . . . 30 4.1.10.1. Use of Provisional Answers
4.1.11. currentLocalDescription . . . . . . . . . . . . . . . 30 4.1.10.2. Rollback
4.1.12. pendingLocalDescription . . . . . . . . . . . . . . . 30 4.1.11. setLocalDescription
4.1.13. currentRemoteDescription . . . . . . . . . . . . . . 30 4.1.12. setRemoteDescription
4.1.14. pendingRemoteDescription . . . . . . . . . . . . . . 31 4.1.13. currentLocalDescription
4.1.15. canTrickleIceCandidates . . . . . . . . . . . . . . . 31 4.1.14. pendingLocalDescription
4.1.16. setConfiguration . . . . . . . . . . . . . . . . . . 31 4.1.15. currentRemoteDescription
4.1.17. addIceCandidate . . . . . . . . . . . . . . . . . . . 32 4.1.16. pendingRemoteDescription
4.2. RtpTransceiver . . . . . . . . . . . . . . . . . . . . . 33 4.1.17. canTrickleIceCandidates
4.2.1. stop . . . . . . . . . . . . . . . . . . . . . . . . 33 4.1.18. setConfiguration
4.2.2. stopped . . . . . . . . . . . . . . . . . . . . . . . 33 4.1.19. addIceCandidate
4.2.3. setDirection . . . . . . . . . . . . . . . . . . . . 33 4.1.20. onicecandidate Event
4.2.4. direction . . . . . . . . . . . . . . . . . . . . . . 34 4.2. RtpTransceiver
4.2.5. currentDirection . . . . . . . . . . . . . . . . . . 34 4.2.1. stop
4.2.6. setCodecPreferences . . . . . . . . . . . . . . . . . 34 4.2.2. stopped
5. SDP Interaction Procedures . . . . . . . . . . . . . . . . . 35 4.2.3. setDirection
5.1. Requirements Overview . . . . . . . . . . . . . . . . . . 35 4.2.4. direction
5.1.1. Usage Requirements . . . . . . . . . . . . . . . . . 35 4.2.5. currentDirection
5.1.2. Profile Names and Interoperability . . . . . . . . . 35 4.2.6. setCodecPreferences
5.2. Constructing an Offer . . . . . . . . . . . . . . . . . . 37 5. SDP Interaction Procedures
5.2.1. Initial Offers . . . . . . . . . . . . . . . . . . . 37 5.1. Requirements Overview
5.2.2. Subsequent Offers . . . . . . . . . . . . . . . . . . 43 5.1.1. Usage Requirements
5.2.3. Options Handling . . . . . . . . . . . . . . . . . . 47 5.1.2. Profile Names and Interoperability
5.2.3.1. IceRestart . . . . . . . . . . . . . . . . . . . 47 5.2. Constructing an Offer
5.2.3.2. VoiceActivityDetection . . . . . . . . . . . . . 47 5.2.1. Initial Offers
5.3. Generating an Answer . . . . . . . . . . . . . . . . . . 48 5.2.2. Subsequent Offers
5.3.1. Initial Answers . . . . . . . . . . . . . . . . . . . 48 5.2.3. Options Handling
5.3.2. Subsequent Answers . . . . . . . . . . . . . . . . . 55 5.2.3.1. IceRestart
5.3.3. Options Handling . . . . . . . . . . . . . . . . . . 56 5.2.3.2. VoiceActivityDetection
5.3.3.1. VoiceActivityDetection . . . . . . . . . . . . . 56 5.3. Generating an Answer
5.4. Modifying an Offer or Answer . . . . . . . . . . . . . . 57 5.3.1. Initial Answers
5.5. Processing a Local Description . . . . . . . . . . . . . 57 5.3.2. Subsequent Answers
5.6. Processing a Remote Description . . . . . . . . . . . . . 58 5.3.3. Options Handling
5.7. Processing a Rollback . . . . . . . . . . . . . . . . . . 58 5.3.3.1. VoiceActivityDetection
5.8. Parsing a Session Description . . . . . . . . . . . . . . 59 5.4. Modifying an Offer or Answer
5.8.1. Session-Level Parsing . . . . . . . . . . . . . . . . 60 5.5. Processing a Local Description
5.8.2. Media Section Parsing . . . . . . . . . . . . . . . . 61 5.6. Processing a Remote Description
5.8.3. Semantics Verification . . . . . . . . . . . . . . . 64 5.7. Processing a Rollback
5.9. Applying a Local Description . . . . . . . . . . . . . . 65 5.8. Parsing a Session Description
5.10. Applying a Remote Description . . . . . . . . . . . . . . 67 5.8.1. Session-Level Parsing
5.11. Applying an Answer . . . . . . . . . . . . . . . . . . . 71 5.8.2. Media Section Parsing
6. Processing RTP/RTCP . . . . . . . . . . . . . . . . . . . . . 74 5.8.3. Semantics Verification
7. Examples . . . . . . . . . . . . . . . . . . . . . . . . . . 74 5.9. Applying a Local Description
7.1. Simple Example . . . . . . . . . . . . . . . . . . . . . 74 5.10. Applying a Remote Description
7.2. Detailed Example . . . . . . . . . . . . . . . . . . . . 78 5.11. Applying an Answer
7.3. Early Transport Warmup Example . . . . . . . . . . . . . 88 6. Processing RTP/RTCP
8. Security Considerations . . . . . . . . . . . . . . . . . . . 95 7. Examples
9. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 96 7.1. Simple Example
10. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 96 7.2. Detailed Example
11. References . . . . . . . . . . . . . . . . . . . . . . . . . 96 7.3. Early Transport Warmup Example
11.1. Normative References . . . . . . . . . . . . . . . . . . 96 8. Security Considerations
11.2. Informative References . . . . . . . . . . . . . . . . . 100 9. IANA Considerations
10. References
Appendix A. Appendix A . . . . . . . . . . . . . . . . . . . . . 103 10.1. Normative References
Appendix B. Change log . . . . . . . . . . . . . . . . . . . . . 105 10.2. Informative References
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 115 Appendix A. SDP ABNF Syntax
Acknowledgements
Authors' Addresses
1. Introduction 1. Introduction
This document describes how the W3C WEBRTC RTCPeerConnection This document describes how the W3C Web Real-Time Communication
interface [W3C.webrtc] is used to control the setup, management and (WebRTC) RTCPeerConnection interface [W3C.webrtc] is used to control
teardown of a multimedia session. the setup, management, and teardown of a multimedia session.
1.1. General Design of JSEP 1.1. General Design of JSEP
WebRTC call setup has been designed to focus on controlling the media WebRTC call setup has been designed to focus on controlling the media
plane, leaving signaling plane behavior up to the application as much plane, leaving signaling-plane behavior up to the application as much
as possible. The rationale is that different applications may prefer as possible. The rationale is that different applications may prefer
to use different protocols, such as the existing SIP call signaling to use different protocols, such as the existing SIP call signaling
protocol, or something custom to the particular application, perhaps protocol, or something custom to the particular application, perhaps
for a novel use case. In this approach, the key information that for a novel use case. In this approach, the key information that
needs to be exchanged is the multimedia session description, which needs to be exchanged is the multimedia session description, which
specifies the necessary transport and media configuration information specifies the transport and media configuration information necessary
necessary to establish the media plane. to establish the media plane.
With these considerations in mind, this document describes the With these considerations in mind, this document describes the
JavaScript Session Establishment Protocol (JSEP) that allows for full JavaScript Session Establishment Protocol (JSEP), which allows for
control of the signaling state machine from JavaScript. As described full control of the signaling state machine from JavaScript. As
above, JSEP assumes a model in which a JavaScript application described above, JSEP assumes a model in which a JavaScript
executes inside a runtime containing WebRTC APIs (the "JSEP application executes inside a runtime containing WebRTC APIs (the
implementation"). The JSEP implementation is almost entirely "JSEP implementation"). The JSEP implementation is almost entirely
divorced from the core signaling flow, which is instead handled by divorced from the core signaling flow, which is instead handled by
the JavaScript making use of two interfaces: (1) passing in local and the JavaScript making use of two interfaces: (1) passing in local and
remote session descriptions and (2) interacting with the ICE state remote session descriptions and (2) interacting with the Interactive
machine. The combination of the JSEP implementation and the Connectivity Establishment (ICE) state machine [RFC8445]. The
JavaScript application is referred to throughout this document as a combination of the JSEP implementation and the JavaScript application
"JSEP endpoint". is referred to throughout this document as a "JSEP endpoint".
In this document, the use of JSEP is described as if it always occurs In this document, the use of JSEP is described as if it always occurs
between two JSEP endpoints. Note though in many cases it will between two JSEP endpoints. Note, though, that in many cases it will
actually be between a JSEP endpoint and some kind of server, such as actually be between a JSEP endpoint and some kind of server, such as
a gateway or MCU. This distinction is invisible to the JSEP a gateway or Multipoint Control Unit (MCU). This distinction is
endpoint; it just follows the instructions it is given via the API. invisible to the JSEP endpoint; it just follows the instructions it
is given via the API.
JSEP's handling of session descriptions is simple and JSEP's handling of session descriptions is simple and
straightforward. Whenever an offer/answer exchange is needed, the straightforward. Whenever an offer/answer exchange is needed, the
initiating side creates an offer by calling a createOffer() API. The initiating side creates an offer by calling a createOffer API. The
application then uses that offer to set up its local config via the application then uses that offer to set up its local configuration
setLocalDescription() API. The offer is finally sent off to the via the setLocalDescription API. The offer is finally sent off to
remote side over its preferred signaling mechanism (e.g., the remote side over its preferred signaling mechanism (e.g.,
WebSockets); upon receipt of that offer, the remote party installs it WebSockets); upon receipt of that offer, the remote party installs it
using the setRemoteDescription() API. using the setRemoteDescription API.
To complete the offer/answer exchange, the remote party uses the To complete the offer/answer exchange, the remote party uses the
createAnswer() API to generate an appropriate answer, applies it createAnswer API to generate an appropriate answer, applies it using
using the setLocalDescription() API, and sends the answer back to the the setLocalDescription API, and sends the answer back to the
initiator over the signaling channel. When the initiator gets that initiator over the signaling channel. When the initiator gets that
answer, it installs it using the setRemoteDescription() API, and answer, it installs it using the setRemoteDescription API, and
initial setup is complete. This process can be repeated for initial setup is complete. This process can be repeated for
additional offer/answer exchanges. additional offer/answer exchanges.
Regarding ICE [RFC8445], JSEP decouples the ICE state machine from Regarding ICE [RFC8445], JSEP decouples the ICE state machine from
the overall signaling state machine, as the ICE state machine must the overall signaling state machine. The ICE state machine must
remain in the JSEP implementation, because only the implementation remain in the JSEP implementation because only the implementation has
has the necessary knowledge of candidates and other transport the necessary knowledge of candidates and other transport
information. Performing this separation provides additional information. Performing this separation provides additional
flexibility in protocols that decouple session descriptions from flexibility in protocols that decouple session descriptions from
transport. For instance, in traditional SIP, each offer or answer is transport. For instance, in traditional SIP, each offer or answer is
self-contained, including both the session descriptions and the self-contained, including both the session descriptions and the
transport information. However, [I-D.ietf-mmusic-trickle-ice-sip] transport information. However, [RFC8840] allows SIP to be used with
allows SIP to be used with trickle ICE [I-D.ietf-ice-trickle], in Trickle ICE [RFC8838], in which the session description can be sent
which the session description can be sent immediately and the immediately and the transport information can be sent when available.
transport information can be sent when available. Sending transport Sending transport information separately can allow for faster ICE and
information separately can allow for faster ICE and DTLS startup, DTLS startup, since ICE checks can start as soon as any transport
since ICE checks can start as soon as any transport information is information is available rather than waiting for all of it. JSEP's
available rather than waiting for all of it. JSEP's decoupling of decoupling of the ICE and signaling state machines allows it to
the ICE and signaling state machines allows it to accommodate either accommodate either model.
model.
Through its abstraction of signaling, the JSEP approach does require Although it abstracts signaling, the JSEP approach requires that the
the application to be aware of the signaling process. While the application be aware of the signaling process. While the application
application does not need to understand the contents of session does not need to understand the contents of session descriptions to
descriptions to set up a call, the application must call the right set up a call, the application must call the right APIs at the right
APIs at the right times, convert the session descriptions and ICE times, convert the session descriptions and ICE information into the
information into the defined messages of its chosen signaling defined messages of its chosen signaling protocol, and perform the
protocol, and perform the reverse conversion on the messages it reverse conversion on the messages it receives from the other side.
receives from the other side.
One way to make life easier for the application is to provide a One way to make life easier for the application is to provide a
JavaScript library that hides this complexity from the developer; JavaScript library that hides this complexity from the developer;
said library would implement a given signaling protocol along with said library would implement a given signaling protocol along with
its state machine and serialization code, presenting a higher level its state machine and serialization code, presenting a higher-level
call-oriented interface to the application developer. For example, call-oriented interface to the application developer. For example,
libraries exist to adapt the JSEP API into an API suitable for a SIP libraries exist to provide implementations of the SIP [RFC3261] and
or XMPP. Thus, JSEP provides greater control for the experienced Extensible Messaging and Presence Protocol (XMPP) [RFC6120] signaling
developer without forcing any additional complexity on the novice protocols atop the JSEP API. Thus, JSEP provides greater control for
developer. the experienced developer without forcing any additional complexity
on the novice developer.
1.2. Other Approaches Considered 1.2. Other Approaches Considered
One approach that was considered instead of JSEP was to include a One approach that was considered instead of JSEP was to include a
lightweight signaling protocol. Instead of providing session lightweight signaling protocol. Instead of providing session
descriptions to the API, the API would produce and consume messages descriptions to the API, the API would produce and consume messages
from this protocol. While providing a more high-level API, this put from this protocol. While providing a more high-level API, this put
more control of signaling within the JSEP implementation, forcing it more control of signaling within the JSEP implementation, forcing it
to have to understand and handle concepts like signaling glare (see to have to understand and handle concepts like signaling glare (see
[RFC3264], Section 4). [RFC3264], Section 4).
A second approach that was considered but not chosen was to decouple A second approach that was considered but not chosen was to decouple
the management of the media control objects from session the management of the media control objects from session
descriptions, instead offering APIs that would control each component descriptions, instead offering APIs that would control each component
directly. This was rejected based on the argument that requiring directly. This was rejected based on the argument that requiring
exposure of this level of complexity to the application programmer exposure of this level of complexity to the application programmer
would not be beneficial; it would result in an API where even a would not be beneficial; it would (1) result in an API where even a
simple example would require a significant amount of code to simple example would require a significant amount of code to
orchestrate all the needed interactions, as well as creating a large orchestrate all the needed interactions and (2) create a large API
API surface that needed to be agreed upon and documented. In surface that would need to be agreed upon and documented. In
addition, these API points could be called in any order, resulting in addition, these API points could be called in any order, resulting in
a more complex set of interactions with the media subsystem than the a more complex set of interactions with the media subsystem than the
JSEP approach, which specifies how session descriptions are to be JSEP approach, which specifies how session descriptions are to be
evaluated and applied. evaluated and applied.
One variation on JSEP that was considered was to keep the basic One variation on JSEP that was considered was to keep the basic
session description-oriented API, but to move the mechanism for session-description-oriented API but to move the mechanism for
generating offers and answers out of the JSEP implementation. generating offers and answers out of the JSEP implementation.
Instead of providing createOffer/createAnswer methods within the Instead of providing createOffer/createAnswer methods within the
implementation, this approach would instead expose a getCapabilities implementation, this approach would instead expose a getCapabilities
API which would provide the application with the information it API, which would provide the application with the information it
needed in order to generate its own session descriptions. This needed in order to generate its own session descriptions. This
increases the amount of work that the application needs to do; it increases the amount of work that the application needs to do; it
needs to know how to generate session descriptions from capabilities, needs to know how to generate session descriptions from capabilities,
and especially how to generate the correct answer from an arbitrary and especially how to generate the correct answer from an arbitrary
offer and the supported capabilities. While this could certainly be offer and the supported capabilities. While this could certainly be
addressed by using a library like the one mentioned above, it addressed by using a library like the one mentioned above, it
basically forces the use of said library even for a simple example. basically forces the use of said library even for a simple example.
Providing createOffer/createAnswer avoids this problem. Providing createOffer/createAnswer avoids this problem.
1.3. Contradiction regarding bundle-only "m=" sections
Since the approval of the WebRTC specification documents, the IETF
has become aware of an inconsistency between the document specifying
JSEP and the document specifying BUNDLE (this RFC and [RFC8843],
respectively). Rather than delaying publication further to come to a
resolution, the documents are being published as they were originally
approved. The IETF intends to restart work on these technologies,
and revised versions of these documents will be published as soon as
a resolution becomes available.
The specific issue involves the handling of "m=" sections that are
designated as bundle-only, as discussed in Section 4.1.1 of this RFC.
Currently, there is divergence between JSEP and BUNDLE, as well as
between these specifications and existing browser implementations:
* JSEP prescribes that said "m=" sections should use port zero and
add an "a=bundle-only" attribute in initial offers, but not in
answers or subsequent offers.
* BUNDLE prescribes that these "m=" sections should be marked as
described in the previous point, but in all offers and answers.
* Most current browsers do not mark any "m=" sections with port zero
and instead use the same port for all bundled "m=" sections; some
others follow the JSEP behavior.
2. Terminology 2. Terminology
The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT", The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
"SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this "SHOULD", "SHOULD NOT", "RECOMMENDED", "NOT RECOMMENDED", "MAY", and
document are to be interpreted as described in [RFC2119]. "OPTIONAL" in this document are to be interpreted as described in
BCP 14 [RFC2119] [RFC8174] when, and only when, they appear in all
capitals, as shown here.
3. Semantics and Syntax 3. Semantics and Syntax
3.1. Signaling Model 3.1. Signaling Model
JSEP does not specify a particular signaling model or state machine, JSEP does not specify a particular signaling model or state machine,
other than the generic need to exchange session descriptions in the other than the generic need to exchange session descriptions in the
fashion described by [RFC3264] (offer/answer) in order for both sides fashion described by [RFC3264] (offer/answer) in order for both sides
of the session to know how to conduct the session. JSEP provides of the session to know how to conduct the session. JSEP provides
mechanisms to create offers and answers, as well as to apply them to mechanisms to create offers and answers, as well as to apply them to
a session. However, the JSEP implementation is totally decoupled a session. However, the JSEP implementation is totally decoupled
from the actual mechanism by which these offers and answers are from the actual mechanism by which these offers and answers are
communicated to the remote side, including addressing, communicated to the remote side, including addressing,
retransmission, forking, and glare handling. These issues are left retransmission, forking, and glare handling. These issues are left
entirely up to the application; the application has complete control entirely up to the application; the application has complete control
over which offers and answers get handed to the implementation, and over which offers and answers get handed to the implementation, and
when. when.
+-----------+ +-----------+ +-----------+ +-----------+
| Web App |<--- App-Specific Signaling -->| Web App | | Web App |<--- App-Specific Signaling -->| Web App |
+-----------+ +-----------+ +-----------+ +-----------+
^ ^ ^ ^
| SDP | SDP | SDP | SDP
V V V V
+-----------+ +-----------+ +-----------+ +-----------+
| JSEP |<----------- Media ------------>| JSEP | | JSEP |<----------- Media ------------>| JSEP |
| Impl. | | Impl. | | Impl. | | Impl. |
+-----------+ +-----------+ +-----------+ +-----------+
Figure 1: JSEP Signaling Model Figure 1: JSEP Signaling Model
3.2. Session Descriptions and State Machine 3.2. Session Descriptions and State Machine
In order to establish the media plane, the JSEP implementation needs In order to establish the media plane, the JSEP implementation needs
specific parameters to indicate what to transmit to the remote side, specific parameters to indicate what to transmit to the remote side,
as well as how to handle the media that is received. These as well as how to handle the media that is received. These
parameters are determined by the exchange of session descriptions in parameters are determined by the exchange of session descriptions in
offers and answers, and there are certain details to this process offers and answers, and there are certain details to this process
that must be handled in the JSEP APIs. that must be handled in the JSEP APIs.
Whether a session description applies to the local side or the remote Whether a session description applies to the local side or the remote
side affects the meaning of that description. For example, the list side affects the meaning of that description. For example, the list
of codecs sent to a remote party indicates what the local side is of codecs sent to a remote party indicates what the local side is
willing to receive, which, when intersected with the set of codecs willing to receive, which, when intersected with the set of codecs
the remote side supports, specifies what the remote side should send. the remote side supports, specifies what the remote side should send.
However, not all parameters follow this rule; some parameters are However, not all parameters follow this rule; some parameters are
declarative and the remote side MUST either accept them or reject declarative, and the remote side must either accept them or reject
them altogether. An example of such a parameter is the DTLS them altogether. An example of such a parameter is the TLS
fingerprints [RFC8122], which are calculated based on the local fingerprints [RFC8122] as used in the context of DTLS [RFC6347];
certificate(s) offered, and are not subject to negotiation. these fingerprints are calculated based on the local certificate(s)
offered and are not subject to negotiation.
In addition, various RFCs put different conditions on the format of In addition, various RFCs put different conditions on the format of
offers versus answers. For example, an offer may propose an offers versus answers. For example, an offer may propose an
arbitrary number of m= sections (i.e., media descriptions as arbitrary number of "m=" sections (i.e., media descriptions as
described in [RFC4566], Section 5.14), but an answer must contain the described in [RFC4566], Section 5.14), but an answer must contain the
exact same number as the offer. exact same number as the offer.
Lastly, while the exact media parameters are only known only after an Lastly, while the exact media parameters are known only after an
offer and an answer have been exchanged, the offerer may receive ICE offer and an answer have been exchanged, the offerer may receive ICE
checks, and possibly media (e.g., in the case of a re-offer after a checks, and possibly media (e.g., in the case of a re-offer after a
connection has been established) before it receives an answer. To connection has been established) before it receives an answer. To
properly process incoming media in this case, the offerer's media properly process incoming media in this case, the offerer's media
handler must be aware of the details of the offer before the answer handler must be aware of the details of the offer before the answer
arrives. arrives.
Therefore, in order to handle session descriptions properly, the JSEP Therefore, in order to handle session descriptions properly, the JSEP
implementation needs: implementation needs:
skipping to change at page 8, line 37 skipping to change at line 393
2. To know if a session description is an offer or an answer. 2. To know if a session description is an offer or an answer.
3. To allow the offer to be specified independently of the answer. 3. To allow the offer to be specified independently of the answer.
JSEP addresses this by adding both setLocalDescription and JSEP addresses this by adding both setLocalDescription and
setRemoteDescription methods and having session description objects setRemoteDescription methods and having session description objects
contain a type field indicating the type of session description being contain a type field indicating the type of session description being
supplied. This satisfies the requirements listed above for both the supplied. This satisfies the requirements listed above for both the
offerer, who first calls setLocalDescription(sdp [offer]) and then offerer, who first calls setLocalDescription(sdp [offer]) and then
later setRemoteDescription(sdp [answer]), as well as for the later setRemoteDescription(sdp [answer]), and the answerer, who first
answerer, who first calls setRemoteDescription(sdp [offer]) and then calls setRemoteDescription(sdp [offer]) and then later
later setLocalDescription(sdp [answer]). setLocalDescription(sdp [answer]).
During the offer/answer exchange, the outstanding offer is considered During the offer/answer exchange, the outstanding offer is considered
to be "pending" at the offerer and the answerer, as it may either be to be "pending" at the offerer and the answerer, as it may be either
accepted or rejected. If this is a re-offer, each side will also accepted or rejected. If this is a re-offer, each side will also
have "current" local and remote descriptions, which reflect the have "current" local and remote descriptions, which reflect the
result of the last offer/answer exchange. Sections Section 4.1.12, result of the last offer/answer exchange. Sections 4.1.14, 4.1.16,
Section 4.1.14, Section 4.1.11, and Section 4.1.13, provide more 4.1.13, and 4.1.15 provide more detail on pending and current
detail on pending and current descriptions. descriptions.
JSEP also allows for an answer to be treated as provisional by the JSEP also allows for an answer to be treated as provisional by the
application. Provisional answers provide a way for an answerer to application. Provisional answers provide a way for an answerer to
communicate initial session parameters back to the offerer, in order communicate initial session parameters back to the offerer, in order
to allow the session to begin, while allowing a final answer to be to allow the session to begin, while allowing a final answer to be
specified later. This concept of a final answer is important to the specified later. This concept of a final answer is important to the
offer/answer model; when such an answer is received, any extra offer/answer model; when such an answer is received, any extra
resources allocated by the caller can be released, now that the exact resources allocated by the caller can be released, now that the exact
session configuration is known. These "resources" can include things session configuration is known. These "resources" can include things
like extra ICE components, TURN candidates, or video decoders. like extra ICE components, Traversal Using Relays around NAT (TURN)
Provisional answers, on the other hand, do no such deallocation; as a candidates, or video decoders. Provisional answers, on the other
result, multiple dissimilar provisional answers, with their own codec hand, do no such deallocation; as a result, multiple dissimilar
choices, transport parameters, etc., can be received and applied provisional answers, with their own codec choices, transport
during call setup. Note that the final answer itself may be parameters, etc., can be received and applied during call setup.
different than any received provisional answers. Note that the final answer itself may be different than any received
provisional answers.
In [RFC3264], the constraint at the signaling level is that only one In [RFC3264], the constraint at the signaling level is that only one
offer can be outstanding for a given session, but at the media stack offer can be outstanding for a given session, but at the JSEP level,
level, a new offer can be generated at any point. For example, when a new offer can be generated at any point. For example, when using
using SIP for signaling, if one offer is sent, then cancelled using a SIP for signaling, if one offer is sent and is then canceled using a
SIP CANCEL, another offer can be generated even though no answer was SIP CANCEL, another offer can be generated even though no answer was
received for the first offer. To support this, the JSEP media layer received for the first offer. To support this, the JSEP media layer
can provide an offer via the createOffer() method whenever the can provide an offer via the createOffer method whenever the
JavaScript application needs one for the signaling. The answerer can JavaScript application needs one for the signaling. The answerer can
send back zero or more provisional answers, and finally end the send back zero or more provisional answers and then finally end the
offer-answer exchange by sending a final answer. The state machine offer/answer exchange by sending a final answer. The state machine
for this is as follows: for this is as follows:
setRemote(OFFER) setLocal(PRANSWER) setRemote(OFFER) setLocal(PRANSWER)
/-----\ /-----\ /-----\ /-----\
| | | | | | | |
v | v | v | v |
+---------------+ | +---------------+ | +---------------+ | +---------------+ |
| |----/ | |----/ | |----/ | |----/
| have- | setLocal(PRANSWER) | have- | | have- | setLocal(PRANSWER) | have- |
| remote-offer |------------------- >| local-pranswer| | remote-offer |------------------- >| local-pranswer|
skipping to change at page 10, line 43 skipping to change at line 471
| have- | setRemote(PRANSWER) |have- | | have- | setRemote(PRANSWER) |have- |
| local-offer |------------------- >|remote-pranswer| | local-offer |------------------- >|remote-pranswer|
| | | | | | | |
| |----\ | |----\ | |----\ | |----\
+---------------+ | +---------------+ | +---------------+ | +---------------+ |
^ | ^ | ^ | ^ |
| | | | | | | |
\-----/ \-----/ \-----/ \-----/
setLocal(OFFER) setRemote(PRANSWER) setLocal(OFFER) setRemote(PRANSWER)
Figure 2: JSEP State Machine Figure 2: JSEP State Machine
Aside from these state transitions there is no other difference Aside from these state transitions, there is no other difference
between the handling of provisional ("pranswer") and final ("answer") between the handling of provisional ("pranswer") and final ("answer")
answers. answers.
3.3. Session Description Format 3.3. Session Description Format
JSEP's session descriptions use SDP syntax for their internal JSEP's session descriptions use Session Description Protocol (SDP)
representation. While this format is not optimal for manipulation syntax for their internal representation. While this format is not
from JavaScript, it is widely accepted, and frequently updated with optimal for manipulation from JavaScript, it is widely accepted and
new features; any alternate encoding of session descriptions would is frequently updated with new features; any alternate encoding of
have to keep pace with the changes to SDP, at least until the time session descriptions would have to keep pace with the changes to SDP,
that this new encoding eclipsed SDP in popularity. at least until the time that this new encoding eclipsed SDP in
popularity.
However, to provide for future flexibility, the SDP syntax is However, to provide for future flexibility, the SDP syntax is
encapsulated within a SessionDescription object, which can be encapsulated within a SessionDescription object, which can be
constructed from SDP, and be serialized out to SDP. If future constructed from SDP and be serialized out to SDP. If future
specifications agree on a JSON format for session descriptions, we specifications agree on a JSON format for session descriptions, we
could easily enable this object to generate and consume that JSON. could easily enable this object to generate and consume that JSON.
As detailed below, most applications should be able to treat the As detailed below, most applications should be able to treat the
SessionDescriptions produced and consumed by these various API calls SessionDescriptions produced and consumed by these various API calls
as opaque blobs; that is, the application will not need to read or as opaque blobs; that is, the application will not need to read or
change them. change them.
3.4. Session Description Control 3.4. Session Description Control
In order to give the application control over various common session In order to give the application control over various common session
parameters, JSEP provides control surfaces which tell the JSEP parameters, JSEP provides control surfaces that tell the JSEP
implementation how to generate session descriptions. This avoids the implementation how to generate session descriptions. This avoids the
need for JavaScript to modify session descriptions in most cases. need for JavaScript to modify session descriptions in most cases.
Changes to these objects result in changes to the session Changes to these objects result in changes to the session
descriptions generated by subsequent createOffer/Answer calls. descriptions generated by subsequent createOffer/createAnswer calls.
3.4.1. RtpTransceivers 3.4.1. RtpTransceivers
RtpTransceivers allow the application to control the RTP media RtpTransceivers allow the application to control the RTP media
associated with one m= section. Each RtpTransceiver has an RtpSender associated with one "m=" section. Each RtpTransceiver has an
and an RtpReceiver, which an application can use to control the RtpSender and an RtpReceiver, which an application can use to control
sending and receiving of RTP media. The application may also modify the sending and receiving of RTP media. The application may also
the RtpTransceiver directly, for instance, by stopping it. modify the RtpTransceiver directly, for instance, by stopping it.
RtpTransceivers generally have a 1:1 mapping with m= sections, RtpTransceivers generally have a 1:1 mapping with "m=" sections,
although there may be more RtpTransceivers than m= sections when although there may be more RtpTransceivers than "m=" sections when
RtpTransceivers are created but not yet associated with a m= section, RtpTransceivers are created but not yet associated with an "m="
or if RtpTransceivers have been stopped and disassociated from m= section, or if RtpTransceivers have been stopped and disassociated
sections. An RtpTransceiver is said to be associated with an m= from "m=" sections. An RtpTransceiver is said to be associated with
section if its mid property is non-null; otherwise it is said to be an "m=" section if its media identification (mid) property is non-
disassociated. The associated m= section is determined using a null; otherwise, it is said to be disassociated. The associated "m="
mapping between transceivers and m= section indices, formed when section is determined using a mapping between transceivers and "m="
creating an offer or applying a remote offer. section indices, formed when creating an offer or applying a remote
offer.
An RtpTransceiver is never associated with more than one m= section, An RtpTransceiver is never associated with more than one "m="
and once a session description is applied, a m= section is always section, and once a session description is applied, an "m=" section
associated with exactly one RtpTransceiver. However, in certain is always associated with exactly one RtpTransceiver. However, in
cases where a m= section has been rejected, as discussed in certain cases where an "m=" section has been rejected, as discussed
Section 5.2.2 below, that m= section will be "recycled" and in Section 5.2.2 below, that "m=" section will be "recycled" and
associated with a new RtpTransceiver with a new mid value. associated with a new RtpTransceiver with a new MID value.
RtpTransceivers can be created explicitly by the application or RtpTransceivers can be created explicitly by the application or
implicitly by calling setRemoteDescription with an offer that adds implicitly by calling setRemoteDescription with an offer that adds
new m= sections. new "m=" sections.
3.4.2. RtpSenders 3.4.2. RtpSenders
RtpSenders allow the application to control how RTP media is sent. RtpSenders allow the application to control how RTP media is sent.
An RtpSender is conceptually responsible for the outgoing RTP An RtpSender is conceptually responsible for the outgoing RTP
stream(s) described by an m= section. This includes encoding the stream(s) described by an "m=" section. This includes encoding the
attached MediaStreamTrack, sending RTP media packets, and generating/ attached MediaStreamTrack, sending RTP media packets, and generating/
processing RTCP for the outgoing RTP streams(s). processing the RTP Control Protocol (RTCP) for the outgoing RTP
streams(s).
3.4.3. RtpReceivers 3.4.3. RtpReceivers
RtpReceivers allow the application to inspect how RTP media is RtpReceivers allow the application to inspect how RTP media is
received. An RtpReceiver is conceptually responsible for the received. An RtpReceiver is conceptually responsible for the
incoming RTP stream(s) described by an m= section. This includes incoming RTP stream(s) described by an "m=" section. This includes
processing received RTP media packets, decoding the incoming processing received RTP media packets, decoding the incoming
stream(s) to produce a remote MediaStreamTrack, and generating/ stream(s) to produce a remote MediaStreamTrack, and generating/
processing RTCP for the incoming RTP stream(s). processing RTCP for the incoming RTP stream(s).
3.5. ICE 3.5. ICE
3.5.1. ICE Gathering Overview 3.5.1. ICE Gathering Overview
JSEP gathers ICE candidates as needed by the application. Collection JSEP gathers ICE candidates as needed by the application. Collection
of ICE candidates is referred to as a gathering phase, and this is of ICE candidates is referred to as a gathering phase, and this is
triggered either by the addition of a new or recycled m= section to triggered either by the addition of a new or recycled "m=" section to
the local session description, or new ICE credentials in the the local session description or by new ICE credentials in the
description, indicating an ICE restart. Use of new ICE credentials description, indicating an ICE restart. Use of new ICE credentials
can be triggered explicitly by the application, or implicitly by the can be triggered explicitly by the application or implicitly by the
JSEP implementation in response to changes in the ICE configuration. JSEP implementation in response to changes in the ICE configuration.
When the ICE configuration changes in a way that requires a new When the ICE configuration changes in a way that requires a new
gathering phase, a 'needs-ice-restart' bit is set. When this bit is gathering phase, a 'needs-ice-restart' bit is set. When this bit is
set, calls to the createOffer API will generate new ICE credentials. set, calls to the createOffer API will generate new ICE credentials.
This bit is cleared by a call to the setLocalDescription API with new This bit is cleared by a call to the setLocalDescription API with new
ICE credentials from either an offer or an answer, i.e., from either ICE credentials from either an offer or an answer, i.e., from either
a local- or remote-initiated ICE restart. a locally or remotely initiated ICE restart.
When a new gathering phase starts, the ICE agent will notify the When a new gathering phase starts, the ICE agent will notify the
application that gathering is occurring through an event. Then, when application that gathering is occurring through a state change event.
each new ICE candidate becomes available, the ICE agent will supply Then, when each new ICE candidate becomes available, the ICE agent
it to the application via an additional event; these candidates will will supply it to the application via an onicecandidate event; these
also automatically be added to the current and/or pending local candidates will also automatically be added to the current and/or
session description. Finally, when all candidates have been pending local session description. Finally, when all candidates have
gathered, an event will be dispatched to signal that the gathering been gathered, a final onicecandidate event will be dispatched to
process is complete. signal that the gathering process is complete.
Note that gathering phases only gather the candidates needed by Note that gathering phases only gather the candidates needed by
new/recycled/restarting m= sections; other m= sections continue to new/recycled/restarting "m=" sections; other "m=" sections continue
use their existing candidates. Also, if an m= section is bundled to use their existing candidates. Also, if an "m=" section is
(either by a successful bundle negotiation or by being marked as bundled (either by a successful bundle negotiation or by being marked
bundle-only), then candidates will be gathered and exchanged for that as bundle-only), then candidates will be gathered and exchanged for
m= section if and only if its MID is a BUNDLE-tag, as described in that "m=" section if and only if its MID item is a BUNDLE-tag, as
[I-D.ietf-mmusic-sdp-bundle-negotiation]. described in [RFC8843].
3.5.2. ICE Candidate Trickling 3.5.2. ICE Candidate Trickling
Candidate trickling is a technique through which a caller may Candidate trickling is a technique through which a caller may
incrementally provide candidates to the callee after the initial incrementally provide candidates to the callee after the initial
offer has been dispatched; the semantics of "Trickle ICE" are defined offer has been dispatched; the semantics of "Trickle ICE" are defined
in [I-D.ietf-ice-trickle]. This process allows the callee to begin in [RFC8838]. This process allows the callee to begin acting upon
acting upon the call and setting up the ICE (and perhaps DTLS) the call and setting up the ICE (and perhaps DTLS) connections
connections immediately, without having to wait for the caller to immediately, without having to wait for the caller to gather all
gather all possible candidates. This results in faster media setup possible candidates. This results in faster media setup in cases
in cases where gathering is not performed prior to initiating the where gathering is not performed prior to initiating the call.
call.
JSEP supports optional candidate trickling by providing APIs, as JSEP supports optional candidate trickling by providing APIs, as
described above, that provide control and feedback on the ICE described above, that provide control and feedback on the ICE
candidate gathering process. Applications that support candidate candidate gathering process. Applications that support candidate
trickling can send the initial offer immediately and send individual trickling can send the initial offer immediately and send individual
candidates when they get the notified of a new candidate; candidates when they get notified of a new candidate; applications
applications that do not support this feature can simply wait for the that do not support this feature can simply wait for the indication
indication that gathering is complete, and then create and send their that gathering is complete, and then create and send their offer,
offer, with all the candidates, at this time. with all the candidates, at that time.
Upon receipt of trickled candidates, the receiving application will Upon receipt of trickled candidates, the receiving application will
supply them to its ICE agent. This triggers the ICE agent to start supply them to its ICE agent. This triggers the ICE agent to start
using the new remote candidates for connectivity checks. using the new remote candidates for connectivity checks.
3.5.2.1. ICE Candidate Format 3.5.2.1. ICE Candidate Format
In JSEP, ICE candidates are abstracted by an IceCandidate object, and In JSEP, ICE candidates are abstracted by an IceCandidate object, and
as with session descriptions, SDP syntax is used for the internal as with session descriptions, SDP syntax is used for the internal
representation. representation.
The candidate details are specified in an IceCandidate field, using The candidate details are specified in an IceCandidate field, using
the same SDP syntax as the "candidate-attribute" field defined in the same SDP syntax as the "candidate-attribute" field defined in
[I-D.ietf-mmusic-ice-sip-sdp], Section 4.1. Note that this field [RFC8839], Section 5.1. Note that this field does not contain an
does not contain an "a=" prefix, as indicated in the following "a=" prefix, as indicated in the following example:
example:
candidate:1 1 UDP 1694498815 192.0.2.33 10000 typ host candidate:1 1 UDP 1694498815 192.0.2.33 10000 typ host
The IceCandidate object contains a field to indicate which ICE ufrag The IceCandidate object contains a field to indicate which ICE
it is associated with, as defined in [I-D.ietf-mmusic-ice-sip-sdp], username fragment (ufrag) it is associated with, as defined in
Section 4.4. This value is used to determine which session [RFC8839], Section 5.4. This value is used to determine which
description (and thereby which gathering phase) this IceCandidate session description (and thereby which gathering phase) this
belongs to, which helps resolve ambiguities during ICE restarts. If IceCandidate belongs to, which helps resolve ambiguities during ICE
this field is absent in a received IceCandidate (perhaps when restarts. If this field is absent in a received IceCandidate
communicating with a non-JSEP endpoint), the most recently received (perhaps when communicating with a non-JSEP endpoint), the most
session description is assumed. recently received session description is assumed.
The IceCandidate object also contains fields to indicate which m= The IceCandidate object also contains fields to indicate which "m="
section it is associated with, which can be identified in one of two section it is associated with, which can be identified in one of two
ways, either by a m= section index, or a MID. The m= section index ways: either by an "m=" section index or by a MID. The "m=" section
is a zero-based index, with index N referring to the N+1th m= section index is a zero-based index, with index N referring to the N+1th "m="
in the session description referenced by this IceCandidate. The MID section in the session description referenced by this IceCandidate.
is a "media stream identification" value, as defined in [RFC5888], The MID is a "media stream identification" value, as defined in
Section 4, which provides a more robust way to identify the m= [RFC5888], Section 4, which provides a more robust way to identify
section in the session description, using the MID of the associated the "m=" section in the session description, using the MID of the
RtpTransceiver object (which may have been locally generated by the associated RtpTransceiver object (which may have been locally
answerer when interacting with a non-JSEP endpoint that does not generated by the answerer when interacting with a non-JSEP endpoint
support the MID attribute, as discussed in Section 5.10 below). If that does not support the MID attribute, as discussed in Section 5.10
the MID field is present in a received IceCandidate, it MUST be used below). If the MID field is present in a received IceCandidate, it
for identification; otherwise, the m= section index is used instead. MUST be used for identification; otherwise, the "m=" section index is
used instead.
When creating an IceCandidate object, JSEP implementations MUST Implementations MUST be prepared to receive objects with some fields
populate each of the candidate, ufrag, m= section index, and MID missing, as mentioned above.
fields. Implementations MUST also be prepared to receive objects
with some fields missing, as mentioned above.
3.5.3. ICE Candidate Policy 3.5.3. ICE Candidate Policy
Typically, when gathering ICE candidates, the JSEP implementation Typically, when gathering ICE candidates, the JSEP implementation
will gather all possible forms of initial candidates - host, server will gather all possible forms of initial candidates -- host, server-
reflexive, and relay. However, in certain cases, applications may reflexive, and relay. However, in certain cases, applications may
want to have more specific control over the gathering process, due to want to have more specific control over the gathering process, due to
privacy or related concerns. For example, one may want to only use privacy or related concerns. For example, one may want to only use
relay candidates, to leak as little location information as possible relay candidates, to leak as little location information as possible
(keeping in mind that this choice comes with corresponding (keeping in mind that this choice comes with corresponding
operational costs). To accomplish this, JSEP allows the application operational costs). To accomplish this, JSEP allows the application
to restrict which ICE candidates are used in a session. Note that to restrict which ICE candidates are used in a session. Note that
this filtering is applied on top of any restrictions the this filtering is applied on top of any restrictions the
implementation chooses to enforce regarding which IP addresses are implementation chooses to enforce regarding which IP addresses are
permitted for the application, as discussed in permitted for the application, as discussed in [RFC8828].
[I-D.ietf-rtcweb-ip-handling].
There may also be cases where the application wants to change which There may also be cases where the application wants to change which
types of candidates are used while the session is active. A prime types of candidates are used while the session is active. A prime
example is where a callee may initially want to use only relay example is where a callee may initially want to use only relay
candidates, to avoid leaking location information to an arbitrary candidates, to avoid leaking location information to an arbitrary
caller, but then change to use all candidates (for lower operational caller, but then change to use all candidates (for lower operational
cost) once the user has indicated they want to take the call. For cost) once the user has indicated that they want to take the call.
this scenario, the JSEP implementation MUST allow the candidate For this scenario, the JSEP implementation MUST allow the candidate
policy to be changed in mid-session, subject to the aforementioned policy to be changed in mid-session, subject to the aforementioned
interactions with local policy. interactions with local policy.
To administer the ICE candidate policy, the JSEP implementation will To administer the ICE candidate policy, the JSEP implementation will
determine the current setting at the start of each gathering phase. determine the current setting at the start of each gathering phase.
Then, during the gathering phase, the implementation MUST NOT expose Then, during the gathering phase, the implementation MUST NOT expose
candidates disallowed by the current policy to the application, use candidates disallowed by the current policy to the application, use
them as the source of connectivity checks, or indirectly expose them them as the source of connectivity checks, or indirectly expose them
via other fields, such as the raddr/rport attributes for other ICE via other fields, such as the raddr/rport attributes for other ICE
candidates. Later, if a different policy is specified by the candidates. Later, if a different policy is specified by the
application, the application can apply it by kicking off a new application, the application can apply it by kicking off a new
gathering phase via an ICE restart. gathering phase via an ICE restart.
3.5.4. ICE Candidate Pool 3.5.4. ICE Candidate Pool
JSEP applications typically inform the JSEP implementation to begin JSEP applications typically inform the JSEP implementation to begin
ICE gathering via the information supplied to setLocalDescription, as ICE gathering via the information supplied to setLocalDescription, as
the local description indicates the number of ICE components which the local description indicates the number of ICE components that
will be needed and for which candidates must be gathered. However, will be needed and for which candidates must be gathered. However,
to accelerate cases where the application knows the number of ICE to accelerate cases where the application knows the number of ICE
components to use ahead of time, it may ask the implementation to components to use ahead of time, it may ask the implementation to
gather a pool of potential ICE candidates to help ensure rapid media gather a pool of potential ICE candidates to help ensure rapid media
setup. setup.
When setLocalDescription is eventually called, and the JSEP When setLocalDescription is eventually called and the JSEP
implementation goes to gather the needed ICE candidates, it SHOULD implementation prepares to gather the needed ICE candidates, it
start by checking if any candidates are available in the pool. If SHOULD start by checking if any candidates are available in the pool.
there are candidates in the pool, they SHOULD be handed to the If there are candidates in the pool, they SHOULD be handed to the
application immediately via the ICE candidate event. If the pool application immediately via the ICE candidate event. If the pool
becomes depleted, either because a larger-than-expected number of ICE becomes depleted, either because a larger-than-expected number of ICE
components is used, or because the pool has not had enough time to components are used or because the pool has not had enough time to
gather candidates, the remaining candidates are gathered as usual. gather candidates, the remaining candidates are gathered as usual.
This only occurs for the first offer/answer exchange, after which the This only occurs for the first offer/answer exchange, after which the
candidate pool is emptied and no longer used. candidate pool is emptied and no longer used.
One example of where this concept is useful is an application that One example of where this concept is useful is an application that
expects an incoming call at some point in the future, and wants to expects an incoming call at some point in the future, and wants to
minimize the time it takes to establish connectivity, to avoid minimize the time it takes to establish connectivity, to avoid
clipping of initial media. By pre-gathering candidates into the clipping of initial media. By pre-gathering candidates into the
pool, it can exchange and start sending connectivity checks from pool, it can exchange and start sending connectivity checks from
these candidates almost immediately upon receipt of a call. Note these candidates almost immediately upon receipt of a call. Note,
though that by holding on to these pre-gathered candidates, which though, that by holding on to these pre-gathered candidates, which
will be kept alive as long as they may be needed, the application will be kept alive as long as they may be needed, the application
will consume resources on the STUN/TURN servers it is using. will consume resources on the STUN/TURN servers it is using. ("STUN"
stands for "Session Traversal Utilities for NAT".)
3.5.5. ICE Versions 3.5.5. ICE Versions
While this specification formally relies on [RFC8445], at the time of While this specification formally relies on [RFC8445], at the time of
its publication, the majority of WebRTC implementations support the its publication, the majority of WebRTC implementations support the
version of ICE described in [RFC5245]. The use of the "ice2" version of ICE described in [RFC5245]. The "ice2" attribute defined
attribute defined in [RFC8445] can be used to detect the version in in [RFC8445] can be used to detect the version in use by a remote
use by a remote endpoint and to provide a smooth transition from the endpoint and to provide a smooth transition from the older
older specification to the newer one. Implementations MUST be able specification to the newer one. Implementations MUST be able to
to accept remote descriptions that do not have the "ice2" attribute. accept remote descriptions that do not have the "ice2" attribute.
3.6. Video Size Negotiation 3.6. Video Size Negotiation
Video size negotiation is the process through which a receiver can Video size negotiation is the process through which a receiver can
use the "a=imageattr" SDP attribute [RFC6236] to indicate what video use the "a=imageattr" SDP attribute [RFC6236] to indicate what video
frame sizes it is capable of receiving. A receiver may have hard frame sizes it is capable of receiving. A receiver may have hard
limits on what its video decoder can process, or it may have some limits on what its video decoder can process, or it may have some
maximum set by policy. By specifying these limits in an maximum set by policy. By specifying these limits in an
"a=imageattr" attribute, JSEP endpoints can attempt to ensure that "a=imageattr" attribute, JSEP endpoints can attempt to ensure that
the remote sender transmits video at an acceptable resolution. the remote sender transmits video at an acceptable resolution.
However, when communicating with a non-JSEP endpoint that does not However, when communicating with a non-JSEP endpoint that does not
understand this attribute, any signaled limits may be exceeded, and understand this attribute, any signaled limits may be exceeded, and
the JSEP implementation MUST handle this gracefully, e.g., by the JSEP implementation MUST handle this gracefully, e.g., by
discarding the video. discarding the video.
Note that certain codecs support transmission of samples with aspect Note that certain codecs support transmission of samples with aspect
ratios other than 1.0 (i.e., non-square pixels). JSEP ratios other than 1.0 (i.e., non-square pixels). JSEP
implementations will not transmit non-square pixels, but SHOULD implementations will not transmit non-square pixels but SHOULD
receive and render such video with the correct aspect ratio. receive and render such video with the correct aspect ratio.
However, sample aspect ratio has no impact on the size negotiation However, sample aspect ratio has no impact on the size negotiation
described below; all dimensions are measured in pixels, whether described below; all dimensions are measured in pixels, whether
square or not. square or not.
3.6.1. Creating an imageattr Attribute 3.6.1. Creating an imageattr Attribute
The receiver will first intersect any known local limits (e.g., The receiver will first combine any known local limits (e.g.,
hardware decoder capababilities, local policy) to determine the hardware decoder capabilities or local policy) to determine the
absolute minimum and maximum sizes it can receive. If there are no absolute minimum and maximum sizes it can receive. If there are no
known local limits, the "a=imageattr" attribute SHOULD be omitted. known local limits, the "a=imageattr" attribute SHOULD be omitted.
If these local limits preclude receiving any video, i.e., the If these local limits preclude receiving any video, i.e., the
degenerate case of no permitted resolutions, the "a=imageattr" degenerate case of no permitted resolutions, the "a=imageattr"
attribute MUST be omitted, and the m= section MUST be marked as attribute MUST be omitted, and the "m=" section MUST be marked as
sendonly/inactive, as appropriate. sendonly/inactive, as appropriate.
Otherwise, an "a=imageattr" attribute is created with "recv" Otherwise, an "a=imageattr" attribute is created with a "recv"
direction, and the resulting resolution space formed from the direction, and the resulting resolution space formed from the
aforementioned intersection is used to specify its minimum and aforementioned intersection is used to specify its minimum and
maximum x= and y= values. maximum "x=" and "y=" values.
The rules here express a single set of preferences, and therefore, The rules here express a single set of preferences, and therefore,
the "a=imageattr" q= value is not important. It SHOULD be set to the "a=imageattr" "q=" value is not important. It SHOULD be set to
1.0. "1.0".
The "a=imageattr" field is payload type specific. When all video The "a=imageattr" field is payload type specific. When all video
codecs supported have the same capabilities, use of a single codecs supported have the same capabilities, use of a single
attribute, with the wildcard payload type (*), is RECOMMENDED. attribute, with the wildcard payload type (*), is RECOMMENDED.
However, when the supported video codecs have different limitations, However, when the supported video codecs have different limitations,
specific "a=imageattr" attributes MUST be inserted for each payload specific "a=imageattr" attributes MUST be inserted for each payload
type. type.
As an example, consider a system with a multiformat video decoder, As an example, consider a system with a multiformat video decoder,
which is capable of decoding any resolution from 48x48 to 720p, In which is capable of decoding any resolution from 48x48 to 720p. In
this case, the implementation would generate this attribute: this case, the implementation would generate this attribute:
a=imageattr:* recv [x=[48:1280],y=[48:720],q=1.0] a=imageattr:* recv [x=[48:1280],y=[48:720],q=1.0]
This declaration indicates that the receiver is capable of decoding This declaration indicates that the receiver is capable of decoding
any image resolution from 48x48 up to 1280x720 pixels. any image resolution from 48x48 up to 1280x720 pixels.
3.6.2. Interpreting imageattr Attributes 3.6.2. Interpreting imageattr Attributes
[RFC6236] defines "a=imageattr" to be an advisory field. This means [RFC6236] defines "a=imageattr" to be an advisory field. This means
that it does not absolutely constrain the video formats that the that it does not absolutely constrain the video formats that the
sender can use, but gives an indication of the preferred values. sender can use but gives an indication of the preferred values.
This specification prescribes more specific behavior. When a This specification prescribes behavior that is more specific. When a
MediaStreamTrack, which is producing video of a certain resolution MediaStreamTrack, which is producing video of a certain resolution
(the "track resolution"), is attached to a RtpSender, which is (the "track resolution"), is attached to an RtpSender, which is
encoding the track video at the same or lower resolution(s) (the encoding the track video at the same or lower resolution(s) (the
"encoder resolutions"), and a remote description is applied that "encoder resolutions"), and a remote description is applied that
references the sender and contains valid "a=imageattr recv" references the sender and contains valid "a=imageattr recv"
attributes, it MUST follow the rules below to ensure the sender does attributes, it MUST follow the rules below to ensure that the sender
not transmit a resolution that would exceed the size criteria does not transmit a resolution that would exceed the size criteria
specified in the attributes. These rules MUST be followed as long as specified in the attributes. These rules MUST be followed as long as
the attributes remain present in the remote description, including the attributes remain present in the remote description, including
cases in which the track changes its resolution, or is replaced with cases in which the track changes its resolution or is replaced with a
a different track. different track.
Depending on how the RtpSender is configured, it may be producing a Depending on how the RtpSender is configured, it may be producing a
single encoding at a certain resolution, or, if simulcast Section 3.7 single encoding at a certain resolution or, if simulcast
has been negotiated, multiple encodings, each at their own specific (Section 3.7) has been negotiated, multiple encodings, each at their
resolution. In addition, depending on the configuration, each own specific resolution. In addition, depending on the
encoding may have the flexibility to reduce resolution when needed, configuration, each encoding may have the flexibility to reduce
or may be locked to a specific output resolution. resolution when needed or may be locked to a specific output
resolution.
For each encoding being produced by the RtpSender, the set of For each encoding being produced by the RtpSender, the set of
"a=imageattr recv" attributes in the corresponding m= section of the "a=imageattr recv" attributes in the corresponding "m=" section of
remote description is processed to determine what should be the remote description is processed to determine what should be
transmitted. Only attributes that reference the media format transmitted. Only attributes that reference the media format
selected for the encoding are considered; each such attribute is selected for the encoding are considered; each such attribute is
evaluated individually, starting with the attribute with the highest evaluated individually, starting with the attribute with the highest
"q=" value. If multiple attributes have the same "q=" value, they "q=" value. If multiple attributes have the same "q=" value, they
are evaluated in the order they appear in their containing m= are evaluated in the order they appear in their containing "m="
section. Note that while JSEP endpoints will include at most one section. Note that while JSEP endpoints will include at most one
"a=imageattr recv" attribute per media format, JSEP endpoints may "a=imageattr recv" attribute per media format, JSEP endpoints may
receive session descriptions from non-JSEP endpoints with m= sections receive session descriptions from non-JSEP endpoints with "m="
that contain multiple such attributes. sections that contain multiple such attributes.
For each "a=imageattr recv" attribute, the following rules are For each "a=imageattr recv" attribute, the following rules are
applied. If this processing is successful, the encoding is applied. If this processing is successful, the encoding is
transmitted accordingly, and no further attributes are considered for transmitted accordingly, and no further attributes are considered for
that encoding. Otherwise, the next attribute is evaluated, in the that encoding. Otherwise, the next attribute is evaluated, in the
aforementioned order. If none of the supplied attributes can be aforementioned order. If none of the supplied attributes can be
processed successfully, the encoding MUST NOT be transmitted, and an processed successfully, the encoding MUST NOT be transmitted, and an
error SHOULD be raised to the application. error SHOULD be raised to the application.
o The limits from the attribute are compared to the encoder * The limits from the attribute are compared to the encoder
resolution. Only the specific limits mentioned below are resolution. Only the specific limits mentioned below are
considered; any other values, such as picture aspect ratio, MUST considered; any other values, such as picture aspect ratio, MUST
be ignored. When considering a MediaStreamTrack that is producing be ignored. When considering a MediaStreamTrack that is producing
rotated video, the unrotated resolution MUST be used for the rotated video, the unrotated resolution MUST be used for the
checks. This is required regardless of whether the receiver checks. This is required regardless of whether the receiver
supports performing receive-side rotation (e.g., through CVO supports performing receive-side rotation (e.g., through
[TS26.114]), as it significantly simplifies the matching logic. Coordination of Video Orientation (CVO) [TS26.114]), as it
significantly simplifies the matching logic.
o If the attribute includes a "sar=" (sample aspect ratio) value set * If the attribute includes a "sar=" (sample aspect ratio) value set
to something other than "1.0", indicating the receiver wants to to something other than "1.0", indicating that the receiver wants
receive non-square pixels, this cannot be satisfied and the to receive non-square pixels, this cannot be satisfied and the
attribute MUST NOT be used. attribute MUST NOT be used.
o If the encoder resolution exceeds the maximum size permitted by * If the encoder resolution exceeds the maximum size permitted by
the attribute, and the encoder is allowed to adjust its the attribute and the encoder is allowed to adjust its resolution,
resolution, the encoder SHOULD apply downscaling in order to the encoder SHOULD apply downscaling in order to satisfy the
satisfy the limits. Downscaling MUST NOT change the picture limits. Downscaling MUST NOT change the picture aspect ratio of
aspect ratio of the encoding, ignoring any trivial differences due the encoding, ignoring any trivial differences due to rounding.
to rounding. For example, if the encoder resolution is 1280x720, For example, if the encoder resolution is 1280x720 and the
and the attribute specified a maximum of 640x480, the expected attribute specified a maximum of 640x480, the expected output
output resolution would be 640x360. If downscaling cannot be resolution would be 640x360. If downscaling cannot be applied,
applied, the attribute MUST NOT be used. the attribute MUST NOT be used.
o If the encoder resolution is less than the minimum size permitted * If the encoder resolution is less than the minimum size permitted
by the attribute, the attribute MUST NOT be used; the encoder MUST by the attribute, the attribute MUST NOT be used; the encoder MUST
NOT apply upscaling. JSEP implementations SHOULD avoid this NOT apply upscaling. JSEP implementations SHOULD avoid this
situation by allowing receipt of arbitrarily small resolutions, situation by allowing receipt of arbitrarily small resolutions,
perhaps via fallback to a software decoder. perhaps via fallback to a software decoder.
o If the encoder resolution is within the maximum and minimum sizes, * If the encoder resolution is within the maximum and minimum sizes,
no action is needed. no action is needed.
3.7. Simulcast 3.7. Simulcast
JSEP supports simulcast transmission of a MediaStreamTrack, where JSEP supports simulcast transmission of a MediaStreamTrack, where
multiple encodings of the source media can be transmitted within the multiple encodings of the source media can be transmitted within the
context of a single m= section. The current JSEP API is designed to context of a single "m=" section. The current JSEP API is designed
allow applications to send simulcasted media but only to receive a to allow applications to send simulcasted media but only to receive a
single encoding. This allows for multi-user scenarios where each single encoding. This allows for multi-user scenarios where each
sending client sends multiple encodings to a server, which then, for sending client sends multiple encodings to a server, which then, for
each receiving client, chooses the appropriate encoding to forward. each receiving client, chooses the appropriate encoding to forward.
Applications request support for simulcast by configuring multiple Applications request support for simulcast by configuring multiple
encodings on an RtpSender. Upon generation of an offer or answer, encodings on an RtpSender. Upon generation of an offer or answer,
these encodings are indicated via SDP markings on the corresponding these encodings are indicated via SDP markings on the corresponding
m= section, as described below. Receivers that understand simulcast "m=" section, as described below. Receivers that understand
and are willing to receive it will also include SDP markings to simulcast and are willing to receive it will also include SDP
indicate their support, and JSEP endpoints will use these markings to markings to indicate their support, and JSEP endpoints will use these
determine whether simulcast is permitted for a given RtpSender. If markings to determine whether simulcast is permitted for a given
simulcast support is not negotiated, the RtpSender will only use the RtpSender. If simulcast support is not negotiated, the RtpSender
first configured encoding. will only use the first configured encoding.
Note that the exact simulcast parameters are up to the sending Note that the exact simulcast parameters are up to the sending
application. While the aforementioned SDP markings are provided to application. While the aforementioned SDP markings are provided to
ensure the remote side can receive and demux multiple simulcast ensure that the remote side can receive and demux multiple simulcast
encodings, the specific resolutions and bitrates to be used for each encodings, the specific resolutions and bitrates to be used for each
encoding are purely a send-side decision in JSEP. encoding are purely a send-side decision in JSEP.
JSEP currently does not provide a mechanism to configure receipt of JSEP currently does not provide a mechanism to configure receipt of
simulcast. This means that if simulcast is offered by the remote simulcast. This means that if simulcast is offered by the remote
endpoint, the answer generated by a JSEP endpoint will not indicate endpoint, the answer generated by a JSEP endpoint will not indicate
support for receipt of simulcast, and as such the remote endpoint support for receipt of simulcast, and as such the remote endpoint
will only send a single encoding per m= section. will only send a single encoding per "m=" section.
In addition, JSEP does not provide a mechanism to handle an incoming In addition, JSEP does not provide a mechanism to handle an incoming
offer requesting simulcast from the JSEP endpoint. This means that offer requesting simulcast from the JSEP endpoint. This means that
setting up simulcast in the case where the JSEP endpoint receives the setting up simulcast in the case where the JSEP endpoint receives the
initial offer requires out-of-band signaling or SDP inspection. initial offer requires out-of-band signaling or SDP inspection.
However, in the case where the JSEP endpoint sets up simulcast in its However, in the case where the JSEP endpoint sets up simulcast in its
in initial offer, any established simulcast streams will continue to initial offer, any established simulcast streams will continue to
work upon receipt of an incoming re-offer. Future versions of this work upon receipt of an incoming re-offer. Future versions of this
specification may add additional APIs to handle the incoming initial specification may add additional APIs to handle the incoming initial
offer scenario. offer scenario.
When using JSEP to transmit multiple encodings from a RtpSender, the When using JSEP to transmit multiple encodings from an RtpSender, the
techniques from [I-D.ietf-mmusic-sdp-simulcast] and techniques from [RFC8853] and [RFC8851] are used. Specifically, when
[I-D.ietf-mmusic-rid] are used. Specifically, when multiple multiple encodings have been configured for an RtpSender, the "m="
encodings have been configured for a RtpSender, the m= section for section for the RtpSender will include an "a=simulcast" attribute, as
the RtpSender will include an "a=simulcast" attribute, as defined in defined in [RFC8853], Section 5.1, with a "send" simulcast stream
[I-D.ietf-mmusic-sdp-simulcast], Section 6.2, with a "send" simulcast description that lists each desired encoding, and no "recv" simulcast
stream description that lists each desired encoding, and no "recv" stream description. The "m=" section will also include an "a=rid"
simulcast stream description. The m= section will also include an attribute for each encoding, as specified in [RFC8851], Section 4;
"a=rid" attribute for each encoding, as specified in the use of Restriction Identifiers (RIDs, also called rid-ids or
[I-D.ietf-mmusic-rid], Section 4; the use of RID identifiers allows RtpStreamIds) allows the individual encodings to be disambiguated
the individual encodings to be disambiguated even though they are all even though they are all part of the same "m=" section.
part of the same m= section.
3.8. Interactions With Forking 3.8. Interactions with Forking
Some call signaling systems allow various types of forking where an Some call signaling systems allow various types of forking where an
SDP Offer may be provided to more than one device. For example, SIP SDP Offer may be provided to more than one device. For example, SIP
[RFC3261] defines both a "Parallel Search" and "Sequential Search". [RFC3261] defines both a "parallel search" and "sequential search".
Although these are primarily signaling level issues that are outside Although these are primarily signaling-level issues that are outside
the scope of JSEP, they do have some impact on the configuration of the scope of JSEP, they do have some impact on the configuration of
the media plane that is relevant. When forking happens at the the media plane that is relevant. When forking happens at the
signaling layer, the JavaScript application responsible for the signaling layer, the JavaScript application responsible for the
signaling needs to make the decisions about what media should be sent signaling needs to make the decisions about what media should be sent
or received at any point of time, as well as which remote endpoint it or received at any point in time, as well as which remote endpoint it
should communicate with; JSEP is used to make sure the media engine should communicate with; JSEP is used to make sure the media engine
can make the RTP and media perform as required by the application. can make the RTP and media perform as required by the application.
The basic operations that the applications can have the media engine The basic operations that the applications can have the media engine
do are: do are as follows:
o Start exchanging media with a given remote peer, but keep all the * Start exchanging media with a given remote peer, but keep all the
resources reserved in the offer. resources reserved in the offer.
o Start exchanging media with a given remote peer, and free any * Start exchanging media with a given remote peer, and free any
resources in the offer that are not being used. resources in the offer that are not being used.
3.8.1. Sequential Forking 3.8.1. Sequential Forking
Sequential forking involves a call being dispatched to multiple Sequential forking involves a call being dispatched to multiple
remote callees, where each callee can accept the call, but only one remote callees, where each callee can accept the call, but only one
active session ever exists at a time; no mixing of received media is active session ever exists at a time; no mixing of received media is
performed. performed.
JSEP handles sequential forking well, allowing the application to JSEP handles sequential forking well, allowing the application to
easily control the policy for selecting the desired remote endpoint. easily control the policy for selecting the desired remote endpoint.
When an answer arrives from one of the callees, the application can When an answer arrives from one of the callees, the application can
choose to apply it either as a provisional answer, leaving open the choose to apply it as either (1) a provisional answer, leaving open
possibility of using a different answer in the future, or apply it as the possibility of using a different answer in the future or (2) a
a final answer, ending the setup flow. final answer, ending the setup flow.
In a "first-one-wins" situation, the first answer will be applied as In a "first-one-wins" situation, the first answer will be applied as
a final answer, and the application will reject any subsequent a final answer, and the application will reject any subsequent
answers. In SIP parlance, this would be ACK + BYE. answers. In SIP parlance, this would be ACK + BYE.
In a "last-one-wins" situation, all answers would be applied as In a "last-one-wins" situation, all answers would be applied as
provisional answers, and any previous call leg will be terminated. provisional answers, and any previous call leg will be terminated.
At some point, the application will end the setup process, perhaps At some point, the application will end the setup process, perhaps
with a timer; at this point, the application could reapply the with a timer; at this point, the application could reapply the
pending remote description as a final answer. pending remote description as a final answer.
3.8.2. Parallel Forking 3.8.2. Parallel Forking
Parallel forking involves a call being dispatched to multiple remote Parallel forking involves a call being dispatched to multiple remote
callees, where each callee can accept the call, and multiple callees, where each callee can accept the call and multiple
simultaneous active signaling sessions can be established as a simultaneous active signaling sessions can be established as a
result. If multiple callees send media at the same time, the result. If multiple callees send media at the same time, the
possibilities for handling this are described in [RFC3960], possibilities for handling this are described in [RFC3960],
Section 3.1. Most SIP devices today only support exchanging media Section 3.1. Most SIP devices today only support exchanging media
with a single device at a time, and do not try to mix multiple early with a single device at a time and do not try to mix multiple early
media audio sources, as that could result in a confusing situation. media audio sources, as that could result in a confusing situation.
For example, consider having a European ringback tone mixed together For example, consider having a European ringback tone mixed together
with the North American ringback tone - the resulting sound would not with the North American ringback tone -- the resulting sound would
be like either tone, and would confuse the user. If the signaling not be like either tone and would confuse the user. If the signaling
application wishes to only exchange media with one of the remote application wishes to only exchange media with one of the remote
endpoints at a time, then from a media engine point of view, this is endpoints at a time, then from a media engine point of view, this is
exactly like the sequential forking case. exactly like the sequential forking case.
In the parallel forking case where the JavaScript application wishes In the parallel forking case where the JavaScript application wishes
to simultaneously exchange media with multiple peers, the flow is to simultaneously exchange media with multiple peers, the flow is
slightly more complex, but the JavaScript application can follow the slightly more complex, but the JavaScript application can follow the
strategy that [RFC3960] describes using UPDATE. The UPDATE approach strategy that [RFC3960] describes, using UPDATE. The UPDATE approach
allows the signaling to set up a separate media flow for each peer allows the signaling to set up a separate media flow for each peer
that it wishes to exchange media with. In JSEP, this offer used in that it wishes to exchange media with. In JSEP, this offer used in
the UPDATE would be formed by simply creating a new PeerConnection the UPDATE would be formed by simply creating a new PeerConnection
(see Section 4.1) and making sure that the same local media streams (see Section 4.1) and making sure that the same local media streams
have been added into this new PeerConnection. Then the new have been added into this new PeerConnection. Then the new
PeerConnection object would produce a SDP offer that could be used by PeerConnection object would produce an SDP offer that could be used
the signaling to perform the UPDATE strategy discussed in [RFC3960]. by the signaling to perform the UPDATE strategy discussed in
[RFC3960].
As a result of sharing the media streams, the application will end up As a result of sharing the media streams, the application will end up
with N parallel PeerConnection sessions, each with a local and remote with N parallel PeerConnection sessions, each with a local and remote
description and their own local and remote addresses. The media flow description and their own local and remote addresses. The media flow
from these sessions can be managed using setDirection (see from these sessions can be managed using setDirection (see
Section 4.2.3), or the application can choose to play out the media Section 4.2.3), or the application can choose to play out the media
from all sessions mixed together. Of course, if the application from all sessions mixed together. Of course, if the application
wants to only keep a single session, it can simply terminate the wants to only keep a single session, it can simply terminate the
sessions that it no longer needs. sessions that it no longer needs.
4. Interface 4. Interface
This section details the basic operations that must be present to This section details the basic operations that must be present to
implement JSEP functionality. The actual API exposed in the W3C API implement JSEP functionality. The actual API exposed in the W3C API
may have somewhat different syntax, but should map easily to these may have somewhat different syntax but should map easily to these
concepts. concepts.
4.1. PeerConnection 4.1. PeerConnection
4.1.1. Constructor 4.1.1. Constructor
The PeerConnection constructor allows the application to specify The PeerConnection constructor allows the application to specify
global parameters for the media session, such as the STUN/TURN global parameters for the media session, such as the STUN/TURN
servers and credentials to use when gathering candidates, as well as servers and credentials to use when gathering candidates, as well as
the initial ICE candidate policy and pool size, and also the bundle the initial ICE candidate policy and pool size, and also the bundle
policy to use. policy to use.
If an ICE candidate policy is specified, it functions as described in If an ICE candidate policy is specified, it functions as described in
Section 3.5.3, causing the JSEP implementation to only surface the Section 3.5.3, causing the JSEP implementation to only surface the
permitted candidates (including any implementation-internal permitted candidates (including any implementation-internal
filtering) to the application, and only use those candidates for filtering) to the application and only use those candidates for
connectivity checks. The set of available policies is as follows: connectivity checks. The set of available policies is as follows:
all: All candidates permitted by implementation policy will be all: All candidates permitted by implementation policy will be
gathered and used. gathered and used.
relay: All candidates except relay candidates will be filtered out. relay: All candidates except relay candidates will be filtered out.
This obfuscates the location information that might be ascertained This obfuscates the location information that might be ascertained
by the remote peer from the received candidates. Depending on how by the remote peer from the received candidates. Depending on how
the application deploys and chooses relay servers, this could the application deploys and chooses relay servers, this could
obfuscate location to a metro or possibly even global level. obfuscate location to a metro or possibly even global level.
The default ICE candidate policy MUST be set to "all" as this is The default ICE candidate policy MUST be set to "all", as this is
generally the desired policy, and also typically reduces use of generally the desired policy and also typically reduces the use of
application TURN server resources significantly. application TURN server resources significantly.
If a size is specified for the ICE candidate pool, this indicates the If a size is specified for the ICE candidate pool, this indicates the
number of ICE components to pre-gather candidates for. Because pre- number of ICE components to pre-gather candidates for. Because
gathering results in utilizing STUN/TURN server resources for pre-gathering results in utilizing STUN/TURN server resources for
potentially long periods of time, this must only occur upon potentially long periods of time, this MUST only occur upon
application request, and therefore the default candidate pool size application request, and therefore the default candidate pool size
MUST be zero. MUST be zero.
The application can specify its preferred policy regarding use of The application can specify its preferred policy regarding use of
bundle, the multiplexing mechanism defined in bundle, the multiplexing mechanism defined in [RFC8843]. Regardless
[I-D.ietf-mmusic-sdp-bundle-negotiation]. Regardless of policy, the of policy, the application will always try to negotiate bundle onto a
application will always try to negotiate bundle onto a single single transport and will offer a single bundle group across all "m="
transport, and will offer a single bundle group across all m=
sections; use of this single transport is contingent upon the sections; use of this single transport is contingent upon the
answerer accepting bundle. However, by specifying a policy from the answerer accepting bundle. However, by specifying a policy from the
list below, the application can control exactly how aggressively it list below, the application can control exactly how aggressively it
will try to bundle media streams together, which affects how it will will try to bundle media streams together, which affects how it will
interoperate with a non-bundle-aware endpoint. When negotiating with interoperate with a non-bundle-aware endpoint. When negotiating with
a non-bundle-aware endpoint, only the streams not marked as bundle- a non-bundle-aware endpoint, only the streams not marked as bundle-
only streams will be established. only streams will be established.
The set of available policies is as follows: The set of available policies is as follows:
balanced: The first m= section of each type (audio, video, or balanced: The first "m=" section of each type (audio, video, or
application) will contain transport parameters, which will allow application) will contain transport parameters, which will allow
an answerer to unbundle that section. The second and any an answerer to unbundle that section. The second and any
subsequent m= section of each type will be marked bundle-only. subsequent "m=" sections of each type will be marked bundle-only.
The result is that if there are N distinct media types, then The result is that if there are N distinct media types, then
candidates will be gathered for for N media streams. This policy candidates will be gathered for N media streams. This policy
balances desire to multiplex with the need to ensure basic audio balances desire to multiplex with the need to ensure that basic
and video can still be negotiated in legacy cases. When acting as audio and video can still be negotiated in legacy cases. When
answerer, if there is no bundle group in the offer, the acting as answerer, if there is no bundle group in the offer, the
implementation will reject all but the first m= section of each implementation will reject all but the first "m=" section of each
type. type.
max-compat: All m= sections will contain transport parameters; none max-compat: All "m=" sections will contain transport parameters;
will be marked as bundle-only. This policy will allow all streams none will be marked as bundle-only. This policy will allow all
to be received by non-bundle-aware endpoints, but require separate streams to be received by non-bundle-aware endpoints but will
candidates to be gathered for each media stream. require separate candidates to be gathered for each media stream.
max-bundle: Only the first m= section will contain transport max-bundle: Only the first "m=" section will contain transport
parameters; all streams other than the first will be marked as parameters; all streams other than the first will be marked as
bundle-only. This policy aims to minimize candidate gathering and bundle-only. This policy aims to minimize candidate gathering and
maximize multiplexing, at the cost of less compatibility with maximize multiplexing, at the cost of less compatibility with
legacy endpoints. When acting as answerer, the implementation legacy endpoints. When acting as answerer, the implementation
will reject any m= sections other than the first m= section, will reject any "m=" sections other than the first "m=" section,
unless they are in the same bundle group as that m= section. unless they are in the same bundle group as that "m=" section.
As it provides the best tradeoff between performance and As it provides the best trade-off between performance and
compatibility with legacy endpoints, the default bundle policy MUST compatibility with legacy endpoints, the default bundle policy MUST
be set to "balanced". be set to "balanced".
The application can specify its preferred policy regarding use of The application can specify its preferred policy regarding use of
RTP/RTCP multiplexing [RFC5761] using one of the following policies: RTP/RTCP multiplexing [RFC5761] using one of the following policies:
negotiate: The JSEP implementation will gather both RTP and RTCP negotiate: The JSEP implementation will gather both RTP and RTCP
candidates but also will offer "a=rtcp-mux", thus allowing for candidates but also will offer "a=rtcp-mux", thus allowing for
compatibility with either multiplexing or non-multiplexing compatibility with either multiplexing or non-multiplexing
endpoints. endpoints.
require: The JSEP implementation will only gather RTP candidates and require: The JSEP implementation will only gather RTP candidates and
will insert an "a=rtcp-mux-only" indication into any new m= will insert an "a=rtcp-mux-only" indication into any new "m="
sections in offers it generates. This halves the number of sections in offers it generates. This halves the number of
candidates that the offerer needs to gather. Applying a candidates that the offerer needs to gather. Applying a
description with an m= section that does not contain an "a=rtcp- description with an "m=" section that does not contain an "a=rtcp-
mux" attribute will cause an error to be returned. mux" attribute will cause an error to be returned.
The default multiplexing policy MUST be set to "require". The default multiplexing policy MUST be set to "require".
Implementations MAY choose to reject attempts by the application to Implementations MAY choose to reject attempts by the application to
set the multiplexing policy to "negotiate". set the multiplexing policy to "negotiate".
4.1.2. addTrack 4.1.2. addTrack
The addTrack method adds a MediaStreamTrack to the PeerConnection, The addTrack method adds a MediaStreamTrack to the PeerConnection,
using the MediaStream argument to associate the track with other using the MediaStream argument to associate the track with other
tracks in the same MediaStream, so that they can be added to the same tracks in the same MediaStream, so that they can be added to the same
"LS" group when creating an offer or answer. Adding tracks to the "LS" (Lip Synchronization) group when creating an offer or answer.
same "LS" group indicates that the playback of these tracks should be Adding tracks to the same "LS" group indicates that the playback of
synchronized for proper lip sync, as described in [RFC5888], these tracks should be synchronized for proper lip sync, as described
Section 7. addTrack attempts to minimize the number of transceivers in [RFC5888], Section 7. addTrack attempts to minimize the number of
as follows: If the PeerConnection is in the "have-remote-offer" transceivers as follows: if the PeerConnection is in the
state, the track will be attached to the first compatible transceiver "have-remote-offer" state, the track will be attached to the first
that was created by the most recent call to setRemoteDescription() compatible transceiver that was created by the most recent call to
and does not have a local track. Otherwise, a new transceiver will setRemoteDescription and does not have a local track. Otherwise, a
be created, as described in Section 4.1.4. new transceiver will be created, as described in Section 4.1.4.
4.1.3. removeTrack 4.1.3. removeTrack
The removeTrack method removes a MediaStreamTrack from the The removeTrack method removes a MediaStreamTrack from the
PeerConnection, using the RtpSender argument to indicate which sender PeerConnection, using the RtpSender argument to indicate which sender
should have its track removed. The sender's track is cleared, and should have its track removed. The sender's track is cleared, and
the sender stops sending. Future calls to createOffer will mark the the sender stops sending. Future calls to createOffer will mark the
m= section associated with the sender as recvonly (if "m=" section associated with the sender as recvonly (if
transceiver.direction is sendrecv) or as inactive (if transceiver.direction is sendrecv) or as inactive (if
transceiver.direction is sendonly). transceiver.direction is sendonly).
4.1.4. addTransceiver 4.1.4. addTransceiver
The addTransceiver method adds a new RtpTransceiver to the The addTransceiver method adds a new RtpTransceiver to the
PeerConnection. If a MediaStreamTrack argument is provided, then the PeerConnection. If a MediaStreamTrack argument is provided, then the
transceiver will be configured with that media type and the track transceiver will be configured with that media type and the track
will be attached to the transceiver. Otherwise, the application MUST will be attached to the transceiver. Otherwise, the application MUST
explicitly specify the type; this mode is useful for creating explicitly specify the type; this mode is useful for creating
recvonly transceivers as well as for creating transceivers to which a recvonly transceivers as well as for creating transceivers to which a
track can be attached at some later point. track can be attached at some later point.
At the time of creation, the application can also specify a At the time of creation, the application can also specify a
transceiver direction attribute, a set of MediaStreams which the transceiver direction attribute, a set of MediaStreams that the
transceiver is associated with (allowing LS group assignments), and a transceiver is associated with (allowing "LS" group assignments), and
set of encodings for the media (used for simulcast as described in a set of encodings for the media (used for simulcast as described in
Section 3.7). Section 3.7).
4.1.5. createDataChannel 4.1.5. onaddtrack Event
The onaddtrack event is dispatched to the application when a new
remote track has been signaled as a result of a setRemoteDescription
call. The new track is supplied as a MediaStreamTrack object in the
event, along with the MediaStream(s) the track is part of.
4.1.6. createDataChannel
The createDataChannel method creates a new data channel and attaches The createDataChannel method creates a new data channel and attaches
it to the PeerConnection. If no data channel currently exists for it to the PeerConnection. If no data channel currently exists for
this PeerConnection, then a new offer/answer exchange is required. this PeerConnection, then a new offer/answer exchange is required.
All data channels on a given PeerConnection share the same SCTP/DTLS All data channels on a given PeerConnection share the same SCTP/DTLS
association and therefore the same m= section, so subsequent creation association ("SCTP" stands for "Stream Control Transmission
of data channels does not have any impact on the JSEP state. Protocol") and therefore the same "m=" section, so subsequent
creation of data channels does not have any impact on the JSEP state.
The createDataChannel method also includes a number of arguments The createDataChannel method also includes a number of arguments that
which are used by the PeerConnection (e.g., maxPacketLifetime) but are used by the PeerConnection (e.g., maxPacketLifetime) but are not
are not reflected in the SDP and do not affect the JSEP state. reflected in the SDP and do not affect the JSEP state.
4.1.6. createOffer 4.1.7. ondatachannel Event
The createOffer method generates a blob of SDP that contains a The ondatachannel event is dispatched to the application when a new
[RFC3264] offer with the supported configurations for the session, data channel has been negotiated by the remote side, which can occur
at any time after the underlying SCTP/DTLS association has been
established. The new data channel object is supplied in the event.
4.1.8. createOffer
The createOffer method generates a blob of SDP that contains an offer
per [RFC3264] with the supported configurations for the session,
including descriptions of the media added to this PeerConnection, the including descriptions of the media added to this PeerConnection, the
codec/RTP/RTCP options supported by this implementation, and any codec, RTP, and RTCP options supported by this implementation, and
candidates that have been gathered by the ICE agent. An options any candidates that have been gathered by the ICE agent. An options
parameter may be supplied to provide additional control over the parameter may be supplied to provide additional control over the
generated offer. This options parameter allows an application to generated offer. This options parameter allows an application to
trigger an ICE restart, for the purpose of reestablishing trigger an ICE restart, for the purpose of reestablishing
connectivity. connectivity.
In the initial offer, the generated SDP will contain all desired In the initial offer, the generated SDP will contain all desired
functionality for the session (functionality that is supported but functionality for the session (functionality that is supported but
not desired by default may be omitted); for each SDP line, the not desired by default may be omitted); for each SDP line, the
generation of the SDP will follow the process defined for generating generation of the SDP will follow the process defined for generating
an initial offer from the document that specifies the given SDP line. an initial offer from the specification that defines the given SDP
The exact handling of initial offer generation is detailed in line. The exact handling of initial offer generation is detailed in
Section 5.2.1 below. Section 5.2.1 below.
In the event createOffer is called after the session is established, In the event createOffer is called after the session is established,
createOffer will generate an offer to modify the current session createOffer will generate an offer to modify the current session
based on any changes that have been made to the session, e.g., adding based on any changes that have been made to the session, e.g., adding
or stopping RtpTransceivers, or requesting an ICE restart. For each or stopping RtpTransceivers, or requesting an ICE restart. For each
existing stream, the generation of each SDP line must follow the existing stream, the generation of each SDP line MUST follow the
process defined for generating an updated offer from the RFC that process defined for generating an updated offer from the RFC that
specifies the given SDP line. For each new stream, the generation of specifies the given SDP line. For each new stream, the generation of
the SDP must follow the process of generating an initial offer, as the SDP MUST follow the process of generating an initial offer, as
mentioned above. If no changes have been made, or for SDP lines that mentioned above. If no changes have been made, or for SDP lines that
are unaffected by the requested changes, the offer will only contain are unaffected by the requested changes, the offer will only contain
the parameters negotiated by the last offer-answer exchange. The the parameters negotiated by the last offer/answer exchange. The
exact handling of subsequent offer generation is detailed in exact handling of subsequent offer generation is detailed in
Section 5.2.2. below. Section 5.2.2 below.
Session descriptions generated by createOffer must be immediately Session descriptions generated by createOffer MUST be immediately
usable by setLocalDescription; if a system has limited resources usable by setLocalDescription; if a system has limited resources
(e.g. a finite number of decoders), createOffer should return an (e.g., a finite number of decoders), createOffer SHOULD return an
offer that reflects the current state of the system, so that offer that reflects the current state of the system, so that
setLocalDescription will succeed when it attempts to acquire those setLocalDescription will succeed when it attempts to acquire those
resources. resources.
Calling this method may do things such as generating new ICE Calling this method may do things such as generating new ICE
credentials, but does not change the PeerConnection state, trigger credentials, but it does not change the PeerConnection state, trigger
candidate gathering, or cause media to start or stop flowing. candidate gathering, or cause media to start or stop flowing.
Specifically, the offer is not applied, and does not become the Specifically, the offer is not applied, and does not become the
pending local description, until setLocalDescription is called. pending local description, until setLocalDescription is called.
4.1.7. createAnswer 4.1.9. createAnswer
The createAnswer method generates a blob of SDP that contains a The createAnswer method generates a blob of SDP that contains an SDP
[RFC3264] SDP answer with the supported configuration for the session answer per [RFC3264] with the supported configuration for the session
that is compatible with the parameters supplied in the most recent that is compatible with the parameters supplied in the most recent
call to setRemoteDescription, which MUST have been called prior to call to setRemoteDescription; setRemoteDescription MUST have been
calling createAnswer. Like createOffer, the returned blob contains called prior to calling createAnswer. Like createOffer, the returned
descriptions of the media added to this PeerConnection, the blob contains descriptions of the media added to this PeerConnection,
codec/RTP/RTCP options negotiated for this session, and any the codec/RTP/RTCP options negotiated for this session, and any
candidates that have been gathered by the ICE agent. An options candidates that have been gathered by the ICE agent. An options
parameter may be supplied to provide additional control over the parameter may be supplied to provide additional control over the
generated answer. generated answer.
As an answer, the generated SDP will contain a specific configuration As an answer, the generated SDP will contain a specific configuration
that specifies how the media plane should be established; for each that specifies how the media plane should be established; for each
SDP line, the generation of the SDP must follow the process defined SDP line, the generation of the SDP MUST follow the process defined
for generating an answer from the document that specifies the given for generating an answer from the specification that defines the
SDP line. The exact handling of answer generation is detailed in given SDP line. The exact handling of answer generation is detailed
Section 5.3. below. in Section 5.3 below.
Session descriptions generated by createAnswer must be immediately Session descriptions generated by createAnswer MUST be immediately
usable by setLocalDescription; like createOffer, the returned usable by setLocalDescription; like createOffer, the returned
description should reflect the current state of the system. description SHOULD reflect the current state of the system.
Calling this method may do things such as generating new ICE Calling this method may do things such as generating new ICE
credentials, but does not change the PeerConnection state, trigger credentials, but it does not change the PeerConnection state, trigger
candidate gathering, or or cause a media state change. Specifically, candidate gathering, or cause a media state change. Specifically,
the answer is not applied, and does not become the current local the answer is not applied, and does not become the current local
description, until setLocalDescription is called. description, until setLocalDescription is called.
4.1.8. SessionDescriptionType 4.1.10. SessionDescriptionType
Session description objects (RTCSessionDescription) may be of type Session description objects (RTCSessionDescription) may be of type
"offer", "pranswer", "answer" or "rollback". These types provide "offer", "pranswer", "answer", or "rollback". These types provide
information as to how the description parameter should be parsed, and information as to how the description parameter should be parsed and
how the media state should be changed. how the media state should be changed.
"offer" indicates that a description should be parsed as an offer; "offer" indicates that a description MUST be parsed as an offer; said
said description may include many possible media configurations. A description may include many possible media configurations. A
description used as an "offer" may be applied anytime the description used as an "offer" may be applied any time the
PeerConnection is in a stable state, or as an update to a previously PeerConnection is in a "stable" state or applied as an update to a
supplied but unanswered "offer". previously supplied but unanswered "offer".
"pranswer" indicates that a description should be parsed as an "pranswer" indicates that a description MUST be parsed as an answer,
answer, but not a final answer, and so should not result in the but not a final answer, and so MUST NOT result in the freeing of
freeing of allocated resources. It may result in the start of media allocated resources. It may result in the start of media
transmission, if the answer does not specify an inactive media transmission, if the answer does not specify an inactive media
direction. A description used as a "pranswer" may be applied as a direction. A description used as a "pranswer" may be applied as a
response to an "offer", or an update to a previously sent "pranswer". response to an "offer" or as an update to a previously sent
"pranswer".
"answer" indicates that a description should be parsed as an answer, "answer" indicates that a description MUST be parsed as an answer,
the offer-answer exchange should be considered complete, and any the offer/answer exchange MUST be considered complete, and any
resources (decoders, candidates) that are no longer needed can be resources (decoders, candidates) that are no longer needed SHOULD be
released. A description used as an "answer" may be applied as a released. A description used as an "answer" may be applied as a
response to an "offer", or an update to a previously sent "pranswer". response to an "offer" or as an update to a previously sent
"pranswer".
The only difference between a provisional and final answer is that The only difference between a provisional and final answer is that
the final answer results in the freeing of any unused resources that the final answer results in the freeing of any unused resources that
were allocated as a result of the offer. As such, the application were allocated as a result of the offer. As such, the application
can use some discretion on whether an answer should be applied as can use some discretion on whether an answer should be applied as
provisional or final, and can change the type of the session provisional or final and can change the type of the session
description as needed. For example, in a serial forking scenario, an description as needed. For example, in a serial forking scenario, an
application may receive multiple "final" answers, one from each application may receive multiple "final" answers, one from each
remote endpoint. The application could choose to accept the initial remote endpoint. The application could choose to accept the initial
answers as provisional answers, and only apply an answer as final answers as provisional answers and only apply an answer as final when
when it receives one that meets its criteria (e.g. a live user it receives one that meets its criteria (e.g., a live user instead of
instead of voicemail). voicemail).
"rollback" is a special session description type implying that the "rollback" is a special session description type indicating that the
state machine should be rolled back to the previous stable state, as state machine MUST be rolled back to the previous "stable" state, as
described in Section 4.1.8.2. The contents MUST be empty. described in Section 4.1.10.2. The contents MUST be empty.
4.1.8.1. Use of Provisional Answers 4.1.10.1. Use of Provisional Answers
Most applications will not need to create answers using the Most applications will not need to create answers using the
"pranswer" type. While it is good practice to send an immediate "pranswer" type. While it is good practice to send an immediate
response to an offer, in order to warm up the session transport and response to an offer, in order to warm up the session transport and
prevent media clipping, the preferred handling for a JSEP application prevent media clipping, the preferred handling for a JSEP application
is to create and send a "sendonly" final answer with a null is to create and send a "sendonly" final answer with a null
MediaStreamTrack immediately after receiving the offer, which will MediaStreamTrack immediately after receiving the offer, which will
prevent media from being sent by the caller, and allow media to be prevent media from being sent by the caller and allow media to be
sent immediately upon answer by the callee. Later, when the callee sent immediately upon answer by the callee. Later, when the callee
actually accepts the call, the application can plug in the real actually accepts the call, the application can plug in the real
MediaStreamTrack and create a new "sendrecv" offer to update the MediaStreamTrack and create a new "sendrecv" offer to update the
previous offer/answer pair and start bidirectional media flow. While previous offer/answer pair and start bidirectional media flow. While
this could also be done with a "sendonly" pranswer, followed by a this could also be done with a "sendonly" pranswer followed by a
"sendrecv" answer, the initial pranswer leaves the offer-answer "sendrecv" answer, the initial pranswer leaves the offer/answer
exchange open, which means that the caller cannot send an updated exchange open, which means that the caller cannot send an updated
offer during this time. offer during this time.
As an example, consider a typical JSEP application that wants to set As an example, consider a typical JSEP application that wants to set
up audio and video as quickly as possible. When the callee receives up audio and video as quickly as possible. When the callee receives
an offer with audio and video MediaStreamTracks, it will send an an offer with audio and video MediaStreamTracks, it will send an
immediate answer accepting these tracks as sendonly (meaning that the immediate answer accepting these tracks as sendonly (meaning that the
caller will not send the callee any media yet, and because the callee caller will not send the callee any media yet, and because the callee
has not yet added its own MediaStreamTracks, the callee will not send has not yet added its own MediaStreamTracks, the callee will not send
any media either). It will then ask the user to accept the call and any media either). It will then ask the user to accept the call and
acquire the needed local tracks. Upon acceptance by the user, the acquire the needed local tracks. Upon acceptance by the user, the
application will plug in the tracks it has acquired, which, because application will plug in the tracks it has acquired, which, because
ICE and DTLS handshaking have likely completed by this point, can ICE handshaking and DTLS handshaking have likely completed by this
start transmitting immediately. The application will also send a new point, can start transmitting immediately. The application will also
offer to the remote side indicating call acceptance and moving the send a new offer to the remote side indicating call acceptance and
audio and video to be two-way media. A detailed example flow along moving the audio and video to be two-way media. A detailed example
these lines is shown in Section 7.3. flow along these lines is shown in Section 7.3.
Of course, some applications may not be able to perform this double Of course, some applications may not be able to perform this double
offer-answer exchange, particularly ones that are attempting to offer/answer exchange, particularly ones that are attempting to
gateway to legacy signaling protocols. In these cases, pranswer can gateway to legacy signaling protocols. In these cases, pranswer can
still provide the application with a mechanism to warm up the still provide the application with a mechanism to warm up the
transport. transport.
4.1.8.2. Rollback 4.1.10.2. Rollback
In certain situations it may be desirable to "undo" a change made to In certain situations, it may be desirable to "undo" a change made to
setLocalDescription or setRemoteDescription. Consider a case where a setLocalDescription or setRemoteDescription. Consider a case where a
call is ongoing, and one side wants to change some of the session call is ongoing and one side wants to change some of the session
parameters; that side generates an updated offer and then calls parameters; that side generates an updated offer and then calls
setLocalDescription. However, the remote side, either before or setLocalDescription. However, the remote side, either before or
after setRemoteDescription, decides it does not want to accept the after setRemoteDescription, decides it does not want to accept the
new parameters, and sends a reject message back to the offerer. Now, new parameters and sends a reject message back to the offerer. Now,
the offerer, and possibly the answerer as well, need to return to a the offerer, and possibly the answerer as well, needs to return to a
stable state and the previous local/remote description. To support "stable" state and the previous local/remote description. To support
this, we introduce the concept of "rollback", which discards any this, we introduce the concept of "rollback", which discards any
proposed changes to the session, returning the state machine to the proposed changes to the session, returning the state machine to the
stable state. A rollback is performed by supplying a session "stable" state. A rollback is performed by supplying a session
description of type "rollback" with empty contents to either description of type "rollback" with empty contents to either
setLocalDescription or setRemoteDescription. setLocalDescription or setRemoteDescription.
4.1.9. setLocalDescription 4.1.11. setLocalDescription
The setLocalDescription method instructs the PeerConnection to apply The setLocalDescription method instructs the PeerConnection to apply
the supplied session description as its local configuration. The the supplied session description as its local configuration. The
type field indicates whether the description should be processed as type field indicates whether the description should be processed as
an offer, provisional answer, final answer, or rollback; offers and an offer, provisional answer, final answer, or rollback; offers and
answers are checked differently, using the various rules that exist answers are checked differently, using the various rules that exist
for each SDP line. for each SDP line.
This API changes the local media state; among other things, it sets This API changes the local media state; among other things, it sets
up local resources for receiving and decoding media. In order to up local resources for receiving and decoding media. In order to
successfully handle scenarios where the application wants to offer to successfully handle scenarios where the application wants to offer to
change from one media format to a different, incompatible format, the change from one media format to a different, incompatible format, the
PeerConnection must be able to simultaneously support use of both the PeerConnection MUST be able to simultaneously support use of both the
current and pending local descriptions (e.g., support the codecs that current and pending local descriptions (e.g., support the codecs that
exist in either description). This dual processing begins when the exist in either description). This dual processing begins when the
PeerConnection enters the "have-local-offer" state, and continues PeerConnection enters the "have-local-offer" state, and it continues
until setRemoteDescription is called with either a final answer, at until setRemoteDescription is called with either (1) a final answer,
which point the PeerConnection can fully adopt the pending local at which point the PeerConnection can fully adopt the pending local
description, or a rollback, which results in a revert to the current description or (2) a rollback, which results in a revert to the
local description. current local description.
This API indirectly controls the candidate gathering process. When a This API indirectly controls the candidate gathering process. When a
local description is supplied, and the number of transports currently local description is supplied and the number of transports currently
in use does not match the number of transports needed by the local in use does not match the number of transports needed by the local
description, the PeerConnection will create transports as needed and description, the PeerConnection will create transports as needed and
begin gathering candidates for each transport, using ones from the begin gathering candidates for each transport, using ones from the
candidate pool if available. candidate pool if available.
If setRemoteDescription was previously called with an offer, and If (1) setRemoteDescription was previously called with an offer, (2)
setLocalDescription is called with an answer (provisional or final), setLocalDescription is called with an answer (provisional or final),
and the media directions are compatible, and media is available to (3) the media directions are compatible, and (4) media is available
send, this will result in the starting of media transmission. to send, this will result in the starting of media transmission.
4.1.10. setRemoteDescription 4.1.12. setRemoteDescription
The setRemoteDescription method instructs the PeerConnection to apply The setRemoteDescription method instructs the PeerConnection to apply
the supplied session description as the desired remote configuration. the supplied session description as the desired remote configuration.
As in setLocalDescription, the type field of the description As in setLocalDescription, the type field of the description
indicates how it should be processed. indicates how it should be processed.
This API changes the local media state; among other things, it sets This API changes the local media state; among other things, it sets
up local resources for sending and encoding media. up local resources for sending and encoding media.
If setLocalDescription was previously called with an offer, and If (1) setLocalDescription was previously called with an offer, (2)
setRemoteDescription is called with an answer (provisional or final), setRemoteDescription is called with an answer (provisional or final),
and the media directions are compatible, and media is available to (3) the media directions are compatible, and (4) media is available
send, this will result in the starting of media transmission. to send, this will result in the starting of media transmission.
4.1.11. currentLocalDescription 4.1.13. currentLocalDescription
The currentLocalDescription method returns the current negotiated The currentLocalDescription method returns the current negotiated
local description - i.e., the local description from the last local description -- i.e., the local description from the last
successful offer/answer exchange - in addition to any local successful offer/answer exchange -- in addition to any local
candidates that have been generated by the ICE agent since the local candidates that have been generated by the ICE agent since the local
description was set. description was set.
A null object will be returned if an offer/answer exchange has not A null object will be returned if an offer/answer exchange has not
yet been completed. yet been completed.
4.1.12. pendingLocalDescription 4.1.14. pendingLocalDescription
The pendingLocalDescription method returns a copy of the local The pendingLocalDescription method returns a copy of the local
description currently in negotiation - i.e., a local offer set description currently in negotiation -- i.e., a local offer set
without any corresponding remote answer - in addition to any local without any corresponding remote answer -- in addition to any local
candidates that have been generated by the ICE agent since the local candidates that have been generated by the ICE agent since the local
description was set. description was set.
A null object will be returned if the state of the PeerConnection is A null object will be returned if the state of the PeerConnection is
"stable" or "have-remote-offer". "stable" or "have-remote-offer".
4.1.13. currentRemoteDescription 4.1.15. currentRemoteDescription
The currentRemoteDescription method returns a copy of the current The currentRemoteDescription method returns a copy of the current
negotiated remote description - i.e., the remote description from the negotiated remote description -- i.e., the remote description from
last successful offer/answer exchange - in addition to any remote the last successful offer/answer exchange -- in addition to any
candidates that have been supplied via processIceMessage since the remote candidates that have been supplied via processIceMessage since
remote description was set. the remote description was set.
A null object will be returned if an offer/answer exchange has not A null object will be returned if an offer/answer exchange has not
yet been completed. yet been completed.
4.1.14. pendingRemoteDescription 4.1.16. pendingRemoteDescription
The pendingRemoteDescription method returns a copy of the remote The pendingRemoteDescription method returns a copy of the remote
description currently in negotiation - i.e., a remote offer set description currently in negotiation -- i.e., a remote offer set
without any corresponding local answer - in addition to any remote without any corresponding local answer -- in addition to any remote
candidates that have been supplied via processIceMessage since the candidates that have been supplied via processIceMessage since the
remote description was set. remote description was set.
A null object will be returned if the state of the PeerConnection is A null object will be returned if the state of the PeerConnection is
"stable" or "have-local-offer". "stable" or "have-local-offer".
4.1.15. canTrickleIceCandidates 4.1.17. canTrickleIceCandidates
The canTrickleIceCandidates property indicates whether the remote The canTrickleIceCandidates property indicates whether the remote
side supports receiving trickled candidates. There are three side supports receiving trickled candidates. There are three
potential values: potential values:
null: No SDP has been received from the other side, so it is not null: No SDP has been received from the other side, so it is not
known if it can handle trickle. This is the initial value before known if it can handle trickle. This is the initial value before
setRemoteDescription() is called. setRemoteDescription is called.
true: SDP has been received from the other side indicating that it true: SDP has been received from the other side indicating that it
can support trickle. can support trickle.
false: SDP has been received from the other side indicating that it false: SDP has been received from the other side indicating that it
cannot support trickle. cannot support trickle.
As described in Section 3.5.2, JSEP implementations always provide As described in Section 3.5.2, JSEP implementations always provide
candidates to the application individually, consistent with what is candidates to the application individually, consistent with what is
needed for Trickle ICE. However, applications can use the needed for Trickle ICE. However, applications can use the
canTrickleIceCandidates property to determine whether their peer can canTrickleIceCandidates property to determine whether their peer can
actually do Trickle ICE, i.e., whether it is safe to send an initial actually do Trickle ICE, i.e., whether it is safe to send an initial
offer or answer followed later by candidates as they are gathered. offer or answer followed later by candidates as they are gathered.
As "true" is the only value that definitively indicates remote As "true" is the only value that definitively indicates remote
Trickle ICE support, an application which compares Trickle ICE support, an application that compares
canTrickleIceCandidates against "true" will by default attempt Half canTrickleIceCandidates against "true" will by default attempt Half
Trickle on initial offers and Full Trickle on subsequent interactions Trickle on initial offers and Full Trickle on subsequent interactions
with a Trickle ICE-compatible agent. with a Trickle ICE-compatible agent.
4.1.16. setConfiguration 4.1.18. setConfiguration
The setConfiguration method allows the global configuration of the The setConfiguration method allows the global configuration of the
PeerConnection, which was initially set by constructor parameters, to PeerConnection, which was initially set by constructor parameters, to
be changed during the session. The effects of this method call be changed during the session. The effects of calling this method
depend on when it is invoked, and differ depending on which specific depend on when it is invoked, and they will differ, depending on
parameters are changed: which specific parameters are changed:
o Any changes to the STUN/TURN servers to use affect the next * Any changes to the STUN/TURN servers to use affect the next
gathering phase. If an ICE gathering phase has already started or gathering phase. If an ICE gathering phase has already started or
completed, the 'needs-ice-restart' bit mentioned in Section 3.5.1 completed, the 'needs-ice-restart' bit mentioned in Section 3.5.1
will be set. This will cause the next call to createOffer to will be set. This will cause the next call to createOffer to
generate new ICE credentials, for the purpose of forcing an ICE generate new ICE credentials, for the purpose of forcing an ICE
restart and kicking off a new gathering phase, in which the new restart and kicking off a new gathering phase, in which the new
servers will be used. If the ICE candidate pool has a nonzero servers will be used. If the ICE candidate pool has a nonzero
size, and a local description has not yet been applied, any size and a local description has not yet been applied, any
existing candidates will be discarded, and new candidates will be existing candidates will be discarded, and new candidates will be
gathered from the new servers. gathered from the new servers.
o Any change to the ICE candidate policy affects the next gathering * Any change to the ICE candidate policy affects the next gathering
phase. If an ICE gathering phase has already started or phase. If an ICE gathering phase has already started or
completed, the 'needs-ice-restart' bit will be set. Either way, completed, the 'needs-ice-restart' bit will be set. Either way,
changes to the policy have no effect on the candidate pool, changes to the policy have no effect on the candidate pool,
because pooled candidates are not made available to the because pooled candidates are not made available to the
application until a gathering phase occurs, and so any necessary application until a gathering phase occurs, and so any necessary
filtering can still be done on any pooled candidates. filtering can still be done on any pooled candidates.
o The ICE candidate pool size MUST NOT be changed after applying a * The ICE candidate pool size MUST NOT be changed after applying a
local description. If a local description has not yet been local description. If a local description has not yet been
applied, any changes to the ICE candidate pool size take effect applied, any changes to the ICE candidate pool size take effect
immediately; if increased, additional candidates are pre-gathered; immediately; if increased, additional candidates are pre-gathered;
if decreased, the now-superfluous candidates are discarded. if decreased, the now-superfluous candidates are discarded.
o The bundle and RTCP-multiplexing policies MUST NOT be changed * The bundle and RTCP-multiplexing policies MUST NOT be changed
after the construction of the PeerConnection. after the construction of the PeerConnection.
This call may result in a change to the state of the ICE Agent. Calling this method may result in a change to the state of the ICE
agent.
4.1.17. addIceCandidate 4.1.19. addIceCandidate
The addIceCandidate method provides an update to the ICE agent via an The addIceCandidate method provides an update to the ICE agent via an
IceCandidate object Section 3.5.2.1. If the IceCandidate's candidate IceCandidate object (Section 3.5.2.1). If the IceCandidate's
field is filled in, the IceCandidate is treated as a new remote ICE candidate field is non-null, the IceCandidate is treated as a new
candidate, which will be added to the current and/or pending remote remote ICE candidate, which will be added to the current and/or
description according to the rules defined for Trickle ICE. pending remote description according to the rules defined for Trickle
Otherwise, the IceCandidate is treated as an end-of-candidates ICE. Otherwise, the IceCandidate is treated as an end-of-candidates
indication, as defined in [I-D.ietf-ice-trickle]. indication, as defined in [RFC8838], Section 14.
In either case, the m= section index, MID, and ufrag fields from the In either case, the "m=" section index, MID, and ufrag fields from
supplied IceCandidate are used to determine which m= section and ICE the supplied IceCandidate are used to determine which "m=" section
candidate generation the IceCandidate belongs to, as described in and ICE candidate generation the IceCandidate belongs to, as
Section 3.5.2.1 above. In the case of an end-of-candidates described in Section 3.5.2.1 above. In the case of an end-of-
indication, the absence of both the m= section index and MID fields candidates indication, null values for the "m=" section index and MID
is interpreted to mean that the indication applies to all m= sections fields are interpreted to mean that the indication applies to all
in the specified ICE candidate generation. However, if both fields "m=" sections in the specified ICE candidate generation. However, if
are absent for a new remote candidate, this MUST be treated as an both fields are null for a new remote candidate, this MUST be treated
invalid condition, as specified below. as an invalid condition, as specified below.
If any IceCandidate fields contain invalid values, or an error occurs If any IceCandidate fields contain invalid values or an error occurs
during the processing of the IceCandidate object, the supplied during the processing of the IceCandidate object, the supplied
IceCandidate MUST be ignored and an error MUST be returned. IceCandidate MUST be ignored and an error MUST be returned.
Otherwise, the new remote candidate or end-of-candidates indication Otherwise, the new remote candidate or end-of-candidates indication
is supplied to the ICE agent. In the case of a new remote candidate, is supplied to the ICE agent. In the case of a new remote candidate,
connectivity checks will be sent to the new candidate. connectivity checks will be sent to the new candidate, assuming
setLocalDescription has already been called to initialize the ICE
gathering process.
4.1.20. onicecandidate Event
The onicecandidate event is dispatched to the application in two
situations: (1) when the ICE agent has discovered a new allowed local
ICE candidate during ICE gathering, as outlined in Section 3.5.1 and
subject to the restrictions discussed in Section 3.5.3, or (2) when
an ICE gathering phase completes. The event contains a single
IceCandidate object, as defined in Section 3.5.2.1.
In the first case, the newly discovered candidate is reflected in the
IceCandidate object, and all of its fields MUST be non-null. This
candidate will also be added to the current and/or pending local
description according to the rules defined for Trickle ICE.
In the second case, the event's IceCandidate object MUST have its
candidate field set to null to indicate that the current gathering
phase is complete, i.e., there will be no further onicecandidate
events in this phase. However, the IceCandidate's ufrag field MUST
be specified to indicate which ICE candidate generation is ending.
The IceCandidate's "m=" section index and MID fields MAY be specified
to indicate that the event applies to a specific "m=" section, or set
to null to indicate it applies to all "m=" sections in the current
ICE candidate generation. This event can be used by the application
to generate an end-of-candidates indication, as defined in [RFC8838],
Section 13.
4.2. RtpTransceiver 4.2. RtpTransceiver
4.2.1. stop 4.2.1. stop
The stop method stops an RtpTransceiver. This will cause future The stop method stops an RtpTransceiver. This will cause future
calls to createOffer to generate a zero port for the associated m= calls to createOffer to generate a zero port for the associated "m="
section. See below for more details. section. See below for more details.
4.2.2. stopped 4.2.2. stopped
The stopped property indicates whether the transceiver has been The stopped property indicates whether the transceiver has been
stopped, either by a call to stopTransceiver or by applying an answer stopped, either by a call to stop or by applying an answer that
that rejects the associated m= section. In either of these cases, it rejects the associated "m=" section. In either of these cases, it is
is set to "true", and otherwise will be set to "false". set to "true" and otherwise will be set to "false".
A stopped RtpTransceiver does not send any outgoing RTP or RTCP or A stopped RtpTransceiver does not send any outgoing RTP or RTCP or
process any incoming RTP or RTCP. It cannot be restarted. process any incoming RTP or RTCP. It cannot be restarted.
4.2.3. setDirection 4.2.3. setDirection
The setDirection method sets the direction of a transceiver, which The setDirection method sets the direction of a transceiver, which
affects the direction property of the associated m= section on future affects the direction property of the associated "m=" section on
calls to createOffer and createAnswer. The permitted values for future calls to createOffer and createAnswer. The permitted values
direction are "recvonly", "sendrecv", "sendonly", and "inactive", for direction are "recvonly", "sendrecv", "sendonly", and "inactive",
mirroring the identically-named directional attributes defined in mirroring the identically named direction attributes defined in
[RFC4566], Section 6. [RFC4566], Section 6.
When creating offers, the transceiver direction is directly reflected When creating offers, the transceiver direction is directly reflected
in the output, even for re-offers. When creating answers, the in the output, even for re-offers. When creating answers, the
transceiver direction is intersected with the offered direction, as transceiver direction is intersected with the offered direction, as
explained in Section 5.3 below. explained in Section 5.3 below.
Note that while setDirection sets the direction property of the Note that while setDirection sets the direction property of the
transceiver immediately ( Section 4.2.4), this property does not transceiver immediately (Section 4.2.4), this property does not
immediately affect whether the transceiver's RtpSender will send or immediately affect whether the transceiver's RtpSender will send or
its RtpReceiver will receive. The direction in effect is represented its RtpReceiver will receive. The direction in effect is represented
by the currentDirection property, which is only updated when an by the currentDirection property, which is only updated when an
answer is applied. answer is applied.
4.2.4. direction 4.2.4. direction
The direction property indicates the last value passed into The direction property indicates the last value passed into
setDirection. If setDirection has never been called, it is set to setDirection. If setDirection has never been called, it is set to
the direction the transceiver was initialized with. the direction the transceiver was initialized with.
4.2.5. currentDirection 4.2.5. currentDirection
The currentDirection property indicates the last negotiated direction The currentDirection property indicates the last negotiated direction
for the transceiver's associated m= section. More specifically, it for the transceiver's associated "m=" section. More specifically, it
indicates the [RFC3264] directional attribute of the associated m= indicates the direction attribute [RFC3264] of the associated "m="
section in the last applied answer (including provisional answers), section in the last applied answer (including provisional answers),
with "send" and "recv" directions reversed if it was a remote answer. with "send" and "recv" directions reversed if it was a remote answer.
For example, if the directional attribute for the associated m= For example, if the direction attribute for the associated "m="
section in a remote answer is "recvonly", currentDirection is set to section in a remote answer is "recvonly", currentDirection is set to
"sendonly". "sendonly".
If an answer that references this transceiver has not yet been If an answer that references this transceiver has not yet been
applied, or if the transceiver is stopped, currentDirection is set to applied or if the transceiver is stopped, currentDirection is set to
null. "null".
4.2.6. setCodecPreferences 4.2.6. setCodecPreferences
The setCodecPreferences method sets the codec preferences of a The setCodecPreferences method sets the codec preferences of a
transceiver, which in turn affect the presence and order of codecs of transceiver, which in turn affect the presence and order of codecs of
the associated m= section on future calls to createOffer and the associated "m=" section on future calls to createOffer and
createAnswer. Note that setCodecPreferences does not directly affect createAnswer. Note that setCodecPreferences does not directly affect
which codec the implementation decides to send. It only affects which codec the implementation decides to send. It only affects
which codecs the implementation indicates that it prefers to receive, which codecs the implementation indicates that it prefers to receive,
via the offer or answer. Even when a codec is excluded by via the offer or answer. Even when a codec is excluded by
setCodecPreferences, it still may be used to send until the next setCodecPreferences, it still may be used to send until the next
offer/answer exchange discards it. offer/answer exchange discards it.
The codec preferences of an RtpTransceiver can cause codecs to be The codec preferences of an RtpTransceiver can cause codecs to be
excluded by subsequent calls to createOffer and createAnswer, in excluded by subsequent calls to createOffer and createAnswer, in
which case the corresponding media formats in the associated m= which case the corresponding media formats in the associated "m="
section will be excluded. The codec preferences cannot add media section will be excluded. The codec preferences cannot add media
formats that would otherwise not be present. formats that would otherwise not be present.
The codec preferences of an RtpTransceiver can also determine the The codec preferences of an RtpTransceiver can also determine the
order of codecs in subsequent calls to createOffer and createAnswer, order of codecs in subsequent calls to createOffer and createAnswer,
in which case the order of the media formats in the associated m= in which case the order of the media formats in the associated "m="
section will follow the specified preferences. section will follow the specified preferences.
5. SDP Interaction Procedures 5. SDP Interaction Procedures
This section describes the specific procedures to be followed when This section describes the specific procedures to be followed when
creating and parsing SDP objects. creating and parsing SDP objects.
5.1. Requirements Overview 5.1. Requirements Overview
JSEP implementations must comply with the specifications listed below JSEP implementations MUST comply with the specifications listed below
that govern the creation and processing of offers and answers. that govern the creation and processing of offers and answers.
5.1.1. Usage Requirements 5.1.1. Usage Requirements
All session descriptions handled by JSEP implementations, both local All session descriptions handled by JSEP implementations, both local
and remote, MUST indicate support for the following specifications. and remote, MUST indicate support for the following specifications.
If any of these are absent, this omission MUST be treated as an If any of these are absent, this omission MUST be treated as an
error. error.
o ICE, as specified in [RFC8445], MUST be used. Note that the * ICE, as specified in [RFC8445], MUST be used. Note that the
remote endpoint may use a Lite implementation; implementations remote endpoint may use a lite implementation; implementations
MUST properly handle remote endpoints which do ICE-Lite. MUST properly handle remote endpoints that use ICE-lite. The
remote endpoint may also use an older version of ICE;
implementations MUST properly handle remote endpoints that use ICE
as specified in [RFC5245].
o DTLS [RFC6347] or DTLS-SRTP [RFC5763], MUST be used, as * DTLS [RFC6347] or DTLS-SRTP [RFC5763] MUST be used, as appropriate
appropriate for the media type, as specified in for the media type, as specified in [RFC8827].
[I-D.ietf-rtcweb-security-arch]
The SDES SRTP keying mechanism from [RFC4568] MUST NOT be used, as The SDP security descriptions mechanism for SRTP keying [RFC4568]
discussed in [I-D.ietf-rtcweb-security-arch]. MUST NOT be used, as discussed in [RFC8827].
5.1.2. Profile Names and Interoperability 5.1.2. Profile Names and Interoperability
For media m= sections, JSEP implementations MUST support the For media "m=" sections, JSEP implementations MUST support the
"UDP/TLS/RTP/SAVPF" profile specified in [RFC5764] as well as the "UDP/TLS/RTP/SAVPF" profile specified in [RFC5764] as well as the
"TCP/DTLS/RTP/SAVPF" profile specified in [RFC7850], and MUST "TCP/DTLS/RTP/SAVPF" profile specified in [RFC7850] and MUST indicate
indicate one of these profiles for each media m= line they produce in one of these profiles for each media "m=" line they produce in an
an offer. For data m= sections, implementations MUST support the offer. For data "m=" sections, implementations MUST support the
"UDP/DTLS/SCTP" profile as well as the "TCP/DTLS/SCTP" profile, and "UDP/DTLS/SCTP" profile as well as the "TCP/DTLS/SCTP" profile and
MUST indicate one of these profiles for each data m= line they MUST indicate one of these profiles for each data "m=" line they
produce in an offer. The exact profile to use is determined by the produce in an offer. The exact profile to use is determined by the
protocol associated with the current default or selected ICE protocol associated with the current default or selected ICE
candidate, as described in [I-D.ietf-mmusic-ice-sip-sdp], candidate, as described in [RFC8839], Section 4.2.1.2.
Section 3.2.1.2.
Unfortunately, in an attempt at compatibility, some endpoints Unfortunately, in an attempt at compatibility, some endpoints
generate other profile strings even when they mean to support one of generate other profile strings even when they mean to support one of
these profiles. For instance, an endpoint might generate "RTP/AVP" these profiles. For instance, an endpoint might generate "RTP/AVP"
but supply "a=fingerprint" and "a=rtcp-fb" attributes, indicating its but supply "a=fingerprint" and "a=rtcp-fb" attributes, indicating its
willingness to support "UDP/TLS/RTP/SAVPF" or "TCP/DTLS/RTP/SAVPF". willingness to support "UDP/TLS/RTP/SAVPF" or "TCP/DTLS/RTP/SAVPF".
In order to simplify compatibility with such endpoints, JSEP In order to simplify compatibility with such endpoints, JSEP
implementations MUST follow the following rules when processing the implementations MUST follow the following rules when processing the
media m= sections in a received offer: media "m=" sections in a received offer:
o Any profile in the offer matching one of the following MUST be * Any profile in the offer matching one of the following MUST be
accepted: accepted:
* "RTP/AVP" (Defined in [RFC4566], Section 8.2.2) - "RTP/AVP" (defined in [RFC4566], Section 8.2.2)
* "RTP/AVPF" (Defined in [RFC4585], Section 9) - "RTP/AVPF" (defined in [RFC4585], Section 9)
* "RTP/SAVP" (Defined in [RFC3711], Section 12) - "RTP/SAVP" (defined in [RFC3711], Section 12)
* "RTP/SAVPF" (Defined in [RFC5124], Section 6) - "RTP/SAVPF" (defined in [RFC5124], Section 6)
* "TCP/DTLS/RTP/SAVP" (Defined in [RFC7850], Section 3.4) - "TCP/DTLS/RTP/SAVP" (defined in [RFC7850], Section 3.4)
* "TCP/DTLS/RTP/SAVPF" (Defined in [RFC7850], Section 3.5) - "TCP/DTLS/RTP/SAVPF" (defined in [RFC7850], Section 3.5)
* "UDP/TLS/RTP/SAVP" (Defined in [RFC5764], Section 9) - "UDP/TLS/RTP/SAVP" (defined in [RFC5764], Section 9)
* "UDP/TLS/RTP/SAVPF" (Defined in [RFC5764], Section 9) - "UDP/TLS/RTP/SAVPF" (defined in [RFC5764], Section 9)
o The profile in any "m=" line in any generated answer MUST exactly * The profile in any "m=" line in any generated answer MUST exactly
match the profile provided in the offer. match the profile provided in the offer.
o Because DTLS-SRTP is REQUIRED, the choice of SAVP or AVP has no * Because DTLS-SRTP is REQUIRED, the choice of SAVP or AVP has no
effect; support for DTLS-SRTP is determined by the presence of one effect; support for DTLS-SRTP is determined by the presence of one
or more "a=fingerprint" attribute. Note that lack of an or more "a=fingerprint" attributes. Note that lack of an
"a=fingerprint" attribute will lead to negotiation failure. "a=fingerprint" attribute will lead to negotiation failure.
o The use of AVPF or AVP simply controls the timing rules used for * The use of AVPF or AVP simply controls the timing rules used for
RTCP feedback. If AVPF is provided, or an "a=rtcp-fb" attribute RTCP feedback. If AVPF is provided or an "a=rtcp-fb" attribute is
is present, assume AVPF timing, i.e., a default value of "trr- present, assume AVPF timing, i.e., a default value of "trr-int=0".
int=0". Otherwise, assume that AVPF is being used in an AVP Otherwise, assume that AVPF is being used in an AVP-compatible
compatible mode and use a value of "trr-int=4000". mode and use a value of "trr-int=4000".
o For data m= sections, implementations MUST support receiving the * For data "m=" sections, implementations MUST support receiving the
"UDP/DTLS/SCTP", "TCP/DTLS/SCTP", or "DTLS/SCTP" (for backwards "UDP/DTLS/SCTP", "TCP/DTLS/SCTP", or "DTLS/SCTP" (for backwards
compatibility) profiles. compatibility) profiles.
Note that re-offers by JSEP implementations MUST use the correct Note that re-offers by JSEP implementations MUST use the correct
profile strings even if the initial offer/answer exchange used an profile strings even if the initial offer/answer exchange used an
(incorrect) older profile string. This simplifies JSEP behavior, (incorrect) older profile string. This simplifies JSEP behavior,
with minimal downside, as any remote endpoint that fails to handle with minimal downside, as any remote endpoint that fails to handle
such a re-offer will also fail to handle a JSEP endpoint's initial such a re-offer will also fail to handle a JSEP endpoint's initial
offer. offer.
5.2. Constructing an Offer 5.2. Constructing an Offer
When createOffer is called, a new SDP description must be created When createOffer is called, a new SDP description MUST be created
that includes the functionality specified in that includes the functionality specified in [RFC8834]. The exact
[I-D.ietf-rtcweb-rtp-usage]. The exact details of this process are details of this process are explained below.
explained below.
5.2.1. Initial Offers 5.2.1. Initial Offers
When createOffer is called for the first time, the result is known as When createOffer is called for the first time, the result is known as
the initial offer. the initial offer.
The first step in generating an initial offer is to generate session- The first step in generating an initial offer is to generate session-
level attributes, as specified in [RFC4566], Section 5. level attributes, as specified in [RFC4566], Section 5.
Specifically: Specifically:
o The first SDP line MUST be "v=0", as specified in [RFC4566], * The first SDP line MUST be "v=0" as defined in [RFC4566],
Section 5.1 Section 5.1.
o The second SDP line MUST be an "o=" line, as specified in * The second SDP line MUST be an "o=" line as defined in [RFC4566],
[RFC4566], Section 5.2. The value of the <username> field SHOULD Section 5.2. The value of the <username> field SHOULD be "-".
be "-". The sess-id MUST be representable by a 64-bit signed The <sess-id> MUST be representable by a 64-bit signed integer,
integer, and the value MUST be less than (2**63)-1. It is and the value MUST be less than 2^(63)-1. It is RECOMMENDED that
RECOMMENDED that the sess-id be constructed by generating a 64-bit the <sess-id> be constructed by generating a 64-bit quantity with
quantity with the highest bit set to zero and the remaining 63 the highest bit set to zero and the remaining 63 bits being
bits being cryptographically random. The value of the <nettype> cryptographically random. The value of the <nettype> <addrtype>
<addrtype> <unicast-address> tuple SHOULD be set to a non- <unicast-address> tuple SHOULD be set to a non-meaningful address,
meaningful address, such as IN IP4 0.0.0.0, to prevent leaking a such as IN IP4 0.0.0.0, to prevent leaking a local IP address in
local IP address in this field; this problem is discussed in this field; this problem is discussed in [RFC8828]. As mentioned
[I-D.ietf-rtcweb-ip-handling]. As mentioned in [RFC4566], the in [RFC4566], the entire "o=" line needs to be unique, but
entire o= line needs to be unique, but selecting a random number selecting a random number for <sess-id> is sufficient to
for <sess-id> is sufficient to accomplish this. accomplish this.
o The third SDP line MUST be a "s=" line, as specified in [RFC4566], * The third SDP line MUST be a "s=" line as defined in [RFC4566],
Section 5.3; to match the "o=" line, a single dash SHOULD be used Section 5.3; to match the "o=" line, a single dash SHOULD be used
as the session name, e.g. "s=-". Note that this differs from the as the session name, e.g., "s=-". Note that this differs from the
advice in [RFC4566] which proposes a single space, but as both advice in [RFC4566], which proposes a single space, but as both
"o=" and "s=" are meaningless in JSEP, having the same meaningless "o=" and "s=" are meaningless in JSEP, having the same meaningless
value seems clearer. value seems clearer.
o Session Information ("i="), URI ("u="), Email Address ("e="), * Session Information ("i="), URI ("u="), Email Address ("e="),
Phone Number ("p="), Repeat Times ("r="), and Time Zones ("z=") Phone Number ("p="), Repeat Times ("r="), and Time Zones ("z=")
lines are not useful in this context and SHOULD NOT be included. lines are not useful in this context and SHOULD NOT be included.
o Encryption Keys ("k=") lines do not provide sufficient security * Encryption Keys ("k=") lines do not provide sufficient security
and MUST NOT be included. and MUST NOT be included.
o A "t=" line MUST be added, as specified in [RFC4566], Section 5.9; * A "t=" line MUST be added, as specified in [RFC4566], Section 5.9;
both <start-time> and <stop-time> SHOULD be set to zero, e.g. "t=0 both <start-time> and <stop-time> SHOULD be set to zero, e.g.,
0". "t=0 0".
o An "a=ice-options" line with the "trickle" and "ice2" options MUST * An "a=ice-options" line with the "trickle" and "ice2" options MUST
be added, as specified in [I-D.ietf-ice-trickle], Section 3 and be added, as specified in [RFC8840], Section 4.1.1 and [RFC8445],
[RFC8445], Section 10. Section 10.
o If WebRTC identity is being used, an "a=identity" line as * If WebRTC identity is being used, an "a=identity" line MUST be
described in [I-D.ietf-rtcweb-security-arch], Section 5. added, as described in [RFC8827], Section 5.
The next step is to generate m= sections, as specified in [RFC4566], The next step is to generate "m=" sections, as specified in
Section 5.14. An m= section is generated for each RtpTransceiver [RFC4566], Section 5.14. An "m=" section is generated for each
that has been added to the PeerConnection, excluding any stopped RtpTransceiver that has been added to the PeerConnection, excluding
RtpTransceivers; this is done in the order the RtpTransceivers were any stopped RtpTransceivers; this is done in the order the
added to the PeerConnection. If there are no such RtpTransceivers, RtpTransceivers were added to the PeerConnection. If there are no
no m= sections are generated; more can be added later, as discussed such RtpTransceivers, no "m=" sections are generated; more can be
in [RFC3264], Section 5. added later, as discussed in [RFC3264], Section 5.
For each m= section generated for an RtpTransceiver, establish a For each "m=" section generated for an RtpTransceiver, establish a
mapping between the transceiver and the index of the generated m= mapping between the transceiver and the index of the generated "m="
section. section.
Each m= section, provided it is not marked as bundle-only, MUST Each "m=" section, provided it is not marked as bundle-only, MUST
generate a unique set of ICE credentials and gather its own unique contain a unique set of ICE credentials and a unique set of ICE
set of ICE candidates. Bundle-only m= sections MUST NOT contain any candidates. Bundle-only "m=" sections MUST NOT contain any ICE
ICE credentials and MUST NOT gather any candidates. credentials and MUST NOT gather any candidates.
For DTLS, all m= sections MUST use all the certificate(s) that have For DTLS, all "m=" sections MUST use any and all certificates that
been specified for the PeerConnection; as a result, they MUST all have been specified for the PeerConnection; as a result, they MUST
have the same [RFC8122] fingerprint value(s), or these value(s) MUST all have the same fingerprint value or values [RFC8122], or these
be session-level attributes. values MUST be session-level attributes.
Each m= section should be generated as specified in [RFC4566], Each "m=" section MUST be generated as specified in [RFC4566],
Section 5.14. For the m= line itself, the following rules MUST be Section 5.14. For the "m=" line itself, the following rules MUST be
followed: followed:
o If the m= section is marked as bundle-only, then the port value * If the "m=" section is marked as bundle-only, then the <port>
MUST be set to 0. Otherwise, the port value is set to the port of value MUST be set to zero. Otherwise, the <port> value is set to
the default ICE candidate for this m= section, but given that no the port of the default ICE candidate for this "m=" section, but
candidates are available yet, the "dummy" port value of 9 given that no candidates are available yet, the default port value
(Discard) MUST be used, as indicated in [I-D.ietf-ice-trickle], of 9 (Discard) MUST be used, as indicated in [RFC8840],
Section 5.1. Section 4.1.1.
o To properly indicate use of DTLS, the <proto> field MUST be set to * To properly indicate use of DTLS, the <proto> field MUST be set to
"UDP/TLS/RTP/SAVPF", as specified in [RFC5764], Section 8. "UDP/TLS/RTP/SAVPF", as specified in [RFC5764], Section 8.
o If codec preferences have been set for the associated transceiver, * If codec preferences have been set for the associated transceiver,
media formats MUST be generated in the corresponding order, and media formats MUST be generated in the corresponding order and
MUST exclude any codecs not present in the codec preferences. MUST exclude any codecs not present in the codec preferences.
o Unless excluded by the above restrictions, the media formats MUST * Unless excluded by the above restrictions, the media formats MUST
include the mandatory audio/video codecs as specified in include the mandatory audio/video codecs as specified in
[RFC7874], Section 3, and [RFC7742], Section 5. [RFC7874], Section 3 and [RFC7742], Section 5.
The m= line MUST be followed immediately by a "c=" line, as specified The "m=" line MUST be followed immediately by a "c=" line, as
in [RFC4566], Section 5.7. Again, as no candidates are available specified in [RFC4566], Section 5.7. Again, as no candidates are
yet, the "c=" line must contain the "dummy" value "IN IP4 0.0.0.0", available yet, the "c=" line MUST contain the default value "IN IP4
as defined in [I-D.ietf-ice-trickle], Section 5.1. 0.0.0.0", as defined in [RFC8840], Section 4.1.1.
[I-D.ietf-mmusic-sdp-mux-attributes] groups SDP attributes into [RFC8859] groups SDP attributes into different categories. To avoid
different categories. To avoid unnecessary duplication when unnecessary duplication when bundling, attributes of category
bundling, attributes of category IDENTICAL or TRANSPORT MUST NOT be IDENTICAL or TRANSPORT MUST NOT be repeated in bundled "m=" sections,
repeated in bundled m= sections, repeating the guidance from repeating the guidance from [RFC8843], Section 7.1.3. This includes
[I-D.ietf-mmusic-sdp-bundle-negotiation], Section 8.1. This includes "m=" sections for which bundling has been negotiated and is still
m= sections for which bundling has been negotiated and is still desired, as well as "m=" sections marked as bundle-only.
desired, as well as m= sections marked as bundle-only.
The following attributes, which are of a category other than The following attributes, which are of a category other than
IDENTICAL or TRANSPORT, MUST be included in each m= section: IDENTICAL or TRANSPORT, MUST be included in each "m=" section:
o An "a=mid" line, as specified in [RFC5888], Section 4. All MID * An "a=mid" line, as specified in [RFC5888], Section 4. All MID
values MUST be generated in a fashion that does not leak user values MUST be generated in a fashion that does not leak user
information, e.g., randomly or using a per-PeerConnection counter, information, e.g., randomly or using a per-PeerConnection counter,
and SHOULD be 3 bytes or less, to allow them to efficiently fit and SHOULD be 3 bytes or less, to allow them to efficiently fit
into the RTP header extension defined in into the RTP header extension defined in [RFC8843], Section 15.2.
[I-D.ietf-mmusic-sdp-bundle-negotiation], Section 14. Note that Note that this does not set the RtpTransceiver mid property, as
this does not set the RtpTransceiver mid property, as that only that only occurs when the description is applied. The generated
occurs when the description is applied. The generated MID value MID value can be considered a "proposed" MID at this point.
can be considered a "proposed" MID at this point.
o A direction attribute which is the same as that of the associated * A direction attribute that is the same as that of the associated
transceiver. transceiver.
o For each media format on the m= line, "a=rtpmap" and "a=fmtp" * For each media format on the "m=" line, "a=rtpmap" and "a=fmtp"
lines, as specified in [RFC4566], Section 6, and [RFC3264], lines, as specified in [RFC4566], Section 6 and [RFC3264],
Section 5.1. Section 5.1.
o For each primary codec where RTP retransmission should be used, a * For each primary codec where RTP retransmission should be used, a
corresponding "a=rtpmap" line indicating "rtx" with the clock rate corresponding "a=rtpmap" line indicating "rtx" with the clock rate
of the primary codec and an "a=fmtp" line that references the of the primary codec and an "a=fmtp" line that references the
payload type of the primary codec, as specified in [RFC4588], payload type of the primary codec, as specified in [RFC4588],
Section 8.1. Section 8.1.
o For each supported FEC mechanism, "a=rtpmap" and "a=fmtp" lines, * For each supported Forward Error Correction (FEC) mechanism,
as specified in [RFC4566], Section 6. The FEC mechanisms that "a=rtpmap" and "a=fmtp" lines, as specified in [RFC4566],
MUST be supported are specified in [I-D.ietf-rtcweb-fec], Section 6. The FEC mechanisms that MUST be supported are
Section 6, and specific usage for each media type is outlined in specified in [RFC8854], Section 7, and specific usage for each
Sections 4 and 5. media type is outlined in Sections 4 and 5.
o If this m= section is for media with configurable durations of * If this "m=" section is for media with configurable durations of
media per packet, e.g., audio, an "a=maxptime" line, indicating media per packet, e.g., audio, an "a=maxptime" line, indicating
the maximum amount of media, specified in milliseconds, that can the maximum amount of media, specified in milliseconds, that can
be encapsulated in each packet, as specified in [RFC4566], be encapsulated in each packet, as specified in [RFC4566],
Section 6. This value is set to the smallest of the maximum Section 6. This value is set to the smallest of the maximum
duration values across all the codecs included in the m= section. duration values across all the codecs included in the "m="
section.
o If this m= section is for video media, and there are known * If this "m=" section is for video media and there are known
limitations on the size of images which can be decoded, an limitations on the size of images that can be decoded, an
"a=imageattr" line, as specified in Section 3.6. "a=imageattr" line, as specified in Section 3.6.
o For each supported RTP header extension, an "a=extmap" line, as * For each supported RTP header extension, an "a=extmap" line, as
specified in [RFC5285], Section 5. The list of header extensions specified in [RFC5285], Section 5. The list of header extensions
that SHOULD/MUST be supported is specified in that SHOULD/MUST be supported is specified in [RFC8834],
[I-D.ietf-rtcweb-rtp-usage], Section 5.2. Any header extensions Section 5.2. Any header extensions that require encryption MUST
that require encryption MUST be specified as indicated in be specified as indicated in [RFC6904], Section 4.
[RFC6904], Section 4.
o For each supported RTCP feedback mechanism, an "a=rtcp-fb" line, * For each supported RTCP feedback mechanism, an "a=rtcp-fb" line,
as specified in [RFC4585], Section 4.2. The list of RTCP feedback as specified in [RFC4585], Section 4.2. The list of RTCP feedback
mechanisms that SHOULD/MUST be supported is specified in mechanisms that SHOULD/MUST be supported is specified in
[I-D.ietf-rtcweb-rtp-usage], Section 5.1. [RFC8834], Section 5.1.
o If the RtpTransceiver has a sendrecv or sendonly direction: * If the RtpTransceiver has a sendrecv or sendonly direction:
* For each MediaStream that was associated with the transceiver - For each MediaStream that was associated with the transceiver
when it was created via addTrack or addTransceiver, an "a=msid" when it was created via addTrack or addTransceiver, an "a=msid"
line, as specified in [I-D.ietf-mmusic-msid], Section 2, but line, as specified in [RFC8830], Section 2, but omitting the
omitting the "appdata" field. "appdata" field.
o If the RtpTransceiver has a sendrecv or sendonly direction, and * If the RtpTransceiver has a sendrecv or sendonly direction, and
the application has specified RID values or has specified more the application has specified a rid-id for an encoding, or has
than one encoding in the RtpSenders's parameters, an "a=rid" line specified more than one encoding in the RtpSenders's parameters,
for each encoding specified. The "a=rid" line is specified in an "a=rid" line for each encoding specified. The "a=rid" line is
[I-D.ietf-mmusic-rid], and its direction MUST be "send". If the specified in [RFC8851], and its direction MUST be "send". If the
application has chosen a RID value, it MUST be used as the rid- application has chosen a rid-id, it MUST be used; otherwise, a
identifier; otherwise a RID value MUST be generated by the rid-id MUST be generated by the implementation. rid-ids MUST be
implementation. RID values MUST be generated in a fashion that generated in a fashion that does not leak user information, e.g.,
does not leak user information, e.g., randomly or using a per- randomly or using a per-PeerConnection counter (see guidance at
PeerConnection counter, and SHOULD be 3 bytes or less, to allow the end of [RFC8852], Section 3.3), and SHOULD be 3 bytes or less,
them to efficiently fit into the RTP header extension defined in to allow them to efficiently fit into the RTP header extensions
[I-D.ietf-avtext-rid], Section 3. If no encodings have been defined in [RFC8852], Section 3.3. If no encodings have been
specified, or only one encoding is specified but without a RID specified, or only one encoding is specified but without a rid-id,
value, then no "a=rid" lines are generated. then no "a=rid" lines are generated.
o If the RtpTransceiver has a sendrecv or sendonly direction and * If the RtpTransceiver has a sendrecv or sendonly direction and
more than one "a=rid" line has been generated, an "a=simulcast" more than one "a=rid" line has been generated, an "a=simulcast"
line, with direction "send", as defined in line, with direction "send", as defined in [RFC8853], Section 5.1.
[I-D.ietf-mmusic-sdp-simulcast], Section 6.2. The list of RIDs The associated set of rid-ids MUST include all of the rid-ids used
MUST include all of the RID identifiers used in the "a=rid" lines in the "a=rid" lines for this "m=" section.
for this m= section.
o If the bundle policy for this PeerConnection is set to "max- * If (1) the bundle policy for this PeerConnection is set to "max-
bundle", and this is not the first m= section, or the bundle bundle" and this is not the first "m=" section or (2) the bundle
policy is set to "balanced", and this is not the first m= section policy is set to "balanced" and this is not the first "m=" section
for this media type, an "a=bundle-only" line. for this media type, an "a=bundle-only" line.
The following attributes, which are of category IDENTICAL or The following attributes, which are of category IDENTICAL or
TRANSPORT, MUST appear only in "m=" sections which either have a TRANSPORT, MUST appear only in "m=" sections that either have a
unique address or which are associated with the bundle-tag. (In unique address or are associated with the BUNDLE-tag. (In initial
initial offers, this means those "m=" sections which do not contain offers, this means those "m=" sections that do not contain an
an "a=bundle-only" attribute.) "a=bundle-only" attribute.)
o "a=ice-ufrag" and "a=ice-pwd" lines, as specified in * "a=ice-ufrag" and "a=ice-pwd" lines, as specified in [RFC8839],
[I-D.ietf-mmusic-ice-sip-sdp], Section 4.4. Section 5.4.
o For each desired digest algorithm, one or more "a=fingerprint" * For each desired digest algorithm, one or more "a=fingerprint"
lines for each of the endpoint's certificates, as specified in lines for each of the endpoint's certificates, as specified in
[RFC8122], Section 5. [RFC8122], Section 5.
o An "a=setup" line, as specified in [RFC4145], Section 4, and * An "a=setup" line, as specified in [RFC4145], Section 4 and
clarified for use in DTLS-SRTP scenarios in [RFC5763], Section 5. clarified for use in DTLS-SRTP scenarios in [RFC5763], Section 5.
The role value in the offer MUST be "actpass". The role value in the offer MUST be "actpass".
o An "a=tls-id" line, as specified in [I-D.ietf-mmusic-dtls-sdp], * An "a=tls-id" line, as specified in [RFC8842], Section 5.2.
Section 5.2.
o An "a=rtcp" line, as specified in [RFC3605], Section 2.1, * An "a=rtcp" line, as specified in [RFC3605], Section 2.1,
containing the dummy value "9 IN IP4 0.0.0.0", because no containing the default value "9 IN IP4 0.0.0.0", because no
candidates have yet been gathered. candidates have yet been gathered.
o An "a=rtcp-mux" line, as specified in [RFC5761], Section 5.1.3. * An "a=rtcp-mux" line, as specified in [RFC5761], Section 5.1.3.
o If the RTP/RTCP multiplexing policy is "require", an "a=rtcp-mux- * If the RTP/RTCP multiplexing policy is "require", an "a=rtcp-mux-
only" line, as specified in [I-D.ietf-mmusic-mux-exclusive], only" line, as specified in [RFC8858], Section 4.
Section 4.
o An "a=rtcp-rsize" line, as specified in [RFC5506], Section 5. * An "a=rtcp-rsize" line, as specified in [RFC5506], Section 5.
Lastly, if a data channel has been created, a m= section MUST be Lastly, if a data channel has been created, an "m=" section MUST be
generated for data. The <media> field MUST be set to "application" generated for data. The <media> field MUST be set to "application",
and the <proto> field MUST be set to "UDP/DTLS/SCTP" and the <proto> field MUST be set to "UDP/DTLS/SCTP" [RFC8841]. The
[I-D.ietf-mmusic-sctp-sdp]. The "fmt" value MUST be set to "webrtc- <fmt> value MUST be set to "webrtc-datachannel" as specified in
datachannel" as specified in [I-D.ietf-mmusic-sctp-sdp], Section 4.1. [RFC8841], Section 4.2.2.
Within the data m= section, an "a=mid" line MUST be generated and Within the data "m=" section, an "a=mid" line MUST be generated and
included as described above, along with an "a=sctp-port" line included as described above, along with an "a=sctp-port" line
referencing the SCTP port number, as defined in referencing the SCTP port number, as defined in [RFC8841],
[I-D.ietf-mmusic-sctp-sdp], Section 5.1, and, if appropriate, an Section 5.1; and, if appropriate, an "a=max-message-size" line, as
"a=max-message-size" line, as defined in [I-D.ietf-mmusic-sctp-sdp], defined in [RFC8841], Section 6.1.
Section 6.1.
As discussed above, the following attributes of category IDENTICAL or As discussed above, the following attributes of category IDENTICAL or
TRANSPORT are included only if the data m= section either has a TRANSPORT are included only if the data "m=" section either has a
unique address or is associated with the bundle-tag (e.g., if it is unique address or is associated with the BUNDLE-tag (e.g., if it is
the only m= section): the only "m=" section):
o "a=ice-ufrag" * "a=ice-ufrag"
o "a=ice-pwd" * "a=ice-pwd"
o "a=fingerprint" * "a=fingerprint"
o "a=setup" * "a=setup"
o "a=tls-id" * "a=tls-id"
Once all m= sections have been generated, a session-level "a=group" Once all "m=" sections have been generated, a session-level "a=group"
attribute MUST be added as specified in [RFC5888]. This attribute attribute MUST be added as specified in [RFC5888]. This attribute
MUST have semantics "BUNDLE", and MUST include the mid identifiers of MUST have semantics "BUNDLE" and MUST include the mid identifiers of
each m= section. The effect of this is that the JSEP implementation each "m=" section. The effect of this is that the JSEP
offers all m= sections as one bundle group. However, whether the m= implementation offers all "m=" sections as one bundle group.
sections are bundle-only or not depends on the bundle policy. However, whether the "m=" sections are bundle-only or not depends on
the bundle policy.
The next step is to generate session-level lip sync groups as defined The next step is to generate session-level lip sync groups as defined
in [RFC5888], Section 7. For each MediaStream referenced by more in [RFC5888], Section 7. For each MediaStream referenced by more
than one RtpTransceiver (by passing those MediaStreams as arguments than one RtpTransceiver (by passing those MediaStreams as arguments
to the addTrack and addTransceiver methods), a group of type "LS" to the addTrack and addTransceiver methods), a group of type "LS"
MUST be added that contains the mid values for each RtpTransceiver. MUST be added that contains the MID values for each RtpTransceiver.
Attributes which SDP permits to either be at the session level or the Attributes that SDP permits to be at either the session level or the
media level SHOULD generally be at the media level even if they are media level SHOULD generally be at the media level even if they are
identical. This assists development and debugging by making it identical. This assists development and debugging by making it
easier to understand individual media sections, especially if one of easier to understand individual media sections, especially if one of
a set of initially identical attributes is subsequently changed. a set of initially identical attributes is subsequently changed.
However, implementations MAY choose to aggregate attributes at the However, implementations MAY choose to aggregate attributes at the
session level and JSEP implementations MUST be prepared to receive session level, and JSEP implementations MUST be prepared to receive
attributes in either location. attributes in either location.
Attributes other than the ones specified above MAY be included, Attributes other than the ones specified above MAY be included,
except for the following attributes which are specifically except for the following attributes, which are specifically
incompatible with the requirements of [I-D.ietf-rtcweb-rtp-usage], incompatible with the requirements of [RFC8834] and MUST NOT be
and MUST NOT be included: included:
o "a=crypto" * "a=crypto"
o "a=key-mgmt" * "a=key-mgmt"
o "a=ice-lite" * "a=ice-lite"
Note that when bundle is used, any additional attributes that are Note that when bundle is used, any additional attributes that are
added MUST follow the advice in [I-D.ietf-mmusic-sdp-mux-attributes] added MUST follow the advice in [RFC8859] on how those attributes
on how those attributes interact with bundle. interact with bundle.
Note that these requirements are in some cases stricter than those of Note that these requirements are in some cases stricter than those of
SDP. Implementations MUST be prepared to accept compliant SDP even SDP. Implementations MUST be prepared to accept compliant SDP even
if it would not conform to the requirements for generating SDP in if it would not conform to the requirements for generating SDP in
this specification. this specification.
5.2.2. Subsequent Offers 5.2.2. Subsequent Offers
When createOffer is called a second (or later) time, or is called When createOffer is called a second (or later) time or is called
after a local description has already been installed, the processing after a local description has already been installed, the processing
is somewhat different than for an initial offer. is somewhat different than for an initial offer.
If the previous offer was not applied using setLocalDescription, If the previous offer was not applied using setLocalDescription,
meaning the PeerConnection is still in the "stable" state, the steps meaning the PeerConnection is still in the "stable" state, the steps
for generating an initial offer should be followed, subject to the for generating an initial offer MUST be followed, subject to the
following restriction: following restriction:
o The fields of the "o=" line MUST stay the same except for the * The fields of the "o=" line MUST stay the same except for the
<session-version> field, which MUST increment by one on each call <session-version> field, which MUST increment by one on each call
to createOffer if the offer might differ from the output of the to createOffer if the offer might differ from the output of the
previous call to createOffer; implementations MAY opt to increment previous call to createOffer; implementations MAY opt to increment
<session-version> on every call. The value of the generated <session-version> on every call. The value of the generated
<session-version> is independent of the <session-version> of the <session-version> is independent of the <session-version> of the
current local description; in particular, in the case where the current local description; in particular, in the case where the
current version is N, an offer is created and applied with version current version is N, an offer is created and applied with version
N+1, and then that offer is rolled back so that the current N+1, and then that offer is rolled back so that the current
version is again N, the next generated offer will still have version is again N, the next generated offer will still have
version N+2. version N+2.
skipping to change at page 44, line 17 skipping to change at line 2097
scenarios in which this causes problems, but if this is a concern, scenarios in which this causes problems, but if this is a concern,
the solution is simply to use createOffer to ensure a unique the solution is simply to use createOffer to ensure a unique
<session-version>. <session-version>.
If the previous offer was applied using setLocalDescription, but a If the previous offer was applied using setLocalDescription, but a
corresponding answer from the remote side has not yet been applied, corresponding answer from the remote side has not yet been applied,
meaning the PeerConnection is still in the "have-local-offer" state, meaning the PeerConnection is still in the "have-local-offer" state,
an offer is generated by following the steps in the "stable" state an offer is generated by following the steps in the "stable" state
above, along with these exceptions: above, along with these exceptions:
o The "s=" and "t=" lines MUST stay the same. * The "s=" and "t=" lines MUST stay the same.
o If any RtpTransceiver has been added, and there exists an m= * If any RtpTransceiver has been added and there exists an "m="
section with a zero port in the current local description or the section with a zero port in the current local description or the
current remote description, that m= section MUST be recycled by current remote description, that "m=" section MUST be recycled by
generating an m= section for the added RtpTransceiver as if the m= generating an "m=" section for the added RtpTransceiver as if the
section were being added to the session description (including a "m=" section were being added to the session description
new MID value), and placing it at the same index as the m= section (including a new MID value) and placing it at the same index as
with a zero port. the "m=" section with a zero port.
o If an RtpTransceiver is stopped and is not associated with an m= * If an RtpTransceiver is stopped and is not associated with an "m="
section, an m= section MUST NOT be generated for it. This section, an "m=" section MUST NOT be generated for it. This
prevents adding back RtpTransceivers whose m= sections were prevents adding back RtpTransceivers whose "m=" sections were
recycled and used for a new RtpTransceiver in a previous offer/ recycled and used for a new RtpTransceiver in a previous offer/
answer exchange, as described above. answer exchange, as described above.
o If an RtpTransceiver has been stopped and is associated with an m= * If an RtpTransceiver has been stopped and is associated with an
section, and the m= section is not being recycled as described "m=" section, and the "m=" section is not being recycled as
above, an m= section MUST be generated for it with the port set to described above, an "m=" section MUST be generated for it with the
zero and all "a=msid" lines removed. port set to zero and all "a=msid" lines removed.
o For RtpTransceivers that are not stopped, the "a=msid" line(s) * For RtpTransceivers that are not stopped, the "a=msid" line or
MUST stay the same if they are present in the current description, lines MUST stay the same if they are present in the current
regardless of changes to the transceiver's direction or track. If description, regardless of changes to the transceiver's direction
no "a=msid" line is present in the current description, "a=msid" or track. If no "a=msid" line is present in the current
line(s) MUST be generated according to the same rules as for an description, "a=msid" line(s) MUST be generated according to the
initial offer. same rules as for an initial offer.
o Each "m=" and c=" line MUST be filled in with the port, relevant * Each "m=" and "c=" line MUST be filled in with the port, relevant
RTP profile, and address of the default candidate for the m= RTP profile, and address of the default candidate for the "m="
section, as described in [I-D.ietf-mmusic-ice-sip-sdp], section, as described in [RFC8839], Section 4.2.1.2 and clarified
Section 3.2.1.2, and clarified in Section 5.1.2. If no RTP in Section 5.1.2. If no RTP candidates have yet been gathered,
candidates have yet been gathered, dummy values MUST still be default values MUST still be used, as described above.
used, as described above.
o Each "a=mid" line MUST stay the same. * Each "a=mid" line MUST stay the same.
o Each "a=ice-ufrag" and "a=ice-pwd" line MUST stay the same, unless * Each "a=ice-ufrag" and "a=ice-pwd" line MUST stay the same, unless
the ICE configuration has changed (either changes to the supported the ICE configuration has changed (e.g., changes to either the
STUN/TURN servers, or the ICE candidate policy), or the supported STUN/TURN servers or the ICE candidate policy) or the
"IceRestart" option ( Section 5.2.3.1 was specified. If the m= IceRestart option (Section 5.2.3.1) was specified. If the "m="
section is bundled into another m= section, it still MUST NOT section is bundled into another "m=" section, it still MUST NOT
contain any ICE credentials. contain any ICE credentials.
o If the m= section is not bundled into another m= section, its * If the "m=" section is not bundled into another "m=" section, its
"a=rtcp" attribute line MUST be filled in with the port and "a=rtcp" attribute line MUST be filled in with the port and
address of the default RTCP candidate, as indicated in [RFC5761], address of the default RTCP candidate, as indicated in [RFC5761],
Section 5.1.3. If no RTCP candidates have yet been gathered, Section 5.1.3. If no RTCP candidates have yet been gathered,
dummy values MUST be used, as described in the initial offer default values MUST be used, as described in Section 5.2.1 above.
section above.
o If the m= section is not bundled into another m= section, for each * If the "m=" section is not bundled into another "m=" section, for
candidate that has been gathered during the most recent gathering each candidate that has been gathered during the most recent
phase (see Section 3.5.1), an "a=candidate" line MUST be added, as gathering phase (see Section 3.5.1), an "a=candidate" line MUST be
defined in [I-D.ietf-mmusic-ice-sip-sdp], Section 4.1. If added, as defined in [RFC8839], Section 5.1. If candidate
candidate gathering for the section has completed, an "a=end-of- gathering for the section has completed, an "a=end-of-candidates"
candidates" attribute MUST be added, as described in attribute MUST be added, as described in [RFC8840], Section 8.2.
[I-D.ietf-ice-trickle], Section 9.3. If the m= section is bundled If the "m=" section is bundled into another "m=" section, both
into another m= section, both "a=candidate" and "a=end-of- "a=candidate" and "a=end-of-candidates" MUST be omitted.
candidates" MUST be omitted.
o For RtpTransceivers that are still present, the "a=rid" lines MUST * For RtpTransceivers that are still present, the "a=rid" lines MUST
stay the same. stay the same.
o For RtpTransceivers that are still present, any "a=simulcast" line * For RtpTransceivers that are still present, any "a=simulcast" line
MUST stay the same. MUST stay the same.
If the previous offer was applied using setLocalDescription, and a If the previous offer was applied using setLocalDescription, and a
corresponding answer from the remote side has been applied using corresponding answer from the remote side has been applied using
setRemoteDescription, meaning the PeerConnection is in the "have- setRemoteDescription, meaning the PeerConnection is in the "have-
remote-pranswer" or "stable" states, an offer is generated based on remote-pranswer" state or the "stable" state, an offer is generated
the negotiated session descriptions by following the steps mentioned based on the negotiated session descriptions by following the steps
for the "have-local-offer" state above. mentioned for the "have-local-offer" state above.
In addition, for each existing, non-recycled, non-rejected m= section In addition, for each existing, non-recycled, non-rejected "m="
in the new offer, the following adjustments are made based on the section in the new offer, the following adjustments are made based on
contents of the corresponding m= section in the current local or the contents of the corresponding "m=" section in the current local
remote description, as appropriate: or remote description, as appropriate:
o The m= line and corresponding "a=rtpmap" and "a=fmtp" lines MUST * The "m=" line and corresponding "a=rtpmap" and "a=fmtp" lines MUST
only include media formats which have not been excluded by the only include media formats that have not been excluded by the
codec preferences of the associated transceiver, and MUST include codec preferences of the associated transceiver and also MUST
all currently available formats. Media formats that were include all currently available formats. Media formats that were
previously offered but are no longer available (e.g., a shared previously offered but are no longer available (e.g., a shared
hardware codec) MAY be excluded. hardware codec) MAY be excluded.
o Unless codec preferences have been set for the associated * Unless codec preferences have been set for the associated
transceiver, the media formats on the m= line MUST be generated in transceiver, the media formats on the "m=" line MUST be generated
the same order as in the most recent answer. Any media formats in the same order as in the most recent answer. Any media formats
that were not present in the most recent answer MUST be added that were not present in the most recent answer MUST be added
after all existing formats. after all existing formats.
o The RTP header extensions MUST only include those that are present * The RTP header extensions MUST only include those that are present
in the most recent answer. in the most recent answer.
o The RTCP feedback mechanisms MUST only include those that are * The RTCP feedback mechanisms MUST only include those that are
present in the most recent answer, except for the case of format- present in the most recent answer, except for the case of format-
specific mechanisms that are referencing a newly-added media specific mechanisms that are referencing a newly added media
format. format.
o The "a=rtcp" line MUST NOT be added if the most recent answer * The "a=rtcp" line MUST NOT be added if the most recent answer
included an "a=rtcp-mux" line. included an "a=rtcp-mux" line.
o The "a=rtcp-mux" line MUST be the same as that in the most recent * The "a=rtcp-mux" line MUST be the same as that in the most recent
answer. answer.
o The "a=rtcp-mux-only" line MUST NOT be added. * The "a=rtcp-mux-only" line MUST NOT be added.
o The "a=rtcp-rsize" line MUST NOT be added unless present in the * The "a=rtcp-rsize" line MUST NOT be added unless present in the
most recent answer. most recent answer.
o An "a=bundle-only" line MUST NOT be added, as indicated in * An "a=bundle-only" line, as defined in [RFC8843], Section 6, MUST
[I-D.ietf-mmusic-sdp-bundle-negotiation], Section 6. Instead, NOT be added. Instead, JSEP implementations MUST simply omit
JSEP implementations MUST simply omit parameters in the IDENTICAL parameters in the IDENTICAL and TRANSPORT categories for bundled
and TRANSPORT categories for bundled m= sections, as described in "m=" sections, as described in [RFC8843], Section 7.1.3.
[I-D.ietf-mmusic-sdp-bundle-negotiation], Section 8.1.
o Note that if media m= sections are bundled into a data m= section, * Note that if media "m=" sections are bundled into a data "m="
then certain TRANSPORT and IDENTICAL attributes may appear in the section, then certain TRANSPORT and IDENTICAL attributes may
data m= section even if they would otherwise only be appropriate appear in the data "m=" section even if they would otherwise only
for a media m= section (e.g., "a=rtcp-mux"). This cannot happen be appropriate for a media "m=" section (e.g., "a=rtcp-mux").
in initial offers because in the initial offer JSEP This cannot happen in initial offers because in the initial offer
implementations always list media m= sections (if any) before the JSEP implementations always list media "m=" sections (if any)
data m= section (if any), and at least one of those media m= before the data "m=" section (if any), and at least one of those
sections will not have the "a=bundle-only" attribute. Therefore, media "m=" sections will not have the "a=bundle-only" attribute.
in initial offers, any "a=bundle-only" m= sections will be bundled Therefore, in initial offers, any "a=bundle-only" "m=" sections
into a preceding non-bundle-only media m= section. will be bundled into a preceding non-bundle-only media "m="
section.
The "a=group:BUNDLE" attribute MUST include the MID identifiers The "a=group:BUNDLE" attribute MUST include the MID identifiers
specified in the bundle group in the most recent answer, minus any m= specified in the bundle group in the most recent answer, minus any
sections that have been marked as rejected, plus any newly added or "m=" sections that have been marked as rejected, plus any newly added
re-enabled m= sections. In other words, the bundle attribute must or re-enabled "m=" sections. In other words, the bundle attribute
contain all m= sections that were previously bundled, as long as they MUST contain all "m=" sections that were previously bundled, as long
are still alive, as well as any new m= sections. as they are still alive, as well as any new "m=" sections.
"a=group:LS" attributes are generated in the same way as for initial "a=group:LS" attributes are generated in the same way as for initial
offers, with the additional stipulation that any lip sync groups that offers, with the additional stipulation that any lip sync groups that
were present in the most recent answer MUST continue to exist and were present in the most recent answer MUST continue to exist and
MUST contain any previously existing MID identifiers, as long as the MUST contain any previously existing MID identifiers, as long as the
identified m= sections still exist and are not rejected, and the identified "m=" sections still exist and are not rejected, and the
group still contains at least two MID identifiers. This ensures that group still contains at least two MID identifiers. This ensures that
any synchronized "recvonly" m= sections continue to be synchronized any synchronized "recvonly" "m=" sections continue to be synchronized
in the new offer. in the new offer.
5.2.3. Options Handling 5.2.3. Options Handling
The createOffer method takes as a parameter an RTCOfferOptions The createOffer method takes as a parameter an RTCOfferOptions
object. Special processing is performed when generating a SDP object. Special processing is performed when generating an SDP
description if the following options are present. description if the following options are present.
5.2.3.1. IceRestart 5.2.3.1. IceRestart
If the "IceRestart" option is specified, with a value of "true", the If the IceRestart option is specified, with a value of "true", the
offer MUST indicate an ICE restart by generating new ICE ufrag and offer MUST indicate an ICE restart by generating new ICE ufrag and
pwd attributes, as specified in [I-D.ietf-mmusic-ice-sip-sdp], pwd attributes, as specified in [RFC8839], Section 4.4.3.1.1. If
Section 3.4.1.1.1. If this option is specified on an initial offer, this option is specified on an initial offer, it has no effect (since
it has no effect (since a new ICE ufrag and pwd are already a new ICE ufrag and pwd are already generated). Similarly, if the
generated). Similarly, if the ICE configuration has changed, this ICE configuration has changed, this option has no effect, since new
option has no effect, since new ufrag and pwd attributes will be ufrag and pwd attributes will be generated automatically. This
generated automatically. This option is primarily useful for option is primarily useful for reestablishing connectivity in cases
reestablishing connectivity in cases where failures are detected by where failures are detected by the application.
the application.
5.2.3.2. VoiceActivityDetection 5.2.3.2. VoiceActivityDetection
Silence suppression, also known as discontinuous transmission Silence suppression, also known as discontinuous transmission
("DTX"), can reduce the bandwidth used for audio by switching to a ("DTX"), can reduce the bandwidth used for audio by switching to a
special encoding when voice activity is not detected, at the cost of special encoding when voice activity is not detected, at the cost of
some fidelity. some fidelity.
If the "VoiceActivityDetection" option is specified, with a value of If the "VoiceActivityDetection" option is specified, with a value of
"true", the offer MUST indicate support for silence suppression in "true", the offer MUST indicate support for silence suppression in
skipping to change at page 48, line 21 skipping to change at line 2289
indicate that silence suppression for received audio is not desired. indicate that silence suppression for received audio is not desired.
For example, when using the Opus codec, the "usedtx=0" parameter For example, when using the Opus codec, the "usedtx=0" parameter
would be specified in the offer. In addition, the implementation would be specified in the offer. In addition, the implementation
MUST NOT use silence suppression for media it generates, regardless MUST NOT use silence suppression for media it generates, regardless
of whether the "CN" codecs or related fmtp parameters appear in the of whether the "CN" codecs or related fmtp parameters appear in the
peer's description. The impact of these rules is that silence peer's description. The impact of these rules is that silence
suppression in JSEP depends on mutual agreement of both sides, which suppression in JSEP depends on mutual agreement of both sides, which
ensures consistent handling regardless of which codec is used. ensures consistent handling regardless of which codec is used.
The "VoiceActivityDetection" option does not have any impact on the The "VoiceActivityDetection" option does not have any impact on the
setting of the "vad" value in the signaling of the client to mixer setting of the "vad" value in the signaling of the client-to-mixer
audio level header extension described in [RFC6464], Section 4. audio level header extension described in [RFC6464], Section 4.
5.3. Generating an Answer 5.3. Generating an Answer
When createAnswer is called, a new SDP description must be created When createAnswer is called, a new SDP description MUST be created
that is compatible with the supplied remote description as well as that is compatible with the supplied remote description as well as
the requirements specified in [I-D.ietf-rtcweb-rtp-usage]. The exact the requirements specified in [RFC8834]. The exact details of this
details of this process are explained below. process are explained below.
5.3.1. Initial Answers 5.3.1. Initial Answers
When createAnswer is called for the first time after a remote When createAnswer is called for the first time after a remote
description has been provided, the result is known as the initial description has been provided, the result is known as the initial
answer. If no remote description has been installed, an answer answer. If no remote description has been installed, an answer
cannot be generated, and an error MUST be returned. cannot be generated, and an error MUST be returned.
Note that the remote description SDP may not have been created by a Note that the remote description SDP may not have been created by a
JSEP endpoint and may not conform to all the requirements listed in JSEP endpoint and may not conform to all the requirements listed in
Section 5.2. For many cases, this is not a problem. However, if any Section 5.2. For many cases, this is not a problem. However, if any
mandatory SDP attributes are missing, or functionality listed as mandatory SDP attributes are missing or functionality listed as
mandatory-to-use above is not present, this MUST be treated as an mandatory-to-use above is not present, this MUST be treated as an
error, and MUST cause the affected m= sections to be marked as error and MUST cause the affected "m=" sections to be marked as
rejected. rejected.
The first step in generating an initial answer is to generate The first step in generating an initial answer is to generate
session-level attributes. The process here is identical to that session-level attributes. The process here is identical to that
indicated in the initial offers section above, except that the indicated in Section 5.2.1 above, except that the "a=ice-options"
"a=ice-options" line, with the "trickle" option as specified in line, with the "trickle" option as specified in [RFC8840],
[I-D.ietf-ice-trickle], Section 3, and the "ice2" option as specified Section 4.1.3 and the "ice2" option as specified in [RFC8445],
in [RFC8445], Section 10, is only included if such an option was Section 10, is only included if such an option was present in the
present in the offer. offer.
The next step is to generate session-level lip sync groups, as The next step is to generate session-level lip sync groups, as
defined in [RFC5888], Section 7. For each group of type "LS" present defined in [RFC5888], Section 7. For each group of type "LS" present
in the offer, select the local RtpTransceivers that are referenced by in the offer, select the local RtpTransceivers that are referenced by
the MID values in the specified group, and determine which of them the MID values in the specified group, and determine which of them
either reference a common local MediaStream (specified in the calls either reference a common local MediaStream (specified in the calls
to addTrack/addTransceiver used to create them), or have no to addTrack/addTransceiver used to create them) or have no
MediaStream to reference because they were not created by addTrack/ MediaStream to reference because they were not created by addTrack/
addTransceiver. If at least two such RtpTransceivers exist, a group addTransceiver. If at least two such RtpTransceivers exist, a group
of type "LS" with the mid values of these RtpTransceivers MUST be of type "LS" with the MID values of these RtpTransceivers MUST be
added. Otherwise the offered "LS" group MUST be ignored and no added. Otherwise, the offered "LS" group MUST be ignored and no
corresponding group generated in the answer. corresponding group generated in the answer.
As a simple example, consider the following offer of a single audio As a simple example, consider the following offer of a single audio
and single video track contained in the same MediaStream. SDP lines and single video track contained in the same MediaStream. SDP lines
not relevant to this example have been removed for clarity. As not relevant to this example have been removed for clarity. As
explained in Section 5.2, a group of type "LS" has been added that explained in Section 5.2, a group of type "LS" has been added that
references each track's RtpTransceiver. references each track's RtpTransceiver.
a=group:LS a1 v1 a=group:LS a1 v1
m=audio 10000 UDP/TLS/RTP/SAVPF 0 m=audio 10000 UDP/TLS/RTP/SAVPF 0
skipping to change at page 50, line 29 skipping to change at line 2385
an identical "LS" group. an identical "LS" group.
a=group:LS a1 v1 a=group:LS a1 v1
m=audio 20000 UDP/TLS/RTP/SAVPF 0 m=audio 20000 UDP/TLS/RTP/SAVPF 0
a=mid:a1 a=mid:a1
a=recvonly a=recvonly
m=video 20001 UDP/TLS/RTP/SAVPF 96 m=video 20001 UDP/TLS/RTP/SAVPF 96
a=mid:v1 a=mid:v1
a=recvonly a=recvonly
The Section 7.2 example later in this document shows a more involved The example in Section 7.2 shows a more involved case of "LS" group
case of "LS" group generation. generation.
The next step is to generate m= sections for each m= section that is The next step is to generate an "m=" section for each "m=" section
present in the remote offer, as specified in [RFC3264], Section 6. that is present in the remote offer, as specified in [RFC3264],
For the purposes of this discussion, any session-level attributes in Section 6. For the purposes of this discussion, any session-level
the offer that are also valid as media-level attributes are attributes in the offer that are also valid as media-level attributes
considered to be present in each m= section. Each offered m= section are considered to be present in each "m=" section. Each offered "m="
will have an associated RtpTransceiver, as described in Section 5.10. section will have an associated RtpTransceiver, as described in
If there are more RtpTransceivers than there are m= sections, the Section 5.10. If there are more RtpTransceivers than there are "m="
unmatched RtpTransceivers will need to be associated in a subsequent sections, the unmatched RtpTransceivers will need to be associated in
offer. a subsequent offer.
For each offered m= section, if any of the following conditions are For each offered "m=" section, if any of the following conditions are
true, the corresponding m= section in the answer MUST be marked as true, the corresponding "m=" section in the answer MUST be marked as
rejected by setting the port in the m= line to zero, as indicated in rejected by setting the <port> in the "m=" line to zero, as indicated
[RFC3264], Section 6, and further processing for this m= section can in [RFC3264], Section 6, and further processing for this "m=" section
be skipped: can be skipped:
o The associated RtpTransceiver has been stopped. * The associated RtpTransceiver has been stopped.
o None of the offered media formats are supported and, if * There is no offered media format that is both supported and, if
applicable, allowed by codec preferences. applicable, allowed by codec preferences.
o The bundle policy is "max-bundle", and this is not the first m= * The bundle policy is "max-bundle", and this is not the first "m="
section or in the same bundle group as the first m= section. section or in the same bundle group as the first "m=" section.
o The bundle policy is "balanced", and this is not the first m= * The bundle policy is "balanced", and this is not the first "m="
section for this media type or in the same bundle group as the section for this media type or in the same bundle group as the
first m= section for this media type. first "m=" section for this media type.
o This m= section is in a bundle group, and the group's offerer * This "m=" section is in a bundle group, and the group's offerer
tagged m= section is being rejected due to one of the above tagged "m=" section is being rejected due to one of the above
reasons. This requires all m= sections in the bundle group to be reasons. This requires all "m=" sections in the bundle group to
rejected, as specified in be rejected, as specified in [RFC8843], Section 7.3.3.
[I-D.ietf-mmusic-sdp-bundle-negotiation], Section 7.3.3.
Otherwise, each m= section in the answer should then be generated as Otherwise, each "m=" section in the answer MUST then be generated as
specified in [RFC3264], Section 6.1. For the m= line itself, the specified in [RFC3264], Section 6.1. For the "m=" line itself, the
following rules must be followed: following rules MUST be followed:
o The port value would normally be set to the port of the default * The <port> value would normally be set to the port of the default
ICE candidate for this m= section, but given that no candidates ICE candidate for this "m=" section, but given that no candidates
are available yet, the "dummy" port value of 9 (Discard) MUST be are available yet, the default <port> value of 9 (Discard) MUST be
used, as indicated in [I-D.ietf-ice-trickle], Section 5.1. used, as indicated in [RFC8840], Section 4.1.1.
o The <proto> field MUST be set to exactly match the <proto> field * The <proto> field MUST be set to exactly match the <proto> field
for the corresponding m= line in the offer. for the corresponding "m=" line in the offer.
o If codec preferences have been set for the associated transceiver, * If codec preferences have been set for the associated transceiver,
media formats MUST be generated in the corresponding order, media formats MUST be generated in the corresponding order,
regardless of what was offered, and MUST exclude any codecs not regardless of what was offered, and MUST exclude any codecs not
present in the codec preferences. present in the codec preferences.
o Otherwise, the media formats on the m= line MUST be generated in * Otherwise, the media formats on the "m=" line MUST be generated in
the same order as those offered in the current remote description, the same order as those offered in the current remote description,
excluding any currently unsupported formats. Any currently excluding any currently unsupported formats. Any currently
available media formats that are not present in the current remote available media formats that are not present in the current remote
description MUST be added after all existing formats. description MUST be added after all existing formats.
o In either case, the media formats in the answer MUST include at * In either case, the media formats in the answer MUST include at
least one format that is present in the offer, but MAY include least one format that is present in the offer but MAY include
formats that are locally supported but not present in the offer, formats that are locally supported but not present in the offer,
as mentioned in [RFC3264], Section 6.1. If no common format as mentioned in [RFC3264], Section 6.1. If no common format
exists, the m= section is rejected as described above. exists, the "m=" section is rejected as described above.
The m= line MUST be followed immediately by a "c=" line, as specified The "m=" line MUST be followed immediately by a "c=" line, as
in [RFC4566], Section 5.7. Again, as no candidates are available specified in [RFC4566], Section 5.7. Again, as no candidates are
yet, the "c=" line must contain the "dummy" value "IN IP4 0.0.0.0", available yet, the "c=" line MUST contain the default value "IN IP4
as defined in [I-D.ietf-ice-trickle], Section 5.1. 0.0.0.0", as defined in [RFC8840], Section 4.1.3.
If the offer supports bundle, all m= sections to be bundled must use If the offer supports bundle, all "m=" sections to be bundled MUST
the same ICE credentials and candidates; all m= sections not being use the same ICE credentials and candidates; all "m=" sections not
bundled must use unique ICE credentials and candidates. Each m= being bundled MUST use unique ICE credentials and candidates. Each
section MUST contain the following attributes (which are of attribute "m=" section MUST contain the following attributes (which are of
types other than IDENTICAL and TRANSPORT): attribute types other than IDENTICAL or TRANSPORT):
o If and only if present in the offer, an "a=mid" line, as specified * If and only if present in the offer, an "a=mid" line, as specified
in [RFC5888], Section 9.1. The "mid" value MUST match that in [RFC5888], Section 9.1. The MID value MUST match that
specified in the offer. specified in the offer.
o A direction attribute, determined by applying the rules regarding * A direction attribute, determined by applying the rules regarding
the offered direction specified in [RFC3264], Section 6.1, and the offered direction specified in [RFC3264], Section 6.1, and
then intersecting with the direction of the associated then intersecting with the direction of the associated
RtpTransceiver. For example, in the case where an m= section is RtpTransceiver. For example, in the case where an "m=" section is
offered as "sendonly", and the local transceiver is set to offered as "sendonly" and the local transceiver is set to
"sendrecv", the result in the answer is a "recvonly" direction. "sendrecv", the result in the answer is a "recvonly" direction.
o For each media format on the m= line, "a=rtpmap" and "a=fmtp" * For each media format on the "m=" line, "a=rtpmap" and "a=fmtp"
lines, as specified in [RFC4566], Section 6, and [RFC3264], lines, as specified in [RFC4566], Section 6 and [RFC3264],
Section 6.1. Section 6.1.
o If "rtx" is present in the offer, for each primary codec where RTP * If "rtx" is present in the offer, for each primary codec where RTP
retransmission should be used, a corresponding "a=rtpmap" line retransmission should be used, a corresponding "a=rtpmap" line
indicating "rtx" with the clock rate of the primary codec and an indicating "rtx" with the clock rate of the primary codec and an
"a=fmtp" line that references the payload type of the primary "a=fmtp" line that references the payload type of the primary
codec, as specified in [RFC4588], Section 8.1. codec, as specified in [RFC4588], Section 8.1.
o For each supported FEC mechanism, "a=rtpmap" and "a=fmtp" lines, * For each supported FEC mechanism, "a=rtpmap" and "a=fmtp" lines,
as specified in [RFC4566], Section 6. The FEC mechanisms that as specified in [RFC4566], Section 6. The FEC mechanisms that
MUST be supported are specified in [I-D.ietf-rtcweb-fec], MUST be supported are specified in [RFC8854], Section 7, and
Section 6, and specific usage for each media type is outlined in specific usage for each media type is outlined in Sections 4 and
Sections 4 and 5. 5.
o If this m= section is for media with configurable durations of * If this "m=" section is for media with configurable durations of
media per packet, e.g., audio, an "a=maxptime" line, as described media per packet, e.g., audio, an "a=maxptime" line, as described
in Section 5.2. in Section 5.2.
o If this m= section is for video media, and there are known * If this "m=" section is for video media and there are known
limitations on the size of images which can be decoded, an limitations on the size of images that can be decoded, an
"a=imageattr" line, as specified in Section 3.6. "a=imageattr" line, as specified in Section 3.6.
o For each supported RTP header extension that is present in the * For each supported RTP header extension that is present in the
offer, an "a=extmap" line, as specified in [RFC5285], Section 5. offer, an "a=extmap" line, as specified in [RFC5285], Section 5.
The list of header extensions that SHOULD/MUST be supported is The list of header extensions that SHOULD/MUST be supported is
specified in [I-D.ietf-rtcweb-rtp-usage], Section 5.2. Any header specified in [RFC8834], Section 5.2. Any header extensions that
extensions that require encryption MUST be specified as indicated require encryption MUST be specified as indicated in [RFC6904],
in [RFC6904], Section 4. Section 4.
o For each supported RTCP feedback mechanism that is present in the * For each supported RTCP feedback mechanism that is present in the
offer, an "a=rtcp-fb" line, as specified in [RFC4585], offer, an "a=rtcp-fb" line, as specified in [RFC4585],
Section 4.2. The list of RTCP feedback mechanisms that SHOULD/ Section 4.2. The list of RTCP feedback mechanisms that SHOULD/
MUST be supported is specified in [I-D.ietf-rtcweb-rtp-usage], MUST be supported is specified in [RFC8834], Section 5.1.
Section 5.1.
o If the RtpTransceiver has a sendrecv or sendonly direction: * If the RtpTransceiver has a sendrecv or sendonly direction:
* For each MediaStream that was associated with the transceiver - For each MediaStream that was associated with the transceiver
when it was created via addTrack or addTransceiver, an "a=msid" when it was created via addTrack or addTransceiver, an "a=msid"
line, as specified in [I-D.ietf-mmusic-msid], Section 2, but line, as specified in [RFC8830], Section 2, but omitting the
omitting the "appdata" field. "appdata" field.
Each m= section which is not bundled into another m= section, MUST Each "m=" section that is not bundled into another "m=" section MUST
contain the following attributes (which are of category IDENTICAL or contain the following attributes (which are of category IDENTICAL or
TRANSPORT): TRANSPORT):
o "a=ice-ufrag" and "a=ice-pwd" lines, as specified in * "a=ice-ufrag" and "a=ice-pwd" lines, as specified in [RFC8839],
[I-D.ietf-mmusic-ice-sip-sdp], Section 4.4. Section 5.4.
o For each desired digest algorithm, one or more "a=fingerprint" * For each desired digest algorithm, one or more "a=fingerprint"
lines for each of the endpoint's certificates, as specified in lines for each of the endpoint's certificates, as specified in
[RFC8122], Section 5. [RFC8122], Section 5.
o An "a=setup" line, as specified in [RFC4145], Section 4, and * An "a=setup" line, as specified in [RFC4145], Section 4 and
clarified for use in DTLS-SRTP scenarios in [RFC5763], Section 5. clarified for use in DTLS-SRTP scenarios in [RFC5763], Section 5.
The role value in the answer MUST be "active" or "passive". When The role value in the answer MUST be "active" or "passive". When
the offer contains the "actpass" value, as will always be the case the offer contains the "actpass" value, as will always be the case
with JSEP endpoints, the answerer SHOULD use the "active" role. with JSEP endpoints, the answerer SHOULD use the "active" role.
Offers from non-JSEP endpoints MAY send other values for Offers from non-JSEP endpoints MAY send other values for
"a=setup", in which case the answer MUST use a value consistent "a=setup", in which case the answer MUST use a value consistent
with the value in the offer. with the value in the offer.
o An "a=tls-id" line, as specified in [I-D.ietf-mmusic-dtls-sdp], * An "a=tls-id" line, as specified in [RFC8842], Section 5.3.
Section 5.3.
o If present in the offer, an "a=rtcp-mux" line, as specified in * If present in the offer, an "a=rtcp-mux" line, as specified in
[RFC5761], Section 5.1.3. Otherwise, an "a=rtcp" line, as [RFC5761], Section 5.1.3. Otherwise, an "a=rtcp" line, as
specified in [RFC3605], Section 2.1, containing the dummy value "9 specified in [RFC3605], Section 2.1, containing the default value
IN IP4 0.0.0.0" (because no candidates have yet been gathered). "9 IN IP4 0.0.0.0" (because no candidates have yet been gathered).
o If present in the offer, an "a=rtcp-rsize" line, as specified in * If present in the offer, an "a=rtcp-rsize" line, as specified in
[RFC5506], Section 5. [RFC5506], Section 5.
If a data channel m= section has been offered, a m= section MUST also If a data channel "m=" section has been offered, an "m=" section MUST
be generated for data. The <media> field MUST be set to also be generated for data. The <media> field MUST be set to
"application" and the <proto> and <fmt> fields MUST be set to exactly "application", and the <proto> and <fmt> fields MUST be set to
match the fields in the offer. exactly match the fields in the offer.
Within the data m= section, an "a=mid" line MUST be generated and Within the data "m=" section, an "a=mid" line MUST be generated and
included as described above, along with an "a=sctp-port" line included as described above, along with an "a=sctp-port" line
referencing the SCTP port number, as defined in referencing the SCTP port number, as defined in [RFC8841],
[I-D.ietf-mmusic-sctp-sdp], Section 5.1, and, if appropriate, an Section 5.1; and, if appropriate, an "a=max-message-size" line, as
"a=max-message-size" line, as defined in [I-D.ietf-mmusic-sctp-sdp], defined in [RFC8841], Section 6.1.
Section 6.1.
As discussed above, the following attributes of category IDENTICAL or As discussed above, the following attributes of category IDENTICAL or
TRANSPORT are included only if the data m= section is not bundled TRANSPORT are included only if the data "m=" section is not bundled
into another m= section: into another "m=" section:
o "a=ice-ufrag" * "a=ice-ufrag"
o "a=ice-pwd" * "a=ice-pwd"
o "a=fingerprint" * "a=fingerprint"
o "a=setup" * "a=setup"
o "a=tls-id" * "a=tls-id"
Note that if media m= sections are bundled into a data m= section, Note that if media "m=" sections are bundled into a data "m="
then certain TRANSPORT and IDENTICAL attributes may also appear in section, then certain TRANSPORT and IDENTICAL attributes may also
the data m= section even if they would otherwise only be appropriate appear in the data "m=" section even if they would otherwise only be
for a media m= section (e.g., "a=rtcp-mux"). appropriate for a media "m=" section (e.g., "a=rtcp-mux").
If "a=group" attributes with semantics of "BUNDLE" are offered, If "a=group" attributes with semantics of "BUNDLE" are offered,
corresponding session-level "a=group" attributes MUST be added as corresponding session-level "a=group" attributes MUST be added as
specified in [RFC5888]. These attributes MUST have semantics specified in [RFC5888]. These attributes MUST have semantics
"BUNDLE", and MUST include the all mid identifiers from the offered "BUNDLE" and MUST include all mid identifiers from the offered bundle
bundle groups that have not been rejected. Note that regardless of groups that have not been rejected. Note that regardless of the
the presence of "a=bundle-only" in the offer, no m= sections in the presence of "a=bundle-only" in the offer, all "m=" sections in the
answer should have an "a=bundle-only" line. answer MUST NOT have an "a=bundle-only" line.
Attributes that are common between all m= sections MAY be moved to Attributes that are common between all "m=" sections MAY be moved to
session-level, if explicitly defined to be valid at session-level. the session level if explicitly defined to be valid at the session
level.
The attributes prohibited in the creation of offers are also The attributes prohibited in the creation of offers are also
prohibited in the creation of answers. prohibited in the creation of answers.
5.3.2. Subsequent Answers 5.3.2. Subsequent Answers
When createAnswer is called a second (or later) time, or is called When createAnswer is called a second (or later) time or is called
after a local description has already been installed, the processing after a local description has already been installed, the processing
is somewhat different than for an initial answer. is somewhat different than for an initial answer.
If the previous answer was not applied using setLocalDescription, If the previous answer was not applied using setLocalDescription,
meaning the PeerConnection is still in the "have-remote-offer" state, meaning the PeerConnection is still in the "have-remote-offer" state,
the steps for generating an initial answer should be followed, the steps for generating an initial answer MUST be followed, subject
subject to the following restriction: to the following restriction:
o The fields of the "o=" line MUST stay the same except for the * The fields of the "o=" line MUST stay the same except for the
<session-version> field, which MUST increment if the session <session-version> field, which MUST increment if the session
description changes in any way from the previously generated description changes in any way from the previously generated
answer. answer.
If any session description was previously supplied to If any session description was previously supplied to
setLocalDescription, an answer is generated by following the steps in setLocalDescription, an answer is generated by following the steps in
the "have-remote-offer" state above, along with these exceptions: the "have-remote-offer" state above, along with these exceptions:
o The "s=" and "t=" lines MUST stay the same. * The "s=" and "t=" lines MUST stay the same.
o Each "m=" and c=" line MUST be filled in with the port and address * Each "m=" and "c=" line MUST be filled in with the port and
of the default candidate for the m= section, as described in address of the default candidate for the "m=" section, as
[I-D.ietf-mmusic-ice-sip-sdp], Section 3.2.1.2. Note that in described in [RFC8839], Section 4.2.1.2. Note that in certain
certain cases, the m= line protocol may not match that of the cases, the "m=" line protocol may not match that of the default
default candidate, because the m= line protocol value MUST match candidate, because the "m=" line protocol value MUST match what
what was supplied in the offer, as described above. was supplied in the offer, as described above.
o Each "a=ice-ufrag" and "a=ice-pwd" line MUST stay the same, unless * Each "a=ice-ufrag" and "a=ice-pwd" line MUST stay the same, unless
the m= section is restarting, in which case new ICE credentials the "m=" section is restarting, in which case new ICE credentials
must be created as specified in [I-D.ietf-mmusic-ice-sip-sdp], MUST be created as specified in [RFC8839], Section 4.4.1.1.1. If
Section 3.4.1.1.1. If the m= section is bundled into another m= the "m=" section is bundled into another "m=" section, it still
section, it still MUST NOT contain any ICE credentials. MUST NOT contain any ICE credentials.
o Each "a=tls-id" line MUST stay the same unless the offerer's * Each "a=tls-id" line MUST stay the same, unless the offerer's
"a=tls-id" line changed, in which case a new "a=tls-id" value MUST "a=tls-id" line changed, in which case a new tls-id value MUST be
be created, as described in [I-D.ietf-mmusic-dtls-sdp], created, as described in [RFC8842], Section 5.2.
Section 5.2.
o Each "a=setup" line MUST use an "active" or "passive" role value * Each "a=setup" line MUST use an "active" or "passive" role value
consistent with the existing DTLS association, if the association consistent with the existing DTLS association, if the association
is being continued by the offerer. is being continued by the offerer.
o RTCP multiplexing must be used, and an "a=rtcp-mux" line inserted * RTCP multiplexing MUST be used, and an "a=rtcp-mux" line inserted
if and only if the m= section previously used RTCP multiplexing. if and only if the "m=" section previously used RTCP multiplexing.
o If the m= section is not bundled into another m= section and RTCP * If the "m=" section is not bundled into another "m=" section and
multiplexing is not active, an "a=rtcp" attribute line MUST be RTCP multiplexing is not active, an "a=rtcp" attribute line MUST
filled in with the port and address of the default RTCP candidate. be filled in with the port and address of the default RTCP
If no RTCP candidates have yet been gathered, dummy values MUST be candidate. If no RTCP candidates have yet been gathered, default
used, as described in the initial answer section above. values MUST be used, as described in Section 5.3.1 above.
o If the m= section is not bundled into another m= section, for each * If the "m=" section is not bundled into another "m=" section, for
candidate that has been gathered during the most recent gathering each candidate that has been gathered during the most recent
phase (see Section 3.5.1), an "a=candidate" line MUST be added, as gathering phase (see Section 3.5.1), an "a=candidate" line MUST be
defined in [I-D.ietf-mmusic-ice-sip-sdp], Section 4.1. If added, as defined in [RFC8839], Section 5.1. If candidate
candidate gathering for the section has completed, an "a=end-of- gathering for the section has completed, an "a=end-of-candidates"
candidates" attribute MUST be added, as described in attribute MUST be added, as described in [RFC8840], Section 8.2.
[I-D.ietf-ice-trickle], Section 9.3. If the m= section is bundled If the "m=" section is bundled into another "m=" section, both
into another m= section, both "a=candidate" and "a=end-of- "a=candidate" and "a=end-of-candidates" MUST be omitted.
candidates" MUST be omitted.
o For RtpTransceivers that are not stopped, the "a=msid" line(s) * For RtpTransceivers that are not stopped, the "a=msid" line(s)
MUST stay the same, regardless of changes to the transceiver's MUST stay the same, regardless of changes to the transceiver's
direction or track. If no "a=msid" line is present in the current direction or track. If no "a=msid" line is present in the current
description, "a=msid" line(s) MUST be generated according to the description, "a=msid" line(s) MUST be generated according to the
same rules as for an initial answer. same rules as for an initial answer.
5.3.3. Options Handling 5.3.3. Options Handling
The createAnswer method takes as a parameter an RTCAnswerOptions The createAnswer method takes as a parameter an RTCAnswerOptions
object. The set of parameters for RTCAnswerOptions is different than object. The set of parameters for RTCAnswerOptions is different than
those supported in RTCOfferOptions; the IceRestart option is those supported in RTCOfferOptions; the IceRestart option is
unnecessary, as ICE credentials will automatically be changed for all unnecessary, as ICE credentials will automatically be changed for all
m= sections where the offerer chose to perform ICE restart. "m=" sections where the offerer chose to perform ICE restart.
The following options are supported in RTCAnswerOptions. The following options are supported in RTCAnswerOptions.
5.3.3.1. VoiceActivityDetection 5.3.3.1. VoiceActivityDetection
Silence suppression in the answer is handled as described in Silence suppression in the answer is handled as described in
Section 5.2.3.2, with one exception: if support for silence Section 5.2.3.2, with one exception: if support for silence
suppression was not indicated in the offer, the suppression was not indicated in the offer, the
VoiceActivityDetection parameter has no effect, and the answer should VoiceActivityDetection parameter has no effect, and the answer MUST
be generated as if VoiceActivityDetection was set to false. This is be generated as if VoiceActivityDetection was set to "false". This
done on a per-codec basis (e.g., if the offerer somehow offered is done on a per-codec basis (e.g., if the offerer somehow offered
support for CN but set "usedtx=0" for Opus, setting support for CN but set "usedtx=0" for Opus, setting
VoiceActivityDetection to true would result in an answer with CN VoiceActivityDetection to "true" would result in an answer with CN
codecs and "usedtx=0"). The impact of this rule is that an answerer codecs and "usedtx=0"). The impact of this rule is that an answerer
will not try to use silence suppression with any endpoint that does will not try to use silence suppression with any endpoint that does
not offer it, making silence suppression support bilateral even with not offer it, making silence suppression support bilateral even with
non-JSEP endpoints. non-JSEP endpoints.
5.4. Modifying an Offer or Answer 5.4. Modifying an Offer or Answer
The SDP returned from createOffer or createAnswer MUST NOT be changed The SDP returned from createOffer or createAnswer MUST NOT be changed
before passing it to setLocalDescription. If precise control over before passing it to setLocalDescription. If precise control over
the SDP is needed, the aforementioned createOffer/createAnswer the SDP is needed, the aforementioned createOffer/createAnswer
skipping to change at page 57, line 23 skipping to change at line 2700
application MAY modify the SDP to reduce its capabilities before application MAY modify the SDP to reduce its capabilities before
sending it to the far side, as long as it follows the rules above sending it to the far side, as long as it follows the rules above
that define a valid JSEP offer or answer. Likewise, an application that define a valid JSEP offer or answer. Likewise, an application
that has received an offer or answer from a peer MAY modify the that has received an offer or answer from a peer MAY modify the
received SDP, subject to the same constraints, before calling received SDP, subject to the same constraints, before calling
setRemoteDescription. setRemoteDescription.
As always, the application is solely responsible for what it sends to As always, the application is solely responsible for what it sends to
the other party, and all incoming SDP will be processed by the JSEP the other party, and all incoming SDP will be processed by the JSEP
implementation to the extent of its capabilities. It is an error to implementation to the extent of its capabilities. It is an error to
assume that all SDP is well-formed; however, one should be able to assume that all SDP is well formed; however, one should be able to
assume that any implementation of this specification will be able to assume that any implementation of this specification will be able to
process, as a remote offer or answer, unmodified SDP coming from any process, as a remote offer or answer, unmodified SDP coming from any
other implementation of this specification. other implementation of this specification.
5.5. Processing a Local Description 5.5. Processing a Local Description
When a SessionDescription is supplied to setLocalDescription, the When a SessionDescription is supplied to setLocalDescription, the
following steps MUST be performed: following steps MUST be performed:
o If the description is of type "rollback", follow the processing * If the description is of type "rollback", follow the processing
defined in Section 5.7 and skip the processing described in the defined in Section 5.7 and skip the processing described in the
rest of this section. rest of this section.
o Otherwise, the type of the SessionDescription is checked against * Otherwise, the type of the SessionDescription is checked against
the current state of the PeerConnection: the current state of the PeerConnection:
* If the type is "offer", the PeerConnection state MUST be either - If the type is "offer", the PeerConnection state MUST be either
"stable" or "have-local-offer". "stable" or "have-local-offer".
* If the type is "pranswer" or "answer", the PeerConnection state - If the type is "pranswer" or "answer", the PeerConnection state
MUST be either "have-remote-offer" or "have-local-pranswer". MUST be either "have-remote-offer" or "have-local-pranswer".
o If the type is not correct for the current state, processing MUST * If the type is not correct for the current state, processing MUST
stop and an error MUST be returned. stop and an error MUST be returned.
o The SessionDescription is then checked to ensure that its contents * The SessionDescription is then checked to ensure that its contents
are identical to those generated in the last call to createOffer/ are identical to those generated in the last call to createOffer/
createAnswer, and thus have not been altered, as discussed in createAnswer, and thus have not been altered, as discussed in
Section 5.4; otherwise, processing MUST stop and an error MUST be Section 5.4; otherwise, processing MUST stop and an error MUST be
returned. returned.
o Next, the SessionDescription is parsed into a data structure, as * Next, the SessionDescription is parsed into a data structure, as
described in Section 5.8 below. described in Section 5.8 below.
o Finally, the parsed SessionDescription is applied as described in * Finally, the parsed SessionDescription is applied as described in
Section 5.9 below. Section 5.9 below.
5.6. Processing a Remote Description 5.6. Processing a Remote Description
When a SessionDescription is supplied to setRemoteDescription, the When a SessionDescription is supplied to setRemoteDescription, the
following steps MUST be performed: following steps MUST be performed:
o If the description is of type "rollback", follow the processing * If the description is of type "rollback", follow the processing
defined in Section 5.7 and skip the processing described in the defined in Section 5.7 and skip the processing described in the
rest of this section. rest of this section.
o Otherwise, the type of the SessionDescription is checked against * Otherwise, the type of the SessionDescription is checked against
the current state of the PeerConnection: the current state of the PeerConnection:
* If the type is "offer", the PeerConnection state MUST be either - If the type is "offer", the PeerConnection state MUST be either
"stable" or "have-remote-offer". "stable" or "have-remote-offer".
* If the type is "pranswer" or "answer", the PeerConnection state - If the type is "pranswer" or "answer", the PeerConnection state
MUST be either "have-local-offer" or "have-remote-pranswer". MUST be either "have-local-offer" or "have-remote-pranswer".
o If the type is not correct for the current state, processing MUST * If the type is not correct for the current state, processing MUST
stop and an error MUST be returned. stop and an error MUST be returned.
o Next, the SessionDescription is parsed into a data structure, as * Next, the SessionDescription is parsed into a data structure, as
described in Section 5.8 below. If parsing fails for any reason, described in Section 5.8 below. If parsing fails for any reason,
processing MUST stop and an error MUST be returned. processing MUST stop and an error MUST be returned.
o Finally, the parsed SessionDescription is applied as described in * Finally, the parsed SessionDescription is applied as described in
Section 5.10 below. Section 5.10 below.
5.7. Processing a Rollback 5.7. Processing a Rollback
A rollback may be performed if the PeerConnection is in any state A rollback may be performed if the PeerConnection is in any state
except for "stable". This means that both offers and provisional except for "stable". This means that both offers and provisional
answers can be rolled back. Rollback can only be used to cancel answers can be rolled back. Rollback can only be used to cancel
proposed changes; there is no support for rolling back from a stable proposed changes; there is no support for rolling back from a
state to a previous stable state. If a rollback is attempted in the "stable" state to a previous "stable" state. If a rollback is
"stable" state, processing MUST stop and an error MUST be returned. attempted in the "stable" state, processing MUST stop and an error
Note that this implies that once the answerer has performed MUST be returned. Note that this implies that once the answerer has
setLocalDescription with his answer, this cannot be rolled back. performed setLocalDescription with its answer, this cannot be rolled
back.
The effect of rollback MUST be the same regardless of whether The effect of rollback MUST be the same regardless of whether
setLocalDescription or setRemoteDescription is called. setLocalDescription or setRemoteDescription is called.
In order to process rollback, a JSEP implementation abandons the In order to process rollback, a JSEP implementation abandons the
current offer/answer transaction, sets the signaling state to current offer/answer transaction, sets the signaling state to
"stable", and sets the pending local and/or remote description (see "stable", and sets the pending local and/or remote description (see
Section 4.1.12 and Section 4.1.14) to null. Any resources or Sections 4.1.14 and 4.1.16) to "null". Any resources or candidates
candidates that were allocated by the abandoned local description are that were allocated by the abandoned local description are discarded;
discarded; any media that is received is processed according to the any media that is received is processed according to the previous
previous local and remote descriptions. local and remote descriptions.
A rollback disassociates any RtpTransceivers that were associated A rollback disassociates any RtpTransceivers that were associated
with m= sections by the application of the rolled-back session with "m=" sections by the application of the rolled-back session
description (see Section 5.10 and Section 5.9). This means that some description (see Sections 5.10 and 5.9). This means that some
RtpTransceivers that were previously associated will no longer be RtpTransceivers that were previously associated will no longer be
associated with any m= section; in such cases, the value of the associated with any "m=" section; in such cases, the value of the
RtpTransceiver's mid property MUST be set to null, and the mapping RtpTransceiver's mid property MUST be set to "null", and the mapping
between the transceiver and its m= section index MUST be discarded. between the transceiver and its "m=" section index MUST be discarded.
RtpTransceivers that were created by applying a remote offer that was RtpTransceivers that were created by applying a remote offer that was
subsequently rolled back MUST be stopped and removed from the subsequently rolled back MUST be stopped and removed from the
PeerConnection. However, a RtpTransceiver MUST NOT be removed if a PeerConnection. However, an RtpTransceiver MUST NOT be removed if a
track was attached to the RtpTransceiver via the addTrack method. track was attached to the RtpTransceiver via the addTrack method.
This is so that an application may call addTrack, then call This is so that an application may call addTrack, then call
setRemoteDescription with an offer, then roll back that offer, then setRemoteDescription with an offer, then roll back that offer, then
call createOffer and have a m= section for the added track appear in call createOffer and have an "m=" section for the added track appear
the generated offer. in the generated offer.
5.8. Parsing a Session Description 5.8. Parsing a Session Description
The SDP contained in the session description object consists of a The SDP contained in the session description object consists of a
sequence of text lines, each containing a key-value expression, as sequence of text lines, each containing a key-value expression, as
described in [RFC4566], Section 5. The SDP is read, line-by-line, described in [RFC4566], Section 5. The SDP is read, line by line,
and converted to a data structure that contains the deserialized and converted to a data structure that contains the deserialized
information. However, SDP allows many types of lines, not all of information. However, SDP allows many types of lines, not all of
which are relevant to JSEP applications. For each line, the which are relevant to JSEP applications. For each line, the
implementation will first ensure it is syntactically correct implementation will first ensure that it is syntactically correct
according to its defining ABNF, check that it conforms to [RFC4566] according to its defining ABNF, check that it conforms to the
and [RFC3264] semantics, and then either parse and store or discard semantics used in [RFC4566] and [RFC3264], and then either parse and
the provided value, as described below. store or discard the provided value, as described below.
If any line is not well-formed, or cannot be parsed as described, the If any line is not well formed or cannot be parsed as described, the
parser MUST stop with an error and reject the session description, parser MUST stop with an error and reject the session description,
even if the value is to be discarded. This ensures that even if the value is to be discarded. This ensures that
implementations do not accidentally misinterpret ambiguous SDP. implementations do not accidentally misinterpret ambiguous SDP.
5.8.1. Session-Level Parsing 5.8.1. Session-Level Parsing
First, the session-level lines are checked and parsed. These lines First, the session-level lines are checked and parsed. These lines
MUST occur in a specific order, and with a specific syntax, as MUST occur in a specific order, and with a specific syntax, as
defined in [RFC4566], Section 5. Note that while the specific line defined in [RFC4566], Section 5. Note that while the specific line
types (e.g. "v=", "c=") MUST occur in the defined order, lines of the types (e.g., "v=", "c=") MUST occur in the defined order, lines of
same type (typically "a=") can occur in any order. the same type (typically "a=") can occur in any order.
The following non-attribute lines are not meaningful in the JSEP The following non-attribute lines are not meaningful in the JSEP
context and MAY be discarded once they have been checked. context and MAY be discarded once they have been checked.
The "c=" line MUST be checked for syntax but its value is only * The "c=" line MUST be checked for syntax, but its value is only
used for ICE mismatch detection, as defined in [RFC8445], used for ICE mismatch detection, as defined in [RFC8445],
Section 5.4. Note that JSEP implementations should never Section 5.4. Note that JSEP implementations should never
encounter this condition because ICE is required for WebRTC. encounter this condition because ICE is required for WebRTC.
The "i=", "u=", "e=", "p=", "t=", "r=", "z=", and "k=" lines are * The "i=", "u=", "e=", "p=", "t=", "r=", "z=", and "k=" lines MUST
not used by this specification; they MUST be checked for syntax be checked for syntax, but their values are not otherwise used.
but their values are not used.
The remaining non-attribute lines are processed as follows: The remaining non-attribute lines are processed as follows:
The "v=" line MUST have a version of 0, as specified in [RFC4566], * The "v=" line MUST have a version of 0, as specified in [RFC4566],
Section 5.1. Section 5.1.
The "o=" line MUST be parsed as specified in [RFC4566], * The "o=" line MUST be parsed as specified in [RFC4566],
Section 5.2. Section 5.2.
The "b=" line, if present, MUST be parsed as specified in * The "b=" line, if present, MUST be parsed as specified in
[RFC4566], Section 5.8, and the bwtype and bandwidth values [RFC4566], Section 5.8, and the bwtype and bandwidth values
stored. stored.
Finally, the attribute lines are processed. Specific processing MUST Finally, the attribute lines are processed. Specific processing MUST
be applied for the following session-level attribute ("a=") lines: be applied for the following session-level attribute ("a=") lines:
o Any "a=group" lines are parsed as specified in [RFC5888], * Any "a=group" lines are parsed as specified in [RFC5888],
Section 5, and the group's semantics and mids are stored. Section 5, and the group's semantics and mids are stored.
o If present, a single "a=ice-lite" line is parsed as specified in * If present, a single "a=ice-lite" line is parsed as specified in
[I-D.ietf-mmusic-ice-sip-sdp], Section 4.3, and a value indicating [RFC8839], Section 5.3, and a value indicating the presence of
the presence of ice-lite is stored. ice-lite is stored.
o If present, a single "a=ice-ufrag" line is parsed as specified in * If present, a single "a=ice-ufrag" line is parsed as specified in
[I-D.ietf-mmusic-ice-sip-sdp], Section 4.4, and the ufrag value is [RFC8839], Section 5.4, and the ufrag value is stored.
stored.
o If present, a single "a=ice-pwd" line is parsed as specified in * If present, a single "a=ice-pwd" line is parsed as specified in
[I-D.ietf-mmusic-ice-sip-sdp], Section 4.4, and the password value [RFC8839], Section 5.4, and the password value is stored.
is stored.
o If present, a single "a=ice-options" line is parsed as specified * If present, a single "a=ice-options" line is parsed as specified
in [I-D.ietf-mmusic-ice-sip-sdp], Section 4.6, and the set of in [RFC8839], Section 5.6, and the set of specified options is
specified options is stored. stored.
o Any "a=fingerprint" lines are parsed as specified in [RFC8122], * Any "a=fingerprint" lines are parsed as specified in [RFC8122],
Section 5, and the set of fingerprint and algorithm values is Section 5, and the set of fingerprint and algorithm values is
stored. stored.
o If present, a single "a=setup" line is parsed as specified in * If present, a single "a=setup" line is parsed as specified in
[RFC4145], Section 4, and the setup value is stored. [RFC4145], Section 4, and the setup value is stored.
o If present, a single "a=tls-id" line is parsed as specified in * If present, a single "a=tls-id" line is parsed as specified in
[I-D.ietf-mmusic-dtls-sdp] Section 5, and the tls-id value is [RFC8842], Section 5, and the attribute value is stored.
stored.
o Any "a=identity" lines are parsed and the identity values stored * Any "a=identity" lines are parsed and the identity values stored
for subsequent verification, as specified for subsequent verification, as specified in [RFC8827], Section 5.
[I-D.ietf-rtcweb-security-arch], Section 5.
o Any "a=extmap" lines are parsed as specified in [RFC5285], * Any "a=extmap" lines are parsed as specified in [RFC5285],
Section 5, and their values are stored. Section 5, and their values are stored.
Other attributes that are not relevant to JSEP may also be present, Other attributes that are not relevant to JSEP may also be present,
and implementations SHOULD process any that they recognize. As and implementations SHOULD process any that they recognize. As
required by [RFC4566], Section 5.13, unknown attribute lines MUST be required by [RFC4566], Section 5.13, unknown attribute lines MUST be
ignored. ignored.
Once all the session-level lines have been parsed, processing Once all the session-level lines have been parsed, processing
continues with the lines in m= sections. continues with the lines in "m=" sections.
5.8.2. Media Section Parsing 5.8.2. Media Section Parsing
Like the session-level lines, the media section lines MUST occur in Like the session-level lines, the media section lines MUST occur in
the specific order and with the specific syntax defined in [RFC4566], the specific order and with the specific syntax defined in [RFC4566],
Section 5. Section 5.
The "m=" line itself MUST be parsed as described in [RFC4566], The "m=" line itself MUST be parsed as described in [RFC4566],
Section 5.14, and the media, port, proto, and fmt values stored. Section 5.14, and the <media>, <port>, <proto>, and <fmt> values
stored.
Following the "m=" line, specific processing MUST be applied for the Following the "m=" line, specific processing MUST be applied for the
following non-attribute lines: following non-attribute lines:
o As with the "c=" line at the session level, the "c=" line MUST be * As with the "c=" line at the session level, the "c=" line MUST be
parsed according to [RFC4566], Section 5.7, but its value is not parsed according to [RFC4566], Section 5.7, but its value is not
used. used.
o The "b=" line, if present, MUST be parsed as specified in * The "b=" line, if present, MUST be parsed as specified in
[RFC4566], Section 5.8, and the bwtype and bandwidth values [RFC4566], Section 5.8, and the bwtype and bandwidth values
stored. stored.
Specific processing MUST also be applied for the following attribute Specific processing MUST also be applied for the following attribute
lines: lines:
o If present, a single "a=ice-ufrag" line is parsed as specified in * If present, a single "a=ice-ufrag" line is parsed as specified in
[I-D.ietf-mmusic-ice-sip-sdp], Section 4.4, and the ufrag value is [RFC8839], Section 5.4, and the ufrag value is stored.
stored.
o If present, a single "a=ice-pwd" line is parsed as specified in
[I-D.ietf-mmusic-ice-sip-sdp], Section 4.4, and the password value
is stored.
o If present, a single "a=ice-options" line is parsed as specified * If present, a single "a=ice-pwd" line is parsed as specified in
in [I-D.ietf-mmusic-ice-sip-sdp], Section 4.6, and the set of [RFC8839], Section 5.4, and the password value is stored.
specified options is stored.
o Any "a=candidate" attributes MUST be parsed as specified in * If present, a single "a=ice-options" line is parsed as specified
[I-D.ietf-mmusic-ice-sip-sdp], Section 4.1, and their values in [RFC8839], Section 5.6, and the set of specified options is
stored. stored.
o Any "a=remote-candidates" attributes MUST be parsed as specified * Any "a=candidate" attributes MUST be parsed as specified in
in [I-D.ietf-mmusic-ice-sip-sdp], Section 4.2, but their values [RFC8839], Section 5.1, and their values stored.
are ignored.
o If present, a single "a=end-of-candidates" attribute MUST be * Any "a=remote-candidates" attributes MUST be parsed as specified
parsed as specified in [I-D.ietf-ice-trickle], Section 8.2, and in [RFC8839], Section 5.2, but their values are ignored.
its presence or absence flagged and stored.
o Any "a=fingerprint" lines are parsed as specified in [RFC8122], * If present, a single "a=end-of-candidates" attribute MUST be
parsed as specified in [RFC8840], Section 8.1, and its presence or
absence flagged and stored.
* Any "a=fingerprint" lines are parsed as specified in [RFC8122],
Section 5, and the set of fingerprint and algorithm values is Section 5, and the set of fingerprint and algorithm values is
stored. stored.
If the "m=" proto value indicates use of RTP, as described in If the "m=" <proto> value indicates use of RTP, as described in
Section 5.1.2 above, the following attribute lines MUST be processed: Section 5.1.2 above, the following attribute lines MUST be processed:
o The "m=" fmt value MUST be parsed as specified in [RFC4566], * The "m=" <fmt> value MUST be parsed as specified in [RFC4566],
Section 5.14, and the individual values stored. Section 5.14, and the individual values stored.
o Any "a=rtpmap" or "a=fmtp" lines MUST be parsed as specified in * Any "a=rtpmap" or "a=fmtp" lines MUST be parsed as specified in
[RFC4566], Section 6, and their values stored. [RFC4566], Section 6, and their values stored.
o If present, a single "a=ptime" line MUST be parsed as described in * If present, a single "a=ptime" line MUST be parsed as described in
[RFC4566], Section 6, and its value stored. [RFC4566], Section 6, and its value stored.
o If present, a single "a=maxptime" line MUST be parsed as described * If present, a single "a=maxptime" line MUST be parsed as described
in [RFC4566], Section 6, and its value stored. in [RFC4566], Section 6, and its value stored.
o If present, a single direction attribute line (e.g. "a=sendrecv") * If present, a single direction attribute line (e.g., "a=sendrecv")
MUST be parsed as described in [RFC4566], Section 6, and its value MUST be parsed as described in [RFC4566], Section 6, and its value
stored. stored.
o Any "a=ssrc" attributes MUST be parsed as specified in [RFC5576], * Any "a=ssrc" attributes MUST be parsed as specified in [RFC5576],
Section 4.1, and their values stored. Section 4.1, and their values stored.
o Any "a=extmap" attributes MUST be parsed as specified in * Any "a=extmap" attributes MUST be parsed as specified in
[RFC5285], Section 5, and their values stored. [RFC5285], Section 5, and their values stored.
o Any "a=rtcp-fb" attributes MUST be parsed as specified in * Any "a=rtcp-fb" attributes MUST be parsed as specified in
[RFC4585], Section 4.2., and their values stored. [RFC4585], Section 4.2, and their values stored.
o If present, a single "a=rtcp-mux" attribute MUST be parsed as * If present, a single "a=rtcp-mux" attribute MUST be parsed as
specified in [RFC5761], Section 5.1.3, and its presence or absence specified in [RFC5761], Section 5.1.3, and its presence or absence
flagged and stored. flagged and stored.
o If present, a single "a=rtcp-mux-only" attribute MUST be parsed as * If present, a single "a=rtcp-mux-only" attribute MUST be parsed as
specified in [I-D.ietf-mmusic-mux-exclusive], Section 3, and its specified in [RFC8858], Section 3, and its presence or absence
presence or absence flagged and stored. flagged and stored.
o If present, a single "a=rtcp-rsize" attribute MUST be parsed as * If present, a single "a=rtcp-rsize" attribute MUST be parsed as
specified in [RFC5506], Section 5, and its presence or absence specified in [RFC5506], Section 5, and its presence or absence
flagged and stored. flagged and stored.
o If present, a single "a=rtcp" attribute MUST be parsed as * If present, a single "a=rtcp" attribute MUST be parsed as
specified in [RFC3605], Section 2.1, but its value is ignored, as specified in [RFC3605], Section 2.1, but its value is ignored, as
this information is superfluous when using ICE. this information is superfluous when using ICE.
o If present, "a=msid" attributes MUST be parsed as specified in * If present, "a=msid" attributes MUST be parsed as specified in
[I-D.ietf-mmusic-msid], Section 3.2, and their values stored, [RFC8830], Section 3.2, and their values stored, ignoring any
ignoring any "appdata" field. If no "a=msid" attributes are "appdata" field. If no "a=msid" attributes are present, a random
present, a random msid-id value is generated for a "default" msid-id value is generated for a "default" MediaStream for the
MediaStream for the session, if not already present, and this session, if not already present, and this value is stored.
value is stored.
o Any "a=imageattr" attributes MUST be parsed as specified in * Any "a=imageattr" attributes MUST be parsed as specified in
[RFC6236], Section 3, and their values stored. [RFC6236], Section 3, and their values stored.
o Any "a=rid" lines MUST be parsed as specified in * Any "a=rid" lines MUST be parsed as specified in [RFC8851],
[I-D.ietf-mmusic-rid], Section 10, and their values stored. Section 10, and their values stored.
o If present, a single "a=simulcast" line MUST be parsed as * If present, a single "a=simulcast" line MUST be parsed as
specified in [I-D.ietf-mmusic-sdp-simulcast], and its values specified in [RFC8853], and its values stored.
stored.
Otherwise, if the "m=" proto value indicates use of SCTP, the Otherwise, if the "m=" <proto> value indicates use of SCTP, the
following attribute lines MUST be processed: following attribute lines MUST be processed:
o The "m=" fmt value MUST be parsed as specified in * The "m=" <fmt> value MUST be parsed as specified in [RFC8841],
[I-D.ietf-mmusic-sctp-sdp], Section 4.3, and the application Section 4.3, and the application protocol value stored.
protocol value stored.
o An "a=sctp-port" attribute MUST be present, and it MUST be parsed * An "a=sctp-port" attribute MUST be present, and it MUST be parsed
as specified in [I-D.ietf-mmusic-sctp-sdp], Section 5.2, and the as specified in [RFC8841], Section 5.2, and the value stored.
value stored.
o If present, a single "a=max-message-size" attribute MUST be parsed * If present, a single "a=max-message-size" attribute MUST be parsed
as specified in [I-D.ietf-mmusic-sctp-sdp], Section 6, and the as specified in [RFC8841], Section 6, and the value stored.
value stored. Otherwise, use the specified default. Otherwise, use the specified default.
Other attributes that are not relevant to JSEP may also be present, Other attributes that are not relevant to JSEP may also be present,
and implementations SHOULD process any that they recognize. As and implementations SHOULD process any that they recognize. As
required by [RFC4566], Section 5.13, unknown attribute lines MUST be required by [RFC4566], Section 5.13, unknown attribute lines MUST be
ignored. ignored.
5.8.3. Semantics Verification 5.8.3. Semantics Verification
Assuming parsing completes successfully, the parsed description is Assuming that parsing completes successfully, the parsed description
then evaluated to ensure internal consistency as well as proper is then evaluated to ensure internal consistency as well as proper
support for mandatory features. Specifically, the following checks support for mandatory features. Specifically, the following checks
are performed: are performed:
o For each m= section, valid values for each of the mandatory-to-use * For each "m=" section, valid values for each of the mandatory-to-
features enumerated in Section 5.1.1 MUST be present. These use features enumerated in Section 5.1.1 MUST be present. These
values MAY either be present at the media level, or inherited from values MAY be either present at the media level or inherited from
the session level. the session level.
* ICE ufrag and password values, which MUST comply with the size - ICE ufrag and password values, which MUST comply with the size
limits specified in [I-D.ietf-mmusic-ice-sip-sdp], Section 4.4. limits specified in [RFC8839], Section 5.4.
* tls-id value, which MUST be set according to - A tls-id value, which MUST be set according to [RFC8842],
[I-D.ietf-mmusic-dtls-sdp], Section 5. If this is a re-offer Section 5. If this is a re-offer or a response to a re-offer
or a response to a re-offer and the tls-id value is different and the tls-id value is different from that presently in use,
from that presently in use, the DTLS connection is not being the DTLS connection is not being continued and the remote
continued and the remote description MUST be part of an ICE description MUST be part of an ICE restart, together with new
restart, together with new ufrag and password values. ufrag and password values.
* DTLS setup value, which MUST be set according to the rules - A DTLS setup value, which MUST be set according to the rules
specified in [RFC5763], Section 5 and MUST be consistent with specified in [RFC5763], Section 5 and MUST be consistent with
the selected role of the current DTLS connection, if one exists the selected role of the current DTLS connection, if one exists
and is being continued. and is being continued.
* DTLS fingerprint values, where at least one fingerprint MUST be - DTLS fingerprint values, where at least one fingerprint MUST be
present. present.
o All RID values referenced in an "a=simulcast" line MUST exist as * All rid-ids referenced in an "a=simulcast" line MUST exist as
"a=rid" lines. "a=rid" lines.
o Each m= section is also checked to ensure prohibited features are * Each "m=" section is also checked to ensure that prohibited
not used. features are not used.
o If the RTP/RTCP multiplexing policy is "require", each m= section * If the RTP/RTCP multiplexing policy is "require", each "m="
MUST contain an "a=rtcp-mux" attribute. If an m= section contains section MUST contain an "a=rtcp-mux" attribute. If an "m="
an "a=rtcp-mux-only" attribute, that section MUST also contain an section contains an "a=rtcp-mux-only" attribute, that section MUST
"a=rtcp-mux" attribute. also contain an "a=rtcp-mux" attribute.
o If an m= section was present in the previous answer, the state of * If an "m=" section was present in the previous answer, the state
RTP/RTCP multiplexing MUST match what was previously negotiated. of RTP/RTCP multiplexing MUST match what was previously
negotiated.
If this session description is of type "pranswer" or "answer", the If this session description is of type "pranswer" or "answer", the
following additional checks are applied: following additional checks are applied:
o The session description must follow the rules defined in * The session description MUST follow the rules defined in
[RFC3264], Section 6, including the requirement that the number of [RFC3264], Section 6, including the requirement that the number of
m= sections MUST exactly match the number of m= sections in the "m=" sections MUST exactly match the number of "m=" sections in
associated offer. the associated offer.
o For each m= section, the media type and protocol values MUST * For each "m=" section, the media type and protocol values MUST
exactly match the media type and protocol values in the exactly match the media type and protocol values in the
corresponding m= section in the associated offer. corresponding "m=" section in the associated offer.
If any of the preceding checks failed, processing MUST stop and an If any of the preceding checks failed, processing MUST stop and an
error MUST be returned. error MUST be returned.
5.9. Applying a Local Description 5.9. Applying a Local Description
The following steps are performed at the media engine level to apply The following steps are performed at the media engine level to apply
a local description. If an error is returned, the session MUST be a local description. If an error is returned, the session MUST be
restored to the state it was in before performing these steps. restored to the state it was in before performing these steps.
First, m= sections are processed. For each m= section, the following First, "m=" sections are processed. For each "m=" section, the
steps MUST be performed; if any parameters are out of bounds, or following steps MUST be performed; if any parameters are out of
cannot be applied, processing MUST stop and an error MUST be bounds or cannot be applied, processing MUST stop and an error MUST
returned. be returned.
o If this m= section is new, begin gathering candidates for it, as * If this "m=" section is new, begin gathering candidates for it, as
defined in [RFC8445], Section 5.1.1, unless it is definitively defined in [RFC8445], Section 5.1.1, unless it is definitively
being bundled (either this is an offer and the m= section is being bundled (either (1) this is an offer and the "m=" section is
marked bundle-only, or it is an answer and the m= section is marked bundle-only or (2) it is an answer and the "m=" section is
bundled into into another m= section.) bundled into another "m=" section).
o Or, if the ICE ufrag and password values have changed, trigger the * Or, if the ICE ufrag and password values have changed, trigger the
ICE agent to start an ICE restart as described in [RFC8445], ICE agent to start an ICE restart as described in [RFC8445],
Section 9, and begin gathering new candidates for the m= section. Section 9, and begin gathering new candidates for the "m="
If this description is an answer, also start checks on that media section. If this description is an answer, also start checks on
section. that media section.
o If the m= section proto value indicates use of RTP: * If the "m=" section <proto> value indicates use of RTP:
* If there is no RtpTransceiver associated with this m= section, - If there is no RtpTransceiver associated with this "m="
find one and associate it with this m= section according to the section, find one and associate it with this "m=" section
following steps. Note that this situation will only occur when according to the following steps. Note that this situation
applying an offer. will only occur when applying an offer.
+ Find the RtpTransceiver that corresponds to this m= section, o Find the RtpTransceiver that corresponds to this "m="
using the mapping between transceivers and m= section section, using the mapping between transceivers and "m="
indices established when creating the offer. section indices established when creating the offer.
+ Set the value of this RtpTransceiver's mid property to the o Set the value of this RtpTransceiver's mid property to the
MID of the m= section. MID of the "m=" section.
* If RTCP mux is indicated, prepare to demux RTP and RTCP from - If RTCP mux is indicated, prepare to demux RTP and RTCP from
the RTP ICE component, as specified in [RFC5761], the RTP ICE component, as specified in [RFC5761],
Section 5.1.3. Section 5.1.3.
* For each specified RTP header extension, establish a mapping - For each specified RTP header extension, establish a mapping
between the extension ID and URI, as described in [RFC5285], between the extension ID and URI, as described in [RFC5285],
Section 6. Section 6.
* If the MID header extension is supported, prepare to demux RTP - If the MID header extension is supported, prepare to demux RTP
streams intended for this m= section based on the MID header streams intended for this "m=" section based on the MID header
extension, as described in extension, as described in [RFC8843], Section 15.
[I-D.ietf-mmusic-sdp-bundle-negotiation], Section 15.
* For each specified media format, establish a mapping between - For each specified media format, establish a mapping between
the payload type and the actual media format, as described in the payload type and the actual media format, as described in
[RFC3264], Section 6.1. In addition, prepare to demux RTP [RFC3264], Section 6.1. In addition, prepare to demux RTP
streams intended for this m= section based on the media formats streams intended for this "m=" section based on the media
supported by this m= section, as described in formats supported by this "m=" section, as described in
[I-D.ietf-mmusic-sdp-bundle-negotiation], Section 10.2. [RFC8843], Section 9.2.
* For each specified "rtx" media format, establish a mapping - For each specified "rtx" media format, establish a mapping
between the RTX payload type and its associated primary payload between the RTX payload type and its associated primary payload
type, as described in [RFC4588], Sections 8.6 and 8.7. type, as described in Sections 8.6 and 8.7 of [RFC4588].
* If the directional attribute is of type "sendrecv" or - If the direction attribute is of type "sendrecv" or "recvonly",
"recvonly", enable receipt and decoding of media. enable receipt and decoding of media.
Finally, if this description is of type "pranswer" or "answer", Finally, if this description is of type "pranswer" or "answer",
follow the processing defined in Section 5.11 below. follow the processing defined in Section 5.11 below.
5.10. Applying a Remote Description 5.10. Applying a Remote Description
The following steps are performed to apply a remote description. If The following steps are performed to apply a remote description. If
an error is returned, the session MUST be restored to the state it an error is returned, the session MUST be restored to the state it
was in before performing these steps. was in before performing these steps.
If the answer contains any "a=ice-options" attributes where "trickle" If the answer contains any "a=ice-options" attributes where "trickle"
is listed as an attribute, update the PeerConnection canTrickle is listed as an attribute, update the PeerConnection
property to be true. Otherwise, set this property to false. canTrickleIceCandidates property to be "true". Otherwise, set this
property to "false".
The following steps MUST be performed for attributes at the session The following steps MUST be performed for attributes at the session
level; if any parameters are out of bounds, or cannot be applied, level; if any parameters are out of bounds or cannot be applied,
processing MUST stop and an error MUST be returned. processing MUST stop and an error MUST be returned.
o For any specified "CT" bandwidth value, set this as the limit for * For any specified "CT" bandwidth value, set this value as the
the maximum total bitrate for all m= sections, as specified in limit for the maximum total bitrate for all "m=" sections, as
[RFC4566], Section 5.8. Within this overall limit, the specified in [RFC4566], Section 5.8. Within this overall limit,
implementation can dynamically decide how to best allocate the the implementation can dynamically decide how to best allocate the
available bandwidth between m= sections, respecting any specific available bandwidth between "m=" sections, respecting any specific
limits that have been specified for individual m= sections. limits that have been specified for individual "m=" sections.
o For any specified "RR" or "RS" bandwidth values, handle as * For any specified "RR" or "RS" bandwidth values, handle as
specified in [RFC3556], Section 2. specified in [RFC3556], Section 2.
o Any "AS" bandwidth value MUST be ignored, as the meaning of this * Any "AS" bandwidth value ([RFC4566], Section 5.8) MUST be ignored,
construct at the session level is not well defined. as the meaning of this construct at the session level is not well
defined.
For each m= section, the following steps MUST be performed; if any For each "m=" section, the following steps MUST be performed; if any
parameters are out of bounds, or cannot be applied, processing MUST parameters are out of bounds or cannot be applied, processing MUST
stop and an error MUST be returned. stop and an error MUST be returned.
o If the ICE ufrag or password changed from the previous remote * If the ICE ufrag or password changed from the previous remote
description: description:
* If the description is of type "offer", the implementation MUST - If the description is of type "offer", the implementation MUST
note that an ICE restart is needed, as described in note that an ICE restart is needed, as described in [RFC8839],
[I-D.ietf-mmusic-ice-sip-sdp], Section 3.4.1.1.1 Section 4.4.1.1.1.
* If the description is of type "answer" or "pranswer", then - If the description is of type "answer" or "pranswer", then
check to see if the current local description is an ICE check to see if the current local description is an ICE
restart, and if not, generate an error. If the PeerConnection restart, and if not, generate an error. If the PeerConnection
state is "have-remote-pranswer", and the ICE ufrag or password state is "have-remote-pranswer" and the ICE ufrag or password
changed from the previous provisional answer, then signal the changed from the previous provisional answer, then signal the
ICE agent to discard any previous ICE check list state for the ICE agent to discard any previous ICE checklist state for the
m= section. Finally, signal the ICE agent to begin checks. "m=" section. Finally, signal the ICE agent to begin checks.
o If the current local description indicates an ICE restart, and * If the current local description indicates an ICE restart but
either the ICE ufrag or password has not changed from the previous neither the ICE ufrag nor the password has changed from the
remote description, as prescribed by [RFC8445], Section 9, previous remote description (as prescribed by [RFC8445],
generate an error. Section 9), generate an error.
o Configure the ICE components associated with this media section to * Configure the ICE components associated with this media section to
use the supplied ICE remote ufrag and password for their use the supplied ICE remote ufrag and password for their
connectivity checks. connectivity checks.
o Pair any supplied ICE candidates with any gathered local * Pair any supplied ICE candidates with any gathered local
candidates, as described in [RFC8445], Section 6.1.2, and start candidates, as described in [RFC8445], Section 6.1.2, and start
connectivity checks with the appropriate credentials. connectivity checks with the appropriate credentials.
o If an "a=end-of-candidates" attribute is present, process the end- * If an "a=end-of-candidates" attribute is present, process the end-
of-candidates indication as described in [I-D.ietf-ice-trickle], of-candidates indication as described in [RFC8838], Section 14.
Section 11.
o If the m= section proto value indicates use of RTP: * If the "m=" section <proto> value indicates use of RTP:
* If the m= section is being recycled (see Section 5.2.2), - If the "m=" section is being recycled (see Section 5.2.2),
dissociate the currently associated RtpTransceiver by setting disassociate the currently associated RtpTransceiver by setting
its mid property to null, and discard the mapping between the its mid property to "null", and discard the mapping between the
transceiver and its m= section index. transceiver and its "m=" section index.
* If the m= section is not associated with any RtpTransceiver - If the "m=" section is not associated with any RtpTransceiver
(possibly because it was dissociated in the previous step), (possibly because it was disassociated in the previous step),
either find an RtpTransceiver or create one according to the either find an RtpTransceiver or create one according to the
following steps: following steps:
+ If the m= section is sendrecv or recvonly, and there are o If the "m=" section is sendrecv or recvonly, and there are
RtpTransceivers of the same type that were added to the RtpTransceivers of the same type that were added to the
PeerConnection by addTrack and are not associated with any PeerConnection by addTrack and are not associated with any
m= section and are not stopped, find the first (according to "m=" section and are not stopped, find the first (according
the canonical order described in Section 5.2.1) such to the canonical order described in Section 5.2.1) such
RtpTransceiver. RtpTransceiver.
+ If no RtpTransceiver was found in the previous step, create o If no RtpTransceiver was found in the previous step, create
one with a recvonly direction. one with a recvonly direction.
+ Associate the found or created RtpTransceiver with the m= o Associate the found or created RtpTransceiver with the "m="
section by setting the value of the RtpTransceiver's mid section by setting the value of the RtpTransceiver's mid
property to the MID of the m= section, and establish a property to the MID of the "m=" section, and establish a
mapping between the transceiver and the index of the m= mapping between the transceiver and the index of the "m="
section. If the m= section does not include a MID (i.e., section. If the "m=" section does not include a MID (i.e.,
the remote endpoint does not support the MID extension), the remote endpoint does not support the MID extension),
generate a value for the RtpTransceiver mid property, generate a value for the RtpTransceiver mid property,
following the guidance for "a=mid" mentioned in following the guidance for "a=mid" mentioned in
Section 5.2.1. Section 5.2.1.
* For each specified media format that is also supported by the - For each specified media format that is also supported by the
local implementation, establish a mapping between the specified local implementation, establish a mapping between the specified
payload type and the media format, as described in [RFC3264], payload type and the media format, as described in [RFC3264],
Section 6.1. Specifically, this means that the implementation Section 6.1. Specifically, this means that the implementation
records the payload type to be used in outgoing RTP packets records the payload type to be used in outgoing RTP packets
when sending each specified media format, as well as the when sending each specified media format, as well as the
relative preference for each format that is indicated in their relative preference for each format that is indicated in their
ordering. If any indicated media format is not supported by ordering. If any indicated media format is not supported by
the local implementation, it MUST be ignored. the local implementation, it MUST be ignored.
* For each specified "rtx" media format, establish a mapping - For each specified "rtx" media format, establish a mapping
between the RTX payload type and its associated primary payload between the RTX payload type and its associated primary payload
type, as described in [RFC4588], Section 4. If any referenced type, as described in [RFC4588], Section 4. If any referenced
primary payload types are not present, this MUST result in an primary payload types are not present, this MUST result in an
error. Note that RTX payload types may refer to primary error. Note that RTX payload types may refer to primary
payload types which are not supported by the local media payload types that are not supported by the local media
implementation, in which case, the RTX payload type MUST also implementation, in which case the RTX payload type MUST also be
be ignored. ignored.
* For each specified fmtp parameter that is supported by the - For each specified fmtp parameter that is supported by the
local implementation, enable them on the associated media local implementation, enable them on the associated media
formats. formats.
* For each specified SSRC that is signaled in the m= section, - For each specified Synchronization Source (SSRC) that is
prepare to demux RTP streams intended for this m= section using signaled in the "m=" section, prepare to demux RTP streams
that SSRC, as described in intended for this "m=" section using that SSRC, as described in
[I-D.ietf-mmusic-sdp-bundle-negotiation], Section 10.2. [RFC8843], Section 9.2.
* For each specified RTP header extension that is also supported - For each specified RTP header extension that is also supported
by the local implementation, establish a mapping between the by the local implementation, establish a mapping between the
extension ID and URI, as described in [RFC5285], Section 5. extension ID and URI, as described in [RFC5285], Section 5.
Specifically, this means that the implementation records the Specifically, this means that the implementation records the
extension ID to be used in outgoing RTP packets when sending extension ID to be used in outgoing RTP packets when sending
each specified header extension. If any indicated RTP header each specified header extension. If any indicated RTP header
extension is not supported by the local implementation, it MUST extension is not supported by the local implementation, it MUST
be ignored. be ignored.
* For each specified RTCP feedback mechanism that is supported by - For each specified RTCP feedback mechanism that is supported by
the local implementation, enable them on the associated media the local implementation, enable them on the associated media
formats. formats.
* For any specified "TIAS" bandwidth value, set this value as a - For any specified "TIAS" ("Transport Independent Application
Specific Maximum") bandwidth value, set this value as a
constraint on the maximum RTP bitrate to be used when sending constraint on the maximum RTP bitrate to be used when sending
media, as specified in [RFC3890]. If a "TIAS" value is not media, as specified in [RFC3890]. If a "TIAS" value is not
present, but an "AS" value is specified, generate a "TIAS" present but an "AS" value is specified, generate a "TIAS" value
value using this formula: using this formula:
TIAS = AS * 1000 * 0.95 - (50 * 40 * 8) TIAS = AS * 1000 * 0.95 - (50 * 40 * 8)
The 50 is based on 50 packets per second, the 40 is based on an The 1000 changes the unit from kbps to bps (as required by
estimate of total header size, the 1000 changes the unit from TIAS), and the 0.95 is to allocate 5% to RTCP. An estimate of
kbps to bps (as required by TIAS), and the 0.95 is to allocate header overhead is then subtracted out, in which the 50 is
5% to RTCP. "TIAS" is used in preference to "AS" because it based on 50 packets per second, the 40 is based on typical
provides more accurate control of bandwidth. header size (in bytes), and the 8 converts bytes to bits. Note
that "TIAS" is preferred over "AS" because it provides more
accurate control of bandwidth.
* For any "RR" or "RS" bandwidth values, handle as specified in - For any "RR" or "RS" bandwidth values, handle as specified in
[RFC3556], Section 2. [RFC3556], Section 2.
* Any specified "CT" bandwidth value MUST be ignored, as the - Any specified "CT" bandwidth value MUST be ignored, as the
meaning of this construct at the media level is not well meaning of this construct at the media level is not well
defined. defined.
* If the m= section is of type audio: - If the "m=" section is of type "audio":
+ For each specified "CN" media format, configure silence o For each specified "CN" media format, configure silence
suppression for all supported media formats with the same suppression for all supported media formats with the same
clockrate, as described in [RFC3389], Section 5, except for clock rate, as described in [RFC3389], Section 5, except for
formats that have their own internal silence suppression formats that have their own internal silence suppression
mechanisms. Silence suppression for such formats (e.g., mechanisms. Silence suppression for such formats (e.g.,
Opus) is controlled via fmtp parameters, as discussed in Opus) is controlled via fmtp parameters, as discussed in
Section 5.2.3.2. Section 5.2.3.2.
+ For each specified "telephone-event" media format, enable o For each specified "telephone-event" media format, enable
DTMF transmission for all supported media formats with the dual-tone multifrequency (DTMF) transmission for all
same clockrate, as described in [RFC4733], Section 2.5.1.2. supported media formats with the same clock rate, as
If there are any supported media formats that do not have a described in [RFC4733], Section 2.5.1.2. If there are any
corresponding telephone-event format, disable DTMF supported media formats that do not have a corresponding
transmission for those formats. telephone-event format, disable DTMF transmission for those
formats.
+ For any specified "ptime" value, configure the available o For any specified "ptime" value, configure the available
media formats to use the specified packet size when sending. media formats to use the specified packet size when sending.
If the specified size is not supported for a media format, If the specified size is not supported for a media format,
use the next closest value instead. use the next closest value instead.
Finally, if this description is of type "pranswer" or "answer", Finally, if this description is of type "pranswer" or "answer",
follow the processing defined in Section 5.11 below. follow the processing defined in Section 5.11 below.
5.11. Applying an Answer 5.11. Applying an Answer
In addition to the steps mentioned above for processing a local or In addition to the steps mentioned above for processing a local or
remote description, the following steps are performed when processing remote description, the following steps are performed when processing
a description of type "pranswer" or "answer". a description of type "pranswer" or "answer".
For each m= section, the following steps MUST be performed: For each "m=" section, the following steps MUST be performed:
o If the m= section has been rejected (i.e. port is set to zero in * If the "m=" section has been rejected (i.e., the <port> value is
the answer), stop any reception or transmission of media for this set to zero in the answer), stop any reception or transmission of
section, and, unless a non-rejected m= section is bundled with media for this section, and, unless a non-rejected "m=" section is
this m= section, discard any associated ICE components, as bundled with this "m=" section, discard any associated ICE
described in [I-D.ietf-mmusic-ice-sip-sdp], Section 3.4.3.1. components, as described in [RFC8839], Section 4.4.3.1.
o If the remote DTLS fingerprint has been changed or the tls-id has * If the remote DTLS fingerprint has been changed or the value of
changed, tear down the DTLS connection. This includes the case the "a=tls-id" attribute has changed, tear down the DTLS
when the PeerConnection state is "have-remote-pranswer". If a connection. This includes the case when the PeerConnection state
DTLS connection needs to be torn down but the answer does not is "have-remote-pranswer". If a DTLS connection needs to be torn
indicate an ICE restart or, in the case of "have-remote-pranswer", down but the answer does not indicate an ICE restart or, in the
new ICE credentials, an error MUST be generated. If an ICE case of "have-remote-pranswer", new ICE credentials, an error MUST
restart is performed without a change in tls-id or fingerprint, be generated. If an ICE restart is performed without a change in
then the same DTLS connection is continued over the new ICE the tls-id value or fingerprint, then the same DTLS connection is
channel. Note that although JSEP requires that answerers change continued over the new ICE channel. Note that although JSEP
the tls-id value if and only if the offerer does, non-JSEP requires that answerers change the tls-id value if and only if the
answerers are permitted to change the tls-id as long as the offer offerer does, non-JSEP answerers are permitted to change the tls-
contained an ICE restart. Thus, JSEP implementations which id value as long as the offer contained an ICE restart. Thus,
process DTLS data prior to receiving an answer MUST be prepared to JSEP implementations that process DTLS data prior to receiving an
receive either a ClientHello or data from the previous DTLS answer MUST be prepared to receive either a ClientHello or data
connection. from the previous DTLS connection.
o If no valid DTLS connection exists, prepare to start a DTLS * If no valid DTLS connection exists, prepare to start a DTLS
connection, using the specified roles and fingerprints, on any connection, using the specified roles and fingerprints, on any
underlying ICE components, once they are active. underlying ICE components, once they are active.
o If the m= section proto value indicates use of RTP: * If the "m=" section <proto> value indicates use of RTP:
* If the m= section references RTCP feedback mechanisms that were - If the "m=" section references RTCP feedback mechanisms that
not present in the corresponding m= section in the offer, this were not present in the corresponding "m=" section in the
indicates a negotiation problem and MUST result in an error. offer, this indicates a negotiation problem and MUST result in
However, new media formats and new RTP header extension values an error. However, new media formats and new RTP header
are permitted in the answer, as described in [RFC3264], extension values are permitted in the answer, as described in
Section 7, and [RFC5285], Section 6. [RFC3264], Section 7 and [RFC5285], Section 6.
* If the m= section has RTCP mux enabled, discard the RTCP ICE - If the "m=" section has RTCP mux enabled, discard the RTCP ICE
component, if one exists, and begin or continue muxing RTCP component, if one exists, and begin or continue muxing RTCP
over the RTP ICE component, as specified in [RFC5761], over the RTP ICE component, as specified in [RFC5761],
Section 5.1.3. Otherwise, prepare to transmit RTCP over the Section 5.1.3. Otherwise, prepare to transmit RTCP over the
RTCP ICE component; if no RTCP ICE component exists, because RTCP ICE component; if no RTCP ICE component exists because
RTCP mux was previously enabled, this MUST result in an error. RTCP mux was previously enabled, this MUST result in an error.
* If the m= section has reduced-size RTCP enabled, configure the - If the "m=" section has Reduced-Size RTCP enabled, configure
RTCP transmission for this m= section to use reduced-size RTCP, the RTCP transmission for this "m=" section to use Reduced-Size
as specified in [RFC5506]. RTCP, as specified in [RFC5506].
* If the directional attribute in the answer indicates that the - If the direction attribute in the answer indicates that the
JSEP implementation should be sending media ("sendonly" for JSEP implementation should be sending media ("sendonly" for
local answers, "recvonly" for remote answers, or "sendrecv" for local answers, "recvonly" for remote answers, or "sendrecv" for
either type of answer), choose the media format to send as the either type of answer), choose the media format to send as the
most preferred media format from the remote description that is most preferred media format from the remote description that is
also locally supported, as discussed in [RFC3264], Sections 6.1 also locally supported, as discussed in Sections 6.1 and 7 of
and 7, and start transmitting RTP media using that format once [RFC3264], and start transmitting RTP media using that format
the underlying transport layers have been established. If an once the underlying transport layers have been established. If
SSRC has not already been chosen for this outgoing RTP stream, an SSRC has not already been chosen for this outgoing RTP
choose a random one. If media is already being transmitted, stream, choose a unique random one. If media is already being
the same SSRC SHOULD be used unless the clockrate of the new transmitted, the same SSRC SHOULD be used unless the clock rate
codec is different, in which case a new SSRC MUST be chosen, as of the new codec is different, in which case a new SSRC MUST be
specified in [RFC7160], Section 3.1. chosen, as specified in [RFC7160], Section 4.1.
* The payload type mapping from the remote description is used to - The payload type mapping from the remote description is used to
determine payload types for the outgoing RTP streams, including determine payload types for the outgoing RTP streams, including
the payload type for the send media format chosen above. Any the payload type for the send media format chosen above. Any
RTP header extensions that were negotiated should be included RTP header extensions that were negotiated should be included
in the outgoing RTP streams, using the extension mapping from in the outgoing RTP streams, using the extension mapping from
the remote description; if the RID header extension has been the remote description. If the MID header extension has been
negotiated, and RID values are specified, include the RID negotiated, include it in the outgoing RTP streams, as
header extension in the outgoing RTP streams, as indicated in indicated in [RFC8843], Section 15. If the RtpStreamId or
[I-D.ietf-mmusic-rid], Section 4. RepairedRtpStreamId header extensions have been negotiated and
rid-ids have been established, include these header extensions
in the outgoing RTP streams, as indicated in [RFC8851],
Section 4.
* If the m= section is of type audio, and silence suppression was - If the "m=" section is of type "audio", and silence suppression
configured for the send media format as a result of processing was (1) configured for the send media format as a result of
the remote description, and is also enabled for that format in processing the remote description and (2) also enabled for that
the local description, use silence suppression for outgoing format in the local description, use silence suppression for
media, in accordance with the guidance in Section 5.2.3.2. If outgoing media, in accordance with the guidance in
these conditions are not met, silence suppression MUST NOT be Section 5.2.3.2. If these conditions are not met, silence
used for outgoing media. suppression MUST NOT be used for outgoing media.
* If simulcast has been negotiated, send the number of Source RTP - If simulcast has been negotiated, send the appropriate number
Streams as specified in [I-D.ietf-mmusic-sdp-simulcast], of Source RTP Streams as specified in [RFC8853], Section 5.3.3.
Section 6.2.2.
* If the send media format chosen above has a corresponding "rtx" - If the send media format chosen above has a corresponding "rtx"
media format, or a FEC mechanism has been negotiated, establish media format or a FEC mechanism has been negotiated, establish
a Redundancy RTP Stream with a random SSRC for each Source RTP a redundancy RTP stream with a unique random SSRC for each
Stream, and start or continue transmitting RTX/FEC packets as Source RTP Stream, and start or continue transmitting RTX/FEC
needed. packets as needed.
* If the send media format chosen above has a corresponding "red" - If the send media format chosen above has a corresponding "red"
media format of the same clockrate, allow redundant encoding media format of the same clock rate, allow redundant encoding
using the specified format for resiliency purposes, as using the specified format for resiliency purposes, as
discussed in [I-D.ietf-rtcweb-fec], Section 3.2. Note that discussed in [RFC8854], Section 3.2. Note that unlike RTX or
unlike RTX or FEC media formats, the "red" format is FEC media formats, the "red" format is transmitted on the
transmitted on the Source RTP Stream, not the Redundancy RTP Source RTP Stream, not the redundancy RTP stream.
Stream.
* Enable the RTCP feedback mechanisms referenced in the media - Enable the RTCP feedback mechanisms referenced in the media
section for all Source RTP Streams using the specified media section for all Source RTP Streams using the specified media
formats. Specifically, begin or continue sending the requested formats. Specifically, begin or continue sending the requested
feedback types and reacting to received feedback, as specified feedback types and reacting to received feedback, as specified
in [RFC4585], Section 4.2. When sending RTCP feedback, follow in [RFC4585], Section 4.2. When sending RTCP feedback, follow
the rules and recommendations from [RFC8108] Section 5.4.1, to the rules and recommendations from [RFC8108], Section 5.4.1 to
select which SSRC to use. select which SSRC to use.
* If the directional attribute in the answer indicates that the - If the direction attribute in the answer indicates that the
JSEP implementation should not be sending media ("recvonly" for JSEP implementation should not be sending media ("recvonly" for
local answers, "sendonly" for remote answers, or "inactive" for local answers, "sendonly" for remote answers, or "inactive" for
either type of answer) stop transmitting all RTP media, but either type of answer), stop transmitting all RTP media, but
continue sending RTCP, as described in [RFC3264], Section 5.1. continue sending RTCP, as described in [RFC3264], Section 5.1.
o If the m= section proto value indicates use of SCTP: * If the "m=" section <proto> value indicates use of SCTP:
* If an SCTP association exists, and the remote SCTP port has - If an SCTP association exists and the remote SCTP port has
changed, discard the existing SCTP association. This includes changed, discard the existing SCTP association. This includes
the case when the PeerConnection state is "have-remote- the case when the PeerConnection state is "have-remote-
pranswer". pranswer".
* If no valid SCTP association exists, prepare to initiate a SCTP - If no valid SCTP association exists, prepare to initiate an
association over the associated ICE component and DTLS SCTP association over the associated ICE component and DTLS
connection, using the local SCTP port value from the local connection, using the local SCTP port value from the local
description, and the remote SCTP port value from the remote description and the remote SCTP port value from the remote
description, as described in [I-D.ietf-mmusic-sctp-sdp], description, as described in [RFC8841], Section 10.2.
Section 10.2.
If the answer contains valid bundle groups, discard any ICE If the answer contains valid bundle groups, discard any ICE
components for the m= sections that will be bundled onto the primary components for the "m=" sections that will be bundled onto the
ICE components in each bundle, and begin muxing these m= sections primary ICE components in each bundle, and begin muxing these "m="
accordingly, as described in sections accordingly, as described in [RFC8843], Section 7.4.
[I-D.ietf-mmusic-sdp-bundle-negotiation], Section 8.2.
If the description is of type "answer", and there are still remaining If the description is of type "answer" and there are still remaining
candidates in the ICE candidate pool, discard them. candidates in the ICE candidate pool, discard them.
6. Processing RTP/RTCP 6. Processing RTP/RTCP
When bundling, associating incoming RTP/RTCP with the proper m= When bundling, associating incoming RTP/RTCP with the proper "m="
section is defined in [I-D.ietf-mmusic-sdp-bundle-negotiation], section is defined in [RFC8843], Section 9.2. When not bundling, the
Section 10.2. When not bundling, the proper m= section is clear from proper "m=" section is clear from the ICE component over which the
the ICE component over which the RTP/RTCP is received. RTP/RTCP is received.
Once the proper m= section(s) are known, RTP/RTCP is delivered to the Once the proper "m=" section or sections are known, RTP/RTCP is
RtpTransceiver(s) associated with the m= section(s) and further delivered to the RtpTransceiver(s) associated with the "m="
processing of the RTP/RTCP is done at the RtpTransceiver level. This section(s) and further processing of the RTP/RTCP is done at the
includes using RID [I-D.ietf-mmusic-rid] to distinguish between RtpTransceiver level. This includes using the RID mechanism
multiple Encoded Streams, as well as determine which Source RTP [RFC8851] and its associated RtpStreamId and RepairedRtpStreamId
stream should be repaired by a given Redundancy RTP stream. identifiers to distinguish between multiple encoded streams and
determine which Source RTP stream should be repaired by a given
redundancy RTP stream.
7. Examples 7. Examples
Note that this example section shows several SDP fragments. To Note that this example section shows several SDP fragments. To
format in 72 columns, some of the lines in SDP have been split into accommodate RFC line-length restrictions, some of the SDP lines have
multiple lines, where leading whitespace indicates that a line is a been split into multiple lines, where leading whitespace indicates
continuation of the previous line. In addition, some blank lines that a line is a continuation of the previous line. In addition,
have been added to improve readability but are not valid in SDP. some blank lines have been added to improve readability but are not
valid in SDP.
More examples of SDP for WebRTC call flows, including examples with More examples of SDP for WebRTC call flows, including examples with
IPv6 addresses, can be found in [I-D.ietf-rtcweb-sdp]. IPv6 addresses, can be found in [SDP4WebRTC].
7.1. Simple Example 7.1. Simple Example
This section shows a very simple example that sets up a minimal audio This section shows a very simple example that sets up a minimal
/ video call between two JSEP endpoints without using trickle ICE. audio/video call between two JSEP endpoints without using Trickle
The example in the following section provides a more detailed example ICE. The example in the following section provides a more detailed
of what could happen in a JSEP session. example of what could happen in a JSEP session.
The code flow below shows Alice's endpoint initiating the session to The code flow below shows Alice's endpoint initiating the session to
Bob's endpoint. The messages from the JavaScript application in Bob's endpoint. The messages from the JavaScript application in
Alice's browser to the JavaScript in Bob's browser, abbreviated as Alice's browser to the JavaScript in Bob's browser, abbreviated as
AliceJS and BobJS respectively, are assumed to flow over some "AliceJS" and "BobJS", respectively, are assumed to flow over some
signaling protocol via a web server. The JavaScript on both Alice's signaling protocol via a web server. The JavaScript on both Alice's
side and Bob's side waits for all candidates before sending the offer side and Bob's side waits for all candidates before sending the offer
or answer, so the offers and answers are complete; trickle ICE is not or answer, so the offers and answers are complete; Trickle ICE is not
used. The user agents (JSEP implementations) in Alice and Bob's used. The user agents (JSEP implementations) in Alice's and Bob's
browsers, abbreviated as AliceUA and BobUA respectively, are using browsers, abbreviated as "AliceUA" and "BobUA", respectively, are
the default bundle policy of "balanced", and the default RTCP mux both using the default bundle policy of "balanced" and the default
policy of "require". RTCP mux policy of "require".
// set up local media state // set up local media state
AliceJS->AliceUA: create new PeerConnection AliceJS->AliceUA: create new PeerConnection
AliceJS->AliceUA: addTrack with two tracks: audio and video AliceJS->AliceUA: addTrack with two tracks: audio and video
AliceJS->AliceUA: createOffer to get offer AliceJS->AliceUA: createOffer to get offer
AliceJS->AliceUA: setLocalDescription with offer AliceJS->AliceUA: setLocalDescription with offer
AliceUA->AliceJS: multiple onicecandidate events with candidates AliceUA->AliceJS: multiple onicecandidate events with candidates
// wait for ICE gathering to complete // wait for ICE gathering to complete
AliceUA->AliceJS: onicecandidate event with null candidate AliceUA->AliceJS: onicecandidate event with null candidate
AliceJS->AliceUA: get |offer-A1| from pendingLocalDescription AliceJS->AliceUA: get |offer-A1| from pendingLocalDescription
// |offer-A1| is sent over signaling protocol to Bob // |offer-A1| is sent over signaling protocol to Bob
AliceJS->WebServer: signaling with |offer-A1| AliceJS->WebServer: signaling with |offer-A1|
WebServer->BobJS: signaling with |offer-A1| WebServer->BobJS: signaling with |offer-A1|
// |offer-A1| arrives at Bob // |offer-A1| arrives at Bob
BobJS->BobUA: create a PeerConnection BobJS->BobUA: create a PeerConnection
BobJS->BobUA: setRemoteDescription with |offer-A1| BobJS->BobUA: setRemoteDescription with |offer-A1|
BobUA->BobJS: ontrack events for audio and video tracks BobUA->BobJS: ontrack events for audio and video tracks
// Bob accepts call // Bob accepts call
BobJS->BobUA: addTrack with local tracks BobJS->BobUA: addTrack with local tracks
BobJS->BobUA: createAnswer BobJS->BobUA: createAnswer
BobJS->BobUA: setLocalDescription with answer BobJS->BobUA: setLocalDescription with answer
BobUA->BobJS: multiple onicecandidate events with candidates BobUA->BobJS: multiple onicecandidate events with candidates
// wait for ICE gathering to complete // wait for ICE gathering to complete
BobUA->BobJS: onicecandidate event with null candidate BobUA->BobJS: onicecandidate event with null candidate
BobJS->BobUA: get |answer-A1| from currentLocalDescription BobJS->BobUA: get |answer-A1| from currentLocalDescription
// |answer-A1| is sent over signaling protocol to Alice // |answer-A1| is sent over signaling protocol
BobJS->WebServer: signaling with |answer-A1| // to Alice
WebServer->AliceJS: signaling with |answer-A1| BobJS->WebServer: signaling with |answer-A1|
WebServer->AliceJS: signaling with |answer-A1|
// |answer-A1| arrives at Alice // |answer-A1| arrives at Alice
AliceJS->AliceUA: setRemoteDescription with |answer-A1| AliceJS->AliceUA: setRemoteDescription with |answer-A1|
AliceUA->AliceJS: ontrack events for audio and video tracks AliceUA->AliceJS: ontrack events for audio and video tracks
// media flows // media flows
BobUA->AliceUA: media sent from Bob to Alice BobUA->AliceUA: media sent from Bob to Alice
AliceUA->BobUA: media sent from Alice to Bob AliceUA->BobUA: media sent from Alice to Bob
The SDP for |offer-A1| looks like: The SDP for |offer-A1| looks like:
v=0 v=0
o=- 4962303333179871722 1 IN IP4 0.0.0.0 o=- 4962303333179871722 1 IN IP4 0.0.0.0
s=- s=-
t=0 0 t=0 0
a=ice-options:trickle ice2 a=ice-options:trickle ice2
a=group:BUNDLE a1 v1 a=group:BUNDLE a1 v1
a=group:LS a1 v1 a=group:LS a1 v1
skipping to change at page 76, line 48 skipping to change at line 3614
m=video 10102 UDP/TLS/RTP/SAVPF 100 101 102 103 m=video 10102 UDP/TLS/RTP/SAVPF 100 101 102 103
c=IN IP4 203.0.113.100 c=IN IP4 203.0.113.100
a=mid:v1 a=mid:v1
a=sendrecv a=sendrecv
a=rtpmap:100 VP8/90000 a=rtpmap:100 VP8/90000
a=rtpmap:101 H264/90000 a=rtpmap:101 H264/90000
a=fmtp:101 packetization-mode=1;profile-level-id=42e01f a=fmtp:101 packetization-mode=1;profile-level-id=42e01f
a=rtpmap:102 rtx/90000 a=rtpmap:102 rtx/90000
a=fmtp:102 apt=100 a=fmtp:102 apt=100
=rtpmap:103 rtx/90000 a=rtpmap:103 rtx/90000
a=fmtp:103 apt=101 a=fmtp:103 apt=101
a=extmap:1 urn:ietf:params:rtp-hdrext:sdes:mid a=extmap:1 urn:ietf:params:rtp-hdrext:sdes:mid
a=extmap:3 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id a=extmap:3 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id
a=rtcp-fb:100 ccm fir a=rtcp-fb:100 ccm fir
a=rtcp-fb:100 nack a=rtcp-fb:100 nack
a=rtcp-fb:100 nack pli a=rtcp-fb:100 nack pli
a=msid:47017fee-b6c1-4162-929c-a25110252400 a=msid:47017fee-b6c1-4162-929c-a25110252400
a=ice-ufrag:BGKk a=ice-ufrag:BGKk
a=ice-pwd:mqyWsAjvtKwTGnvhPztQ9mIf a=ice-pwd:mqyWsAjvtKwTGnvhPztQ9mIf
a=fingerprint:sha-256 a=fingerprint:sha-256
skipping to change at page 78, line 22 skipping to change at line 3682
m=video 10200 UDP/TLS/RTP/SAVPF 100 101 102 103 m=video 10200 UDP/TLS/RTP/SAVPF 100 101 102 103
c=IN IP4 203.0.113.200 c=IN IP4 203.0.113.200
a=mid:v1 a=mid:v1
a=sendrecv a=sendrecv
a=rtpmap:100 VP8/90000 a=rtpmap:100 VP8/90000
a=rtpmap:101 H264/90000 a=rtpmap:101 H264/90000
a=fmtp:101 packetization-mode=1;profile-level-id=42e01f a=fmtp:101 packetization-mode=1;profile-level-id=42e01f
a=rtpmap:102 rtx/90000 a=rtpmap:102 rtx/90000
a=fmtp:102 apt=100 a=fmtp:102 apt=100
=rtpmap:103 rtx/90000 a=rtpmap:103 rtx/90000
a=fmtp:103 apt=101 a=fmtp:103 apt=101
a=extmap:1 urn:ietf:params:rtp-hdrext:sdes:mid a=extmap:1 urn:ietf:params:rtp-hdrext:sdes:mid
a=extmap:3 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id a=extmap:3 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id
a=rtcp-fb:100 ccm fir a=rtcp-fb:100 ccm fir
a=rtcp-fb:100 nack a=rtcp-fb:100 nack
a=rtcp-fb:100 nack pli a=rtcp-fb:100 nack pli
a=msid:61317484-2ed4-49d7-9eb7-1414322a7aae a=msid:61317484-2ed4-49d7-9eb7-1414322a7aae
7.2. Detailed Example 7.2. Detailed Example
This section shows a more involved example of a session between two This section shows a more involved example of a session between two
JSEP endpoints. Trickle ICE is used in full trickle mode, with a JSEP endpoints. Trickle ICE is used in full trickle mode, with a
bundle policy of "max-bundle", an RTCP mux policy of "require", and a bundle policy of "max-bundle", an RTCP mux policy of "require", and a
single TURN server. Initially, both Alice and Bob establish an audio single TURN server. Initially, both Alice and Bob establish an audio
channel and a data channel. Later, Bob adds two video flows, one for channel and a data channel. Later, Bob adds two video flows -- one
his video feed, and one for screensharing, both supporting FEC, and for his video feed and one for screen sharing, both supporting FEC --
with the video feed configured for simulcast. Alice accepts these with the video feed configured for simulcast. Alice accepts these
video flows, but does not add video flows of her own, so they are video flows but does not add video flows of her own, so they are
handled as recvonly. Alice also specifies a maximum video decoder handled as recvonly. Alice also specifies a maximum video decoder
resolution. resolution.
// set up local media state // set up local media state
AliceJS->AliceUA: create new PeerConnection AliceJS->AliceUA: create new PeerConnection
AliceJS->AliceUA: addTrack with an audio track AliceJS->AliceUA: addTrack with an audio track
AliceJS->AliceUA: createDataChannel to get data channel AliceJS->AliceUA: createDataChannel to get data channel
AliceJS->AliceUA: createOffer to get |offer-B1| AliceJS->AliceUA: createOffer to get |offer-B1|
AliceJS->AliceUA: setLocalDescription with |offer-B1| AliceJS->AliceUA: setLocalDescription with |offer-B1|
// |offer-B1| is sent over signaling protocol to Bob
AliceJS->WebServer: signaling with |offer-B1|
WebServer->BobJS: signaling with |offer-B1|
// |offer-B1| arrives at Bob // |offer-B1| is sent over signaling protocol to Bob
BobJS->BobUA: create a PeerConnection AliceJS->WebServer: signaling with |offer-B1|
BobJS->BobUA: setRemoteDescription with |offer-B1| WebServer->BobJS: signaling with |offer-B1|
BobUA->BobJS: ontrack with audio track from Alice
// candidates are sent to Bob // |offer-B1| arrives at Bob
AliceUA->AliceJS: onicecandidate (host) |offer-B1-candidate-1| BobJS->BobUA: create a PeerConnection
AliceJS->WebServer: signaling with |offer-B1-candidate-1| BobJS->BobUA: setRemoteDescription with |offer-B1|
AliceUA->AliceJS: onicecandidate (srflx) |offer-B1-candidate-2| BobUA->BobJS: ontrack event with audio track from Alice
AliceJS->WebServer: signaling with |offer-B1-candidate-2|
AliceUA->AliceJS: onicecandidate (relay) |offer-B1-candidate-3|
AliceJS->WebServer: signaling with |offer-B1-candidate-3|
WebServer->BobJS: signaling with |offer-B1-candidate-1| // candidates are sent to Bob
BobJS->BobUA: addIceCandidate with |offer-B1-candidate-1| AliceUA->AliceJS: onicecandidate (host) |offer-B1-candidate-1|
WebServer->BobJS: signaling with |offer-B1-candidate-2| AliceJS->WebServer: signaling with |offer-B1-candidate-1|
BobJS->BobUA: addIceCandidate with |offer-B1-candidate-2| AliceUA->AliceJS: onicecandidate (srflx) |offer-B1-candidate-2|
WebServer->BobJS: signaling with |offer-B1-candidate-3| AliceJS->WebServer: signaling with |offer-B1-candidate-2|
BobJS->BobUA: addIceCandidate with |offer-B1-candidate-3| AliceUA->AliceJS: onicecandidate (relay) |offer-B1-candidate-3|
AliceJS->WebServer: signaling with |offer-B1-candidate-3|
// Bob accepts call WebServer->BobJS: signaling with |offer-B1-candidate-1|
BobJS->BobUA: addTrack with local audio BobJS->BobUA: addIceCandidate with |offer-B1-candidate-1|
BobJS->BobUA: createDataChannel to get data channel WebServer->BobJS: signaling with |offer-B1-candidate-2|
BobJS->BobUA: createAnswer to get |answer-B1| BobJS->BobUA: addIceCandidate with |offer-B1-candidate-2|
BobJS->BobUA: setLocalDescription with |answer-B1| WebServer->BobJS: signaling with |offer-B1-candidate-3|
BobJS->BobUA: addIceCandidate with |offer-B1-candidate-3|
// |answer-B1| is sent to Alice // Bob accepts call
BobJS->WebServer: signaling with |answer-B1| BobJS->BobUA: addTrack with local audio
WebServer->AliceJS: signaling with |answer-B1| BobJS->BobUA: createDataChannel to get data channel
AliceJS->AliceUA: setRemoteDescription with |answer-B1| BobJS->BobUA: createAnswer to get |answer-B1|
AliceUA->AliceJS: ontrack event with audio track from Bob BobJS->BobUA: setLocalDescription with |answer-B1|
// candidates are sent to Alice // |answer-B1| is sent to Alice
BobUA->BobJS: onicecandidate (host) |answer-B1-candidate-1| BobJS->WebServer: signaling with |answer-B1|
BobJS->WebServer: signaling with |answer-B1-candidate-1| WebServer->AliceJS: signaling with |answer-B1|
BobUA->BobJS: onicecandidate (srflx) |answer-B1-candidate-2| AliceJS->AliceUA: setRemoteDescription with |answer-B1|
BobJS->WebServer: signaling with |answer-B1-candidate-2| AliceUA->AliceJS: ontrack event with audio track from Bob
BobUA->BobJS: onicecandidate (relay) |answer-B1-candidate-3|
BobJS->WebServer: signaling with |answer-B1-candidate-3|
WebServer->AliceJS: signaling with |answer-B1-candidate-1| // candidates are sent to Alice
AliceJS->AliceUA: addIceCandidate with |answer-B1-candidate-1| BobUA->BobJS: onicecandidate (host) |answer-B1-candidate-1|
WebServer->AliceJS: signaling with |answer-B1-candidate-2| BobJS->WebServer: signaling with |answer-B1-candidate-1|
AliceJS->AliceUA: addIceCandidate with |answer-B1-candidate-2| BobUA->BobJS: onicecandidate (srflx) |answer-B1-candidate-2|
WebServer->AliceJS: signaling with |answer-B1-candidate-3| BobJS->WebServer: signaling with |answer-B1-candidate-2|
AliceJS->AliceUA: addIceCandidate with |answer-B1-candidate-3| BobUA->BobJS: onicecandidate (relay) |answer-B1-candidate-3|
BobJS->WebServer: signaling with |answer-B1-candidate-3|
// data channel opens WebServer->AliceJS: signaling with |answer-B1-candidate-1|
BobUA->BobJS: ondatachannel event AliceJS->AliceUA: addIceCandidate with |answer-B1-candidate-1|
AliceUA->AliceJS: ondatachannel event WebServer->AliceJS: signaling with |answer-B1-candidate-2|
BobUA->BobJS: onopen AliceJS->AliceUA: addIceCandidate with |answer-B1-candidate-2|
AliceUA->AliceJS: onopen WebServer->AliceJS: signaling with |answer-B1-candidate-3|
AliceJS->AliceUA: addIceCandidate with |answer-B1-candidate-3|
// media is flowing between endpoints // data channel opens
BobUA->AliceUA: audio+data sent from Bob to Alice BobUA->BobJS: ondatachannel event
AliceUA->BobUA: audio+data sent from Alice to Bob AliceUA->AliceJS: ondatachannel event
BobUA->BobJS: onopen
AliceUA->AliceJS: onopen
// some time later Bob adds two video streams // media is flowing between endpoints
// note, no candidates exchanged, because of bundle BobUA->AliceUA: audio+data sent from Bob to Alice
BobJS->BobUA: addTrack with first video stream AliceUA->BobUA: audio+data sent from Alice to Bob
BobJS->BobUA: addTrack with second video stream
BobJS->BobUA: createOffer to get |offer-B2|
BobJS->BobUA: setLocalDescription with |offer-B2|
// |offer-B2| is sent to Alice // some time later, Bob adds two video streams
BobJS->WebServer: signaling with |offer-B2| // note: no candidates exchanged, because of bundle
WebServer->AliceJS: signaling with |offer-B2| BobJS->BobUA: addTrack with first video stream
AliceJS->AliceUA: setRemoteDescription with |offer-B2| BobJS->BobUA: addTrack with second video stream
AliceUA->AliceJS: ontrack event with first video track BobJS->BobUA: createOffer to get |offer-B2|
AliceUA->AliceJS: ontrack event with second video track BobJS->BobUA: setLocalDescription with |offer-B2|
AliceJS->AliceUA: createAnswer to get |answer-B2|
AliceJS->AliceUA: setLocalDescription with |answer-B2|
// |answer-B2| is sent over signaling protocol to Bob // |offer-B2| is sent to Alice
AliceJS->WebServer: signaling with |answer-B2| BobJS->WebServer: signaling with |offer-B2|
WebServer->BobJS: signaling with |answer-B2| WebServer->AliceJS: signaling with |offer-B2|
BobJS->BobUA: setRemoteDescription with |answer-B2| AliceJS->AliceUA: setRemoteDescription with |offer-B2|
AliceUA->AliceJS: ontrack event with first video track
AliceUA->AliceJS: ontrack event with second video track
AliceJS->AliceUA: createAnswer to get |answer-B2|
AliceJS->AliceUA: setLocalDescription with |answer-B2|
// media is flowing between endpoints // |answer-B2| is sent over signaling protocol
BobUA->AliceUA: audio+video+data sent from Bob to Alice // to Bob
AliceUA->BobUA: audio+video+data sent from Alice to Bob AliceJS->WebServer: signaling with |answer-B2|
WebServer->BobJS: signaling with |answer-B2|
BobJS->BobUA: setRemoteDescription with |answer-B2|
// media is flowing between endpoints
BobUA->AliceUA: audio+video+data sent from Bob to Alice
AliceUA->BobUA: audio+video+data sent from Alice to Bob
The SDP for |offer-B1| looks like: The SDP for |offer-B1| looks like:
v=0 v=0
o=- 4962303333179871723 1 IN IP4 0.0.0.0 o=- 4962303333179871723 1 IN IP4 0.0.0.0
s=- s=-
t=0 0 t=0 0
a=ice-options:trickle ice2 a=ice-options:trickle ice2
a=group:BUNDLE a1 d1 a=group:BUNDLE a1 d1
skipping to change at page 84, line 20 skipping to change at line 3927
raddr 203.0.113.200 rport 10200 raddr 203.0.113.200 rport 10200
|answer-B1-candidate-3| looks like: |answer-B1-candidate-3| looks like:
ufrag 7sFv ufrag 7sFv
index 0 index 0
mid a1 mid a1
attr candidate:1 1 udp 255 192.0.2.200 12200 typ relay attr candidate:1 1 udp 255 192.0.2.200 12200 typ relay
raddr 198.51.100.200 rport 11200 raddr 198.51.100.200 rport 11200
The SDP for |offer-B2| is shown below. In addition to the new m= The SDP for |offer-B2| is shown below. In addition to the new "m="
sections for video, both of which are offering FEC, and one of which sections for video, both of which are offering FEC and one of which
is offering simulcast, note the increment of the version number in is offering simulcast, note the increment of the version number in
the o= line, changes to the c= line, indicating the local candidate the "o=" line; changes to the "c=" line, indicating the local
that was selected, and the inclusion of gathered candidates as candidate that was selected; and the inclusion of gathered candidates
a=candidate lines. as a=candidate lines.
v=0 v=0
o=- 7729291447651054566 2 IN IP4 0.0.0.0 o=- 7729291447651054566 2 IN IP4 0.0.0.0
s=- s=-
t=0 0 t=0 0
a=ice-options:trickle ice2 a=ice-options:trickle ice2
a=group:BUNDLE a1 d1 v1 v2 a=group:BUNDLE a1 d1 v1 v2
a=group:LS a1 v1 a=group:LS a1 v1
m=audio 12200 UDP/TLS/RTP/SAVPF 96 0 8 97 98 m=audio 12200 UDP/TLS/RTP/SAVPF 96 0 8 97 98
skipping to change at page 85, line 39 skipping to change at line 3989
m=video 12200 UDP/TLS/RTP/SAVPF 100 101 102 103 104 m=video 12200 UDP/TLS/RTP/SAVPF 100 101 102 103 104
c=IN IP4 192.0.2.200 c=IN IP4 192.0.2.200
a=mid:v1 a=mid:v1
a=sendrecv a=sendrecv
a=rtpmap:100 VP8/90000 a=rtpmap:100 VP8/90000
a=rtpmap:101 H264/90000 a=rtpmap:101 H264/90000
a=fmtp:101 packetization-mode=1;profile-level-id=42e01f a=fmtp:101 packetization-mode=1;profile-level-id=42e01f
a=rtpmap:102 rtx/90000 a=rtpmap:102 rtx/90000
a=fmtp:102 apt=100 a=fmtp:102 apt=100
=rtpmap:103 rtx/90000 a=rtpmap:103 rtx/90000
a=fmtp:103 apt=101 a=fmtp:103 apt=101
a=rtpmap:104 flexfec/90000 a=rtpmap:104 flexfec/90000
a=extmap:1 urn:ietf:params:rtp-hdrext:sdes:mid a=extmap:1 urn:ietf:params:rtp-hdrext:sdes:mid
a=extmap:3 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id a=extmap:3 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id
a=rtcp-fb:100 ccm fir a=rtcp-fb:100 ccm fir
a=rtcp-fb:100 nack a=rtcp-fb:100 nack
a=rtcp-fb:100 nack pli a=rtcp-fb:100 nack pli
a=msid:71317484-2ed4-49d7-9eb7-1414322a7aae a=msid:71317484-2ed4-49d7-9eb7-1414322a7aae
a=rid:1 send a=rid:1 send
a=rid:2 send a=rid:2 send
skipping to change at page 86, line 4 skipping to change at line 4002
a=extmap:1 urn:ietf:params:rtp-hdrext:sdes:mid a=extmap:1 urn:ietf:params:rtp-hdrext:sdes:mid
a=extmap:3 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id a=extmap:3 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id
a=rtcp-fb:100 ccm fir a=rtcp-fb:100 ccm fir
a=rtcp-fb:100 nack a=rtcp-fb:100 nack
a=rtcp-fb:100 nack pli a=rtcp-fb:100 nack pli
a=msid:71317484-2ed4-49d7-9eb7-1414322a7aae a=msid:71317484-2ed4-49d7-9eb7-1414322a7aae
a=rid:1 send a=rid:1 send
a=rid:2 send a=rid:2 send
a=rid:3 send a=rid:3 send
a=simulcast:send 1;2;3 a=simulcast:send 1;2;3
m=video 12200 UDP/TLS/RTP/SAVPF 100 101 102 103 104 m=video 12200 UDP/TLS/RTP/SAVPF 100 101 102 103 104
c=IN IP4 192.0.2.200 c=IN IP4 192.0.2.200
a=mid:v2 a=mid:v2
a=sendrecv a=sendrecv
a=rtpmap:100 VP8/90000 a=rtpmap:100 VP8/90000
a=rtpmap:101 H264/90000 a=rtpmap:101 H264/90000
a=fmtp:101 packetization-mode=1;profile-level-id=42e01f a=fmtp:101 packetization-mode=1;profile-level-id=42e01f
a=rtpmap:102 rtx/90000 a=rtpmap:102 rtx/90000
a=fmtp:102 apt=100 a=fmtp:102 apt=100
=rtpmap:103 rtx/90000 a=rtpmap:103 rtx/90000
a=fmtp:103 apt=101 a=fmtp:103 apt=101
a=rtpmap:104 flexfec/90000 a=rtpmap:104 flexfec/90000
a=extmap:1 urn:ietf:params:rtp-hdrext:sdes:mid a=extmap:1 urn:ietf:params:rtp-hdrext:sdes:mid
a=extmap:3 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id a=extmap:3 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id
a=rtcp-fb:100 ccm fir a=rtcp-fb:100 ccm fir
a=rtcp-fb:100 nack a=rtcp-fb:100 nack
a=rtcp-fb:100 nack pli a=rtcp-fb:100 nack pli
a=msid:81317484-2ed4-49d7-9eb7-1414322a7aae a=msid:81317484-2ed4-49d7-9eb7-1414322a7aae
The SDP for |answer-B2| is shown below. In addition to the The SDP for |answer-B2| is shown below. In addition to the
acceptance of the video m= sections, the use of a=recvonly to acceptance of the video "m=" sections, the use of a=recvonly to
indicate one-way video, and the use of a=imageattr to limit the indicate one-way video, and the use of a=imageattr to limit the
received resolution, note the use of setup:passive to maintain the received resolution, note the use of setup:passive to maintain the
existing DTLS roles. existing DTLS roles.
v=0 v=0
o=- 4962303333179871723 2 IN IP4 0.0.0.0 o=- 4962303333179871723 2 IN IP4 0.0.0.0
s=- s=-
t=0 0 t=0 0
a=ice-options:trickle ice2 a=ice-options:trickle ice2
a=group:BUNDLE a1 d1 v1 v2 a=group:BUNDLE a1 d1 v1 v2
skipping to change at page 87, line 38 skipping to change at line 4083
m=video 12100 UDP/TLS/RTP/SAVPF 100 101 102 103 m=video 12100 UDP/TLS/RTP/SAVPF 100 101 102 103
c=IN IP4 192.0.2.100 c=IN IP4 192.0.2.100
a=mid:v1 a=mid:v1
a=recvonly a=recvonly
a=rtpmap:100 VP8/90000 a=rtpmap:100 VP8/90000
a=rtpmap:101 H264/90000 a=rtpmap:101 H264/90000
a=fmtp:101 packetization-mode=1;profile-level-id=42e01f a=fmtp:101 packetization-mode=1;profile-level-id=42e01f
a=rtpmap:102 rtx/90000 a=rtpmap:102 rtx/90000
a=fmtp:102 apt=100 a=fmtp:102 apt=100
=rtpmap:103 rtx/90000 a=rtpmap:103 rtx/90000
a=fmtp:103 apt=101 a=fmtp:103 apt=101
a=imageattr:100 recv [x=[48:1920],y=[48:1080],q=1.0] a=imageattr:100 recv [x=[48:1920],y=[48:1080],q=1.0]
a=extmap:1 urn:ietf:params:rtp-hdrext:sdes:mid a=extmap:1 urn:ietf:params:rtp-hdrext:sdes:mid
a=extmap:3 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id a=extmap:3 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id
a=rtcp-fb:100 ccm fir a=rtcp-fb:100 ccm fir
a=rtcp-fb:100 nack a=rtcp-fb:100 nack
a=rtcp-fb:100 nack pli a=rtcp-fb:100 nack pli
m=video 12100 UDP/TLS/RTP/SAVPF 100 101 102 103 m=video 12100 UDP/TLS/RTP/SAVPF 100 101 102 103
c=IN IP4 192.0.2.100 c=IN IP4 192.0.2.100
a=mid:v2 a=mid:v2
a=recvonly a=recvonly
a=rtpmap:100 VP8/90000 a=rtpmap:100 VP8/90000
a=rtpmap:101 H264/90000 a=rtpmap:101 H264/90000
a=fmtp:101 packetization-mode=1;profile-level-id=42e01f a=fmtp:101 packetization-mode=1;profile-level-id=42e01f
a=rtpmap:102 rtx/90000 a=rtpmap:102 rtx/90000
a=fmtp:102 apt=100 a=fmtp:102 apt=100
=rtpmap:103 rtx/90000 a=rtpmap:103 rtx/90000
a=fmtp:103 apt=101 a=fmtp:103 apt=101
a=imageattr:100 recv [x=[48:1920],y=[48:1080],q=1.0] a=imageattr:100 recv [x=[48:1920],y=[48:1080],q=1.0]
a=extmap:1 urn:ietf:params:rtp-hdrext:sdes:mid a=extmap:1 urn:ietf:params:rtp-hdrext:sdes:mid
a=extmap:3 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id a=extmap:3 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id
a=rtcp-fb:100 ccm fir a=rtcp-fb:100 ccm fir
a=rtcp-fb:100 nack a=rtcp-fb:100 nack
a=rtcp-fb:100 nack pli a=rtcp-fb:100 nack pli
7.3. Early Transport Warmup Example 7.3. Early Transport Warmup Example
This example demonstrates the early warmup technique described in This example demonstrates the early-warmup technique described in
Section 4.1.8.1. Here, Alice's endpoint sends an offer to Bob's Section 4.1.10.1. Here, Alice's endpoint sends an offer to Bob's
endpoint to start an audio/video call. Bob immediately responds with endpoint to start an audio/video call. Bob immediately responds with
an answer that accepts the audio/video m= sections, but marks them as an answer that accepts the audio/video "m=" sections but marks them
sendonly (from his perspective), meaning that Alice will not yet send as sendonly (from his perspective), meaning that Alice will not yet
media. This allows the JSEP implementation to start negotiating ICE send media. This allows the JSEP implementation to start negotiating
and DTLS immediately. Bob's endpoint then prompts him to answer the ICE and DTLS immediately. Bob's endpoint then prompts him to answer
call, and when he does, his endpoint sends a second offer which the call, and when he does, his endpoint sends a second offer, which
enables the audio and video m= sections, and thereby bidirectional enables the audio and video "m=" sections, and thereby bidirectional
media transmission. The advantage of such a flow is that as soon as media transmission. The advantage of such a flow is that as soon as
the first answer is received, the implementation can proceed with ICE the first answer is received, the implementation can proceed with ICE
and DTLS negotiation and establish the session transport. If the and DTLS negotiation and establish the session transport. If the
transport setup completes before the second offer is sent, then media transport setup completes before the second offer is sent, then media
can be transmitted immediately by the callee immediately upon can be transmitted by the callee immediately upon answering the call,
answering the call, minimizing perceived post-dial-delay. The second minimizing perceived post-dial delay. The second offer/answer
offer/answer exchange can also change the preferred codecs or other exchange can also change the preferred codecs or other session
session parameters. parameters.
This example also makes use of the "relay" ICE candidate policy This example also makes use of the "relay" ICE candidate policy
described in Section 3.5.3 to minimize the ICE gathering and checking described in Section 3.5.3 to minimize the ICE gathering and checking
needed. needed.
// set up local media state // set up local media state
AliceJS->AliceUA: create new PeerConnection with "relay" ICE policy AliceJS->AliceUA: create new PeerConnection with "relay" ICE policy
AliceJS->AliceUA: addTrack with two tracks: audio and video AliceJS->AliceUA: addTrack with two tracks: audio and video
AliceJS->AliceUA: createOffer to get |offer-C1| AliceJS->AliceUA: createOffer to get |offer-C1|
AliceJS->AliceUA: setLocalDescription with |offer-C1| AliceJS->AliceUA: setLocalDescription with |offer-C1|
// |offer-C1| is sent over signaling protocol to Bob // |offer-C1| is sent over signaling protocol to Bob
AliceJS->WebServer: signaling with |offer-C1| AliceJS->WebServer: signaling with |offer-C1|
WebServer->BobJS: signaling with |offer-C1| WebServer->BobJS: signaling with |offer-C1|
// |offer-C1| arrives at Bob
BobJS->BobUA: create new PeerConnection with "relay" ICE policy
BobJS->BobUA: setRemoteDescription with |offer-C1|
BobUA->BobJS: ontrack events for audio and video
// a relay candidate is sent to Bob // |offer-C1| arrives at Bob
AliceUA->AliceJS: onicecandidate (relay) |offer-C1-candidate-1| BobJS->BobUA: create new PeerConnection with "relay" ICE policy
AliceJS->WebServer: signaling with |offer-C1-candidate-1| BobJS->BobUA: setRemoteDescription with |offer-C1|
BobUA->BobJS: ontrack events for audio and video
WebServer->BobJS: signaling with |offer-C1-candidate-1| // a relay candidate is sent to Bob
BobJS->BobUA: addIceCandidate with |offer-C1-candidate-1| AliceUA->AliceJS: onicecandidate (relay) |offer-C1-candidate-1|
AliceJS->WebServer: signaling with |offer-C1-candidate-1|
// Bob prepares an early answer to warmup the transport WebServer->BobJS: signaling with |offer-C1-candidate-1|
BobJS->BobUA: addTransceiver with null audio and video tracks BobJS->BobUA: addIceCandidate with |offer-C1-candidate-1|
BobJS->BobUA: transceiver.setDirection(sendonly) for both
BobJS->BobUA: createAnswer
BobJS->BobUA: setLocalDescription with answer
// |answer-C1| is sent over signaling protocol to Alice // Bob prepares an early answer to warm up the
BobJS->WebServer: signaling with |answer-C1| // transport
WebServer->AliceJS: signaling with |answer-C1| BobJS->BobUA: addTransceiver with null audio and video tracks
BobJS->BobUA: transceiver.setDirection(sendonly) for both
BobJS->BobUA: createAnswer
BobJS->BobUA: setLocalDescription with answer
// |answer-C1| (sendonly) arrives at Alice // |answer-C1| is sent over signaling protocol
AliceJS->AliceUA: setRemoteDescription with |answer-C1| // to Alice
AliceUA->AliceJS: ontrack events for audio and video BobJS->WebServer: signaling with |answer-C1|
WebServer->AliceJS: signaling with |answer-C1|
// a relay candidate is sent to Alice // |answer-C1| (sendonly) arrives at Alice
BobUA->BobJS: onicecandidate (relay) |answer-B1-candidate-1| AliceJS->AliceUA: setRemoteDescription with |answer-C1|
BobJS->WebServer: signaling with |answer-B1-candidate-1| AliceUA->AliceJS: ontrack events for audio and video
WebServer->AliceJS: signaling with |answer-B1-candidate-1| // a relay candidate is sent to Alice
AliceJS->AliceUA: addIceCandidate with |answer-B1-candidate-1| BobUA->BobJS: onicecandidate (relay) |answer-B1-candidate-1|
BobJS->WebServer: signaling with |answer-B1-candidate-1|
// ICE and DTLS establish while call is ringing WebServer->AliceJS: signaling with |answer-B1-candidate-1|
AliceJS->AliceUA: addIceCandidate with |answer-B1-candidate-1|
// Bob accepts call, starts media, and sends new offer // ICE and DTLS establish while call is ringing
BobJS->BobUA: transceiver.setTrack with audio and video tracks
BobUA->AliceUA: media sent from Bob to Alice
BobJS->BobUA: transceiver.setDirection(sendrecv) for both
transceivers
BobJS->BobUA: createOffer
BobJS->BobUA: setLocalDescription with offer
// |offer-C2| is sent over signaling protocol to Alice // Bob accepts call, starts media, and sends
BobJS->WebServer: signaling with |offer-C2| // new offer
WebServer->AliceJS: signaling with |offer-C2| BobJS->BobUA: transceiver.setTrack with audio and video tracks
BobUA->AliceUA: media sent from Bob to Alice
BobJS->BobUA: transceiver.setDirection(sendrecv) for both
transceivers
BobJS->BobUA: createOffer
BobJS->BobUA: setLocalDescription with offer
// |offer-C2| (sendrecv) arrives at Alice // |offer-C2| is sent over signaling protocol
AliceJS->AliceUA: setRemoteDescription with |offer-C2| // to Alice
AliceJS->AliceUA: createAnswer BobJS->WebServer: signaling with |offer-C2|
AliceJS->AliceUA: setLocalDescription with |answer-C2| WebServer->AliceJS: signaling with |offer-C2|
AliceUA->BobUA: media sent from Alice to Bob
// |answer-C2| is sent over signaling protocol to Bob // |offer-C2| (sendrecv) arrives at Alice
AliceJS->WebServer: signaling with |answer-C2| AliceJS->AliceUA: setRemoteDescription with |offer-C2|
WebServer->BobJS: signaling with |answer-C2| AliceJS->AliceUA: createAnswer
BobJS->BobUA: setRemoteDescription with |answer-C2| AliceJS->AliceUA: setLocalDescription with |answer-C2|
AliceUA->BobUA: media sent from Alice to Bob
// |answer-C2| is sent over signaling protocol
// to Bob
AliceJS->WebServer: signaling with |answer-C2|
WebServer->BobJS: signaling with |answer-C2|
BobJS->BobUA: setRemoteDescription with |answer-C2|
The SDP for |offer-C1| looks like: The SDP for |offer-C1| looks like:
v=0 v=0
o=- 1070771854436052752 1 IN IP4 0.0.0.0 o=- 1070771854436052752 1 IN IP4 0.0.0.0
s=- s=-
t=0 0 t=0 0
a=ice-options:trickle ice2 a=ice-options:trickle ice2
a=group:BUNDLE a1 v1 a=group:BUNDLE a1 v1
a=group:LS a1 v1 a=group:LS a1 v1
skipping to change at page 91, line 4 skipping to change at line 4242
a=ice-ufrag:4ZcD a=ice-ufrag:4ZcD
a=ice-pwd:ZaaG6OG7tCn4J/lehAGz+HHD a=ice-pwd:ZaaG6OG7tCn4J/lehAGz+HHD
a=fingerprint:sha-256 a=fingerprint:sha-256
C4:68:F8:77:6A:44:F1:98:6D:7C:9F:47:EB:E3:34:A4: C4:68:F8:77:6A:44:F1:98:6D:7C:9F:47:EB:E3:34:A4:
0A:AA:2D:49:08:28:70:2E:1F:AE:18:7D:4E:3E:66:BF 0A:AA:2D:49:08:28:70:2E:1F:AE:18:7D:4E:3E:66:BF
a=setup:actpass a=setup:actpass
a=tls-id:9e5b948ade9c3d41de6617b68f769e55 a=tls-id:9e5b948ade9c3d41de6617b68f769e55
a=rtcp-mux a=rtcp-mux
a=rtcp-mux-only a=rtcp-mux-only
a=rtcp-rsize a=rtcp-rsize
m=video 0 UDP/TLS/RTP/SAVPF 100 101 102 103 m=video 0 UDP/TLS/RTP/SAVPF 100 101 102 103
c=IN IP4 0.0.0.0 c=IN IP4 0.0.0.0
a=mid:v1 a=mid:v1
a=sendrecv a=sendrecv
a=rtpmap:100 VP8/90000 a=rtpmap:100 VP8/90000
a=rtpmap:101 H264/90000 a=rtpmap:101 H264/90000
a=fmtp:101 packetization-mode=1;profile-level-id=42e01f a=fmtp:101 packetization-mode=1;profile-level-id=42e01f
a=rtpmap:102 rtx/90000 a=rtpmap:102 rtx/90000
a=fmtp:102 apt=100 a=fmtp:102 apt=100
=rtpmap:103 rtx/90000 a=rtpmap:103 rtx/90000
a=fmtp:103 apt=101 a=fmtp:103 apt=101
a=extmap:1 urn:ietf:params:rtp-hdrext:sdes:mid a=extmap:1 urn:ietf:params:rtp-hdrext:sdes:mid
a=extmap:3 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id a=extmap:3 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id
a=rtcp-fb:100 ccm fir a=rtcp-fb:100 ccm fir
a=rtcp-fb:100 nack a=rtcp-fb:100 nack
a=rtcp-fb:100 nack pli a=rtcp-fb:100 nack pli
a=msid:bbce3ba6-abfc-ac63-d00a-e15b286f8fce a=msid:bbce3ba6-abfc-ac63-d00a-e15b286f8fce
a=bundle-only a=bundle-only
|offer-C1-candidate-1| looks like: |offer-C1-candidate-1| looks like:
skipping to change at page 92, line 32 skipping to change at line 4315
m=video 9 UDP/TLS/RTP/SAVPF 100 101 102 103 m=video 9 UDP/TLS/RTP/SAVPF 100 101 102 103
c=IN IP4 0.0.0.0 c=IN IP4 0.0.0.0
a=mid:v1 a=mid:v1
a=sendonly a=sendonly
a=rtpmap:100 VP8/90000 a=rtpmap:100 VP8/90000
a=rtpmap:101 H264/90000 a=rtpmap:101 H264/90000
a=fmtp:101 packetization-mode=1;profile-level-id=42e01f a=fmtp:101 packetization-mode=1;profile-level-id=42e01f
a=rtpmap:102 rtx/90000 a=rtpmap:102 rtx/90000
a=fmtp:102 apt=100 a=fmtp:102 apt=100
=rtpmap:103 rtx/90000 a=rtpmap:103 rtx/90000
a=fmtp:103 apt=101 a=fmtp:103 apt=101
a=extmap:1 urn:ietf:params:rtp-hdrext:sdes:mid a=extmap:1 urn:ietf:params:rtp-hdrext:sdes:mid
a=extmap:3 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id a=extmap:3 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id
a=rtcp-fb:100 ccm fir a=rtcp-fb:100 ccm fir
a=rtcp-fb:100 nack a=rtcp-fb:100 nack
a=rtcp-fb:100 nack pli a=rtcp-fb:100 nack pli
a=msid:751f239e-4ae0-c549-aa3d-890de772998b a=msid:751f239e-4ae0-c549-aa3d-890de772998b
|answer-C1-candidate-1| looks like: |answer-C1-candidate-1| looks like:
skipping to change at page 94, line 5 skipping to change at line 4380
m=video 12200 UDP/TLS/RTP/SAVPF 100 101 102 103 m=video 12200 UDP/TLS/RTP/SAVPF 100 101 102 103
c=IN IP4 192.0.2.200 c=IN IP4 192.0.2.200
a=mid:v1 a=mid:v1
a=sendrecv a=sendrecv
a=rtpmap:100 VP8/90000 a=rtpmap:100 VP8/90000
a=rtpmap:101 H264/90000 a=rtpmap:101 H264/90000
a=fmtp:101 packetization-mode=1;profile-level-id=42e01f a=fmtp:101 packetization-mode=1;profile-level-id=42e01f
a=rtpmap:102 rtx/90000 a=rtpmap:102 rtx/90000
a=fmtp:102 apt=100 a=fmtp:102 apt=100
=rtpmap:103 rtx/90000 a=rtpmap:103 rtx/90000
a=fmtp:103 apt=101 a=fmtp:103 apt=101
a=extmap:1 urn:ietf:params:rtp-hdrext:sdes:mid a=extmap:1 urn:ietf:params:rtp-hdrext:sdes:mid
a=extmap:3 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id a=extmap:3 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id
a=rtcp-fb:100 ccm fir a=rtcp-fb:100 ccm fir
a=rtcp-fb:100 nack a=rtcp-fb:100 nack
a=rtcp-fb:100 nack pli a=rtcp-fb:100 nack pli
a=msid:751f239e-4ae0-c549-aa3d-890de772998b a=msid:751f239e-4ae0-c549-aa3d-890de772998b
The SDP for |answer-C2| looks like: The SDP for |answer-C2| looks like:
skipping to change at page 95, line 16 skipping to change at line 4437
m=video 12100 UDP/TLS/RTP/SAVPF 100 101 102 103 m=video 12100 UDP/TLS/RTP/SAVPF 100 101 102 103
c=IN IP4 192.0.2.100 c=IN IP4 192.0.2.100
a=mid:v1 a=mid:v1
a=sendrecv a=sendrecv
a=rtpmap:100 VP8/90000 a=rtpmap:100 VP8/90000
a=rtpmap:101 H264/90000 a=rtpmap:101 H264/90000
a=fmtp:101 packetization-mode=1;profile-level-id=42e01f a=fmtp:101 packetization-mode=1;profile-level-id=42e01f
a=rtpmap:102 rtx/90000 a=rtpmap:102 rtx/90000
a=fmtp:102 apt=100 a=fmtp:102 apt=100
=rtpmap:103 rtx/90000 a=rtpmap:103 rtx/90000
a=fmtp:103 apt=101 a=fmtp:103 apt=101
a=extmap:1 urn:ietf:params:rtp-hdrext:sdes:mid a=extmap:1 urn:ietf:params:rtp-hdrext:sdes:mid
a=extmap:3 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id a=extmap:3 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id
a=rtcp-fb:100 ccm fir a=rtcp-fb:100 ccm fir
a=rtcp-fb:100 nack a=rtcp-fb:100 nack
a=rtcp-fb:100 nack pli a=rtcp-fb:100 nack pli
a=msid:bbce3ba6-abfc-ac63-d00a-e15b286f8fce a=msid:bbce3ba6-abfc-ac63-d00a-e15b286f8fce
8. Security Considerations 8. Security Considerations
The IETF has published separate documents The IETF has published separate documents [RFC8827] [RFC8826]
[I-D.ietf-rtcweb-security-arch] [I-D.ietf-rtcweb-security] describing describing the security architecture for WebRTC as a whole. The
the security architecture for WebRTC as a whole. The remainder of remainder of this section describes security considerations for this
this section describes security considerations for this document. document.
While formally the JSEP interface is an API, it is better to think of While formally the JSEP interface is an API, it is better to think of
it as an Internet protocol, with the application JavaScript being it as an Internet protocol, with the application JavaScript being
untrustworthy from the perspective of the JSEP implementation. Thus, untrustworthy from the perspective of the JSEP implementation. Thus,
the threat model of [RFC3552] applies. In particular, JavaScript can the threat model of [RFC3552] applies. In particular, JavaScript can
call the API in any order and with any inputs, including malicious call the API in any order and with any inputs, including malicious
ones. This is particularly relevant when we consider the SDP which ones. This is particularly relevant when we consider the SDP that is
is passed to setLocalDescription(). While correct API usage requires passed to setLocalDescription. While correct API usage requires that
that the application pass in SDP which was derived from createOffer() the application pass in SDP that was derived from createOffer or
or createAnswer(), there is no guarantee that applications do so. createAnswer, there is no guarantee that applications do so. The
The JSEP implementation MUST be prepared for the JavaScript to pass JSEP implementation MUST be prepared for the JavaScript to pass in
in bogus data instead. bogus data instead.
Conversely, the application programmer needs to be aware that the Conversely, the application programmer needs to be aware that the
JavaScript does not have complete control of endpoint behavior. One JavaScript does not have complete control of endpoint behavior. One
case that bears particular mention is that editing ICE candidates out case that bears particular mention is that editing ICE candidates out
of the SDP or suppressing trickled candidates does not have the of the SDP or suppressing trickled candidates does not have the
expected behavior: implementations will still perform checks from expected behavior: implementations will still perform checks from
those candidates even if they are not sent to the other side. Thus, those candidates even if they are not sent to the other side. Thus,
for instance, it is not possible to prevent the remote peer from for instance, it is not possible to prevent the remote peer from
learning your public IP address by removing server reflexive learning your public IP address by removing server-reflexive
candidates. Applications which wish to conceal their public IP candidates. Applications that wish to conceal their public IP
address should instead configure the ICE agent to use only relay address MUST instead configure the ICE agent to use only relay
candidates. candidates.
9. IANA Considerations 9. IANA Considerations
This document requires no actions from IANA. This document has no IANA actions.
10. Acknowledgements
Harald Alvestrand, Taylor Brandstetter, Suhas Nandakumar, and Peter
Thatcher provided significant text for this draft. Bernard Aboba,
Adam Bergkvist, Dan Burnett, Ben Campbell, Alissa Cooper, Richard
Ejzak, Stefan Hakansson, Ted Hardie, Christer Holmberg Andrew Hutton,
Randell Jesup, Matthew Kaufman, Anant Narayanan, Adam Roach, Robert
Sparks, Neil Stratford, Martin Thomson, Sean Turner, and Magnus
Westerlund all provided valuable feedback on this proposal.
11. References
11.1. Normative References
[I-D.ietf-avtext-rid]
Roach, A., Nandakumar, S., and P. Thatcher, "RTP Stream
Identifier Source Description (SDES)", draft-ietf-avtext-
rid-09 (work in progress), October 2016.
[I-D.ietf-ice-trickle]
Ivov, E., Rescorla, E., Uberti, J., and P. Saint-Andre,
"Trickle ICE: Incremental Provisioning of Candidates for
the Interactive Connectivity Establishment (ICE)
Protocol", draft-ietf-ice-trickle-21 (work in progress),
April 2018.
[I-D.ietf-mmusic-dtls-sdp]
Holmberg, C. and R. Shpount, "Session Description Protocol
(SDP) Offer/Answer Considerations for Datagram Transport
Layer Security (DTLS) and Transport Layer Security (TLS)",
draft-ietf-mmusic-dtls-sdp-32 (work in progress), October
2017.
[I-D.ietf-mmusic-ice-sip-sdp]
Petit-Huguenin, M., Nandakumar, S., and A. Keranen,
"Session Description Protocol (SDP) Offer/Answer
procedures for Interactive Connectivity Establishment
(ICE)", draft-ietf-mmusic-ice-sip-sdp-24 (work in
progress), November 2018.
[I-D.ietf-mmusic-msid]
Alvestrand, H., "WebRTC MediaStream Identification in the
Session Description Protocol", draft-ietf-mmusic-msid-17
(work in progress), December 2018.
[I-D.ietf-mmusic-mux-exclusive]
Holmberg, C., "Indicating Exclusive Support of RTP/RTCP
Multiplexing using SDP", draft-ietf-mmusic-mux-
exclusive-12 (work in progress), May 2017.
[I-D.ietf-mmusic-rid]
Roach, A., "RTP Payload Format Restrictions", draft-ietf-
mmusic-rid-15 (work in progress), May 2018.
[I-D.ietf-mmusic-sctp-sdp]
Holmberg, C., Shpount, R., Loreto, S., and G. Camarillo,
"Session Description Protocol (SDP) Offer/Answer
Procedures For Stream Control Transmission Protocol (SCTP)
over Datagram Transport Layer Security (DTLS) Transport.",
draft-ietf-mmusic-sctp-sdp-26 (work in progress), April
2017.
[I-D.ietf-mmusic-sdp-bundle-negotiation]
Holmberg, C., Alvestrand, H., and C. Jennings,
"Negotiating Media Multiplexing Using the Session
Description Protocol (SDP)", draft-ietf-mmusic-sdp-bundle-
negotiation-54 (work in progress), December 2018.
[I-D.ietf-mmusic-sdp-mux-attributes]
Nandakumar, S., "A Framework for SDP Attributes when
Multiplexing", draft-ietf-mmusic-sdp-mux-attributes-17
(work in progress), February 2018.
[I-D.ietf-mmusic-sdp-simulcast]
Burman, B., Westerlund, M., Nandakumar, S., and M. Zanaty,
"Using Simulcast in SDP and RTP Sessions", draft-ietf-
mmusic-sdp-simulcast-13 (work in progress), June 2018.
[I-D.ietf-rtcweb-fec]
Uberti, J., "WebRTC Forward Error Correction
Requirements", draft-ietf-rtcweb-fec-08 (work in
progress), March 2018.
[I-D.ietf-rtcweb-rtp-usage]
Perkins, C., Westerlund, M., and J. Ott, "Web Real-Time
Communication (WebRTC): Media Transport and Use of RTP",
draft-ietf-rtcweb-rtp-usage-26 (work in progress), March
2016.
[I-D.ietf-rtcweb-security] 10. References
Rescorla, E., "Security Considerations for WebRTC", draft-
ietf-rtcweb-security-11 (work in progress), February 2019.
[I-D.ietf-rtcweb-security-arch] 10.1. Normative References
Rescorla, E., "WebRTC Security Architecture", draft-ietf-
rtcweb-security-arch-18 (work in progress), February 2019.
[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate [RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
Requirement Levels", BCP 14, RFC 2119, Requirement Levels", BCP 14, RFC 2119,
DOI 10.17487/RFC2119, March 1997, DOI 10.17487/RFC2119, March 1997,
<https://www.rfc-editor.org/info/rfc2119>. <https://www.rfc-editor.org/info/rfc2119>.
[RFC3261] Rosenberg, J., Schulzrinne, H., Camarillo, G., Johnston, [RFC3261] Rosenberg, J., Schulzrinne, H., Camarillo, G., Johnston,
A., Peterson, J., Sparks, R., Handley, M., and E. A., Peterson, J., Sparks, R., Handley, M., and E.
Schooler, "SIP: Session Initiation Protocol", RFC 3261, Schooler, "SIP: Session Initiation Protocol", RFC 3261,
DOI 10.17487/RFC3261, June 2002, DOI 10.17487/RFC3261, June 2002,
skipping to change at page 100, line 15 skipping to change at line 4583
[RFC7160] Petit-Huguenin, M. and G. Zorn, Ed., "Support for Multiple [RFC7160] Petit-Huguenin, M. and G. Zorn, Ed., "Support for Multiple
Clock Rates in an RTP Session", RFC 7160, Clock Rates in an RTP Session", RFC 7160,
DOI 10.17487/RFC7160, April 2014, DOI 10.17487/RFC7160, April 2014,
<https://www.rfc-editor.org/info/rfc7160>. <https://www.rfc-editor.org/info/rfc7160>.
[RFC7587] Spittka, J., Vos, K., and JM. Valin, "RTP Payload Format [RFC7587] Spittka, J., Vos, K., and JM. Valin, "RTP Payload Format
for the Opus Speech and Audio Codec", RFC 7587, for the Opus Speech and Audio Codec", RFC 7587,
DOI 10.17487/RFC7587, June 2015, DOI 10.17487/RFC7587, June 2015,
<https://www.rfc-editor.org/info/rfc7587>. <https://www.rfc-editor.org/info/rfc7587>.
[RFC7742] Roach, A., "WebRTC Video Processing and Codec [RFC7742] Roach, A.B., "WebRTC Video Processing and Codec
Requirements", RFC 7742, DOI 10.17487/RFC7742, March 2016, Requirements", RFC 7742, DOI 10.17487/RFC7742, March 2016,
<https://www.rfc-editor.org/info/rfc7742>. <https://www.rfc-editor.org/info/rfc7742>.
[RFC7850] Nandakumar, S., "Registering Values of the SDP 'proto' [RFC7850] Nandakumar, S., "Registering Values of the SDP 'proto'
Field for Transporting RTP Media over TCP under Various Field for Transporting RTP Media over TCP under Various
RTP Profiles", RFC 7850, DOI 10.17487/RFC7850, April 2016, RTP Profiles", RFC 7850, DOI 10.17487/RFC7850, April 2016,
<https://www.rfc-editor.org/info/rfc7850>. <https://www.rfc-editor.org/info/rfc7850>.
[RFC7874] Valin, JM. and C. Bran, "WebRTC Audio Codec and Processing [RFC7874] Valin, JM. and C. Bran, "WebRTC Audio Codec and Processing
Requirements", RFC 7874, DOI 10.17487/RFC7874, May 2016, Requirements", RFC 7874, DOI 10.17487/RFC7874, May 2016,
skipping to change at page 100, line 39 skipping to change at line 4607
"Sending Multiple RTP Streams in a Single RTP Session", "Sending Multiple RTP Streams in a Single RTP Session",
RFC 8108, DOI 10.17487/RFC8108, March 2017, RFC 8108, DOI 10.17487/RFC8108, March 2017,
<https://www.rfc-editor.org/info/rfc8108>. <https://www.rfc-editor.org/info/rfc8108>.
[RFC8122] Lennox, J. and C. Holmberg, "Connection-Oriented Media [RFC8122] Lennox, J. and C. Holmberg, "Connection-Oriented Media
Transport over the Transport Layer Security (TLS) Protocol Transport over the Transport Layer Security (TLS) Protocol
in the Session Description Protocol (SDP)", RFC 8122, in the Session Description Protocol (SDP)", RFC 8122,
DOI 10.17487/RFC8122, March 2017, DOI 10.17487/RFC8122, March 2017,
<https://www.rfc-editor.org/info/rfc8122>. <https://www.rfc-editor.org/info/rfc8122>.
[RFC8174] Leiba, B., "Ambiguity of Uppercase vs Lowercase in RFC
2119 Key Words", BCP 14, RFC 8174, DOI 10.17487/RFC8174,
May 2017, <https://www.rfc-editor.org/info/rfc8174>.
[RFC8445] Keranen, A., Holmberg, C., and J. Rosenberg, "Interactive [RFC8445] Keranen, A., Holmberg, C., and J. Rosenberg, "Interactive
Connectivity Establishment (ICE): A Protocol for Network Connectivity Establishment (ICE): A Protocol for Network
Address Translator (NAT) Traversal", RFC 8445, Address Translator (NAT) Traversal", RFC 8445,
DOI 10.17487/RFC8445, July 2018, DOI 10.17487/RFC8445, July 2018,
<https://www.rfc-editor.org/info/rfc8445>. <https://www.rfc-editor.org/info/rfc8445>.
11.2. Informative References [RFC8826] Rescorla, E., "Security Considerations for WebRTC",
RFC 8826, DOI 10.17487/RFC8826, January 2021,
<https://www.rfc-editor.org/info/rfc8826>.
[I-D.ietf-mmusic-trickle-ice-sip] [RFC8827] Rescorla, E., "WebRTC Security Architecture", RFC 8827,
Ivov, E., Stach, T., Marocco, E., and C. Holmberg, "A DOI 10.17487/RFC8827, January 2021,
<https://www.rfc-editor.org/info/rfc8827>.
[RFC8830] Alvestrand, H., "WebRTC MediaStream Identification in the
Session Description Protocol", RFC 8830,
DOI 10.17487/RFC8830, January 2021,
<https://www.rfc-editor.org/info/rfc8830>.
[RFC8834] Perkins, C., Westerlund, M., and J. Ott, "Media Transport
and Use of RTP in WebRTC", RFC 8834, DOI 10.17487/RFC8834,
January 2021, <https://www.rfc-editor.org/info/rfc8834>.
[RFC8838] Ivov, E., Uberti, J., and P. Saint-Andre, "Trickle ICE:
Incremental Provisioning of Candidates for the Interactive
Connectivity Establishment (ICE) Protocol", RFC 8838,
DOI 10.17487/RFC8838, January 2021,
<https://www.rfc-editor.org/info/rfc8838>.
[RFC8839] Petit-Huguenin, M., Nandakumar, S., Holmberg, C., Keränen,
A., and R. Shpount, "Session Description Protocol (SDP)
Offer/Answer Procedures for Interactive Connectivity
Establishment (ICE)", RFC 8839, DOI 10.17487/RFC8839,
January 2021, <https://www.rfc-editor.org/info/rfc8839>.
[RFC8840] Ivov, E., Stach, T., Marocco, E., and C. Holmberg, "A
Session Initiation Protocol (SIP) Usage for Incremental Session Initiation Protocol (SIP) Usage for Incremental
Provisioning of Candidates for the Interactive Provisioning of Candidates for the Interactive
Connectivity Establishment (Trickle ICE)", draft-ietf- Connectivity Establishment (Trickle ICE)", RFC 8840,
mmusic-trickle-ice-sip-18 (work in progress), June 2018. DOI 10.17487/RFC8840, January 2021,
<https://www.rfc-editor.org/info/rfc8840>.
[I-D.ietf-rtcweb-ip-handling] [RFC8841] Holmberg, C., Shpount, R., Loreto, S., and G. Camarillo,
Uberti, J., "WebRTC IP Address Handling Requirements", "Session Description Protocol (SDP) Offer/Answer
draft-ietf-rtcweb-ip-handling-11 (work in progress), Procedures for Stream Control Transmission Protocol (SCTP)
November 2018. over Datagram Transport Layer Security (DTLS) Transport",
RFC 8841, DOI 10.17487/RFC8841, January 2021,
<https://www.rfc-editor.org/info/rfc8841>.
[I-D.ietf-rtcweb-sdp] [RFC8842] Holmberg, C. and R. Shpount, "Session Description Protocol
Nandakumar, S. and C. Jennings, "Annotated Example SDP for (SDP) Offer/Answer Considerations for Datagram Transport
WebRTC", draft-ietf-rtcweb-sdp-11 (work in progress), Layer Security (DTLS) and Transport Layer Security (TLS)",
October 2018. RFC 8842, DOI 10.17487/RFC8842, January 2021,
<https://www.rfc-editor.org/info/rfc8842>.
[RFC8843] Holmberg, C., Alvestrand, H., and C. Jennings,
"Negotiating Media Multiplexing Using the Session
Description Protocol (SDP)", RFC 8843,
DOI 10.17487/RFC8843, January 2021,
<https://www.rfc-editor.org/info/rfc8843>.
[RFC8851] Roach, A.B., Ed., "RTP Payload Format Restrictions",
RFC 8851, DOI 10.17487/RFC8851, January 2021,
<https://www.rfc-editor.org/info/rfc8851>.
[RFC8852] Roach, A.B., Nandakumar, S., and P. Thatcher, "RTP Stream
Identifier Source Description (SDES)", RFC 8852,
DOI 10.17487/RFC8852, January 2021,
<https://www.rfc-editor.org/info/rfc8852>.
[RFC8853] Burman, B., Westerlund, M., Nandakumar, S., and M. Zanaty,
"Using Simulcast in Session Description Protocol (SDP) and
RTP Sessions", RFC 8853, DOI 10.17487/RFC8853, January
2021, <https://www.rfc-editor.org/info/rfc8853>.
[RFC8854] Uberti, J., "WebRTC Forward Error Correction
Requirements", RFC 8854, DOI 10.17487/RFC8854, January
2021, <https://www.rfc-editor.org/info/rfc8854>.
[RFC8858] Holmberg, C., "Indicating Exclusive Support of RTP and RTP
Control Protocol (RTCP) Multiplexing Using the Session
Description Protocol (SDP)", RFC 8858,
DOI 10.17487/RFC8858, January 2021,
<https://www.rfc-editor.org/info/rfc8858>.
[RFC8859] Nandakumar, S., "A Framework for Session Description
Protocol (SDP) Attributes When Multiplexing", RFC 8859,
DOI 10.17487/RFC8859, January 2021,
<https://www.rfc-editor.org/info/rfc8859>.
10.2. Informative References
[RFC3389] Zopf, R., "Real-time Transport Protocol (RTP) Payload for [RFC3389] Zopf, R., "Real-time Transport Protocol (RTP) Payload for
Comfort Noise (CN)", RFC 3389, DOI 10.17487/RFC3389, Comfort Noise (CN)", RFC 3389, DOI 10.17487/RFC3389,
September 2002, <https://www.rfc-editor.org/info/rfc3389>. September 2002, <https://www.rfc-editor.org/info/rfc3389>.
[RFC3556] Casner, S., "Session Description Protocol (SDP) Bandwidth [RFC3556] Casner, S., "Session Description Protocol (SDP) Bandwidth
Modifiers for RTP Control Protocol (RTCP) Bandwidth", Modifiers for RTP Control Protocol (RTCP) Bandwidth",
RFC 3556, DOI 10.17487/RFC3556, July 2003, RFC 3556, DOI 10.17487/RFC3556, July 2003,
<https://www.rfc-editor.org/info/rfc3556>. <https://www.rfc-editor.org/info/rfc3556>.
skipping to change at page 102, line 27 skipping to change at line 4760
(SRTP) Security Context Using Datagram Transport Layer (SRTP) Security Context Using Datagram Transport Layer
Security (DTLS)", RFC 5763, DOI 10.17487/RFC5763, May Security (DTLS)", RFC 5763, DOI 10.17487/RFC5763, May
2010, <https://www.rfc-editor.org/info/rfc5763>. 2010, <https://www.rfc-editor.org/info/rfc5763>.
[RFC5764] McGrew, D. and E. Rescorla, "Datagram Transport Layer [RFC5764] McGrew, D. and E. Rescorla, "Datagram Transport Layer
Security (DTLS) Extension to Establish Keys for the Secure Security (DTLS) Extension to Establish Keys for the Secure
Real-time Transport Protocol (SRTP)", RFC 5764, Real-time Transport Protocol (SRTP)", RFC 5764,
DOI 10.17487/RFC5764, May 2010, DOI 10.17487/RFC5764, May 2010,
<https://www.rfc-editor.org/info/rfc5764>. <https://www.rfc-editor.org/info/rfc5764>.
[RFC6120] Saint-Andre, P., "Extensible Messaging and Presence
Protocol (XMPP): Core", RFC 6120, DOI 10.17487/RFC6120,
March 2011, <https://www.rfc-editor.org/info/rfc6120>.
[RFC6464] Lennox, J., Ed., Ivov, E., and E. Marocco, "A Real-time [RFC6464] Lennox, J., Ed., Ivov, E., and E. Marocco, "A Real-time
Transport Protocol (RTP) Header Extension for Client-to- Transport Protocol (RTP) Header Extension for Client-to-
Mixer Audio Level Indication", RFC 6464, Mixer Audio Level Indication", RFC 6464,
DOI 10.17487/RFC6464, December 2011, DOI 10.17487/RFC6464, December 2011,
<https://www.rfc-editor.org/info/rfc6464>. <https://www.rfc-editor.org/info/rfc6464>.
[RFC6544] Rosenberg, J., Keranen, A., Lowekamp, B., and A. Roach, [RFC8828] Uberti, J. and G. Shieh, "WebRTC IP Address Handling
"TCP Candidates with Interactive Connectivity Requirements", RFC 8828, DOI 10.17487/RFC8828, January
Establishment (ICE)", RFC 6544, DOI 10.17487/RFC6544, 2021, <https://www.rfc-editor.org/info/rfc8828>.
March 2012, <https://www.rfc-editor.org/info/rfc6544>.
[TS26.114] [SDP4WebRTC]
3GPP TS 26.114 V12.8.0, "3rd Generation Partnership Nandakumar, S. and C. Jennings, "Annotated Example SDP for