draft-ietf-rtcweb-jsep-20.txt   draft-ietf-rtcweb-jsep-21.txt 
Network Working Group J. Uberti Network Working Group J. Uberti
Internet-Draft Google Internet-Draft Google
Intended status: Standards Track C. Jennings Intended status: Standards Track C. Jennings
Expires: September 30, 2017 Cisco Expires: January 4, 2018 Cisco
E. Rescorla, Ed. E. Rescorla, Ed.
Mozilla Mozilla
March 29, 2017 July 3, 2017
Javascript Session Establishment Protocol JavaScript Session Establishment Protocol
draft-ietf-rtcweb-jsep-20 draft-ietf-rtcweb-jsep-21
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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Copyright Notice Copyright Notice
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Table of Contents Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 4 1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 4
1.1. General Design of JSEP . . . . . . . . . . . . . . . . . 4 1.1. General Design of JSEP . . . . . . . . . . . . . . . . . 4
1.2. Other Approaches Considered . . . . . . . . . . . . . . . 5 1.2. Other Approaches Considered . . . . . . . . . . . . . . . 6
2. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 6 2. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 6
3. Semantics and Syntax . . . . . . . . . . . . . . . . . . . . 6 3. Semantics and Syntax . . . . . . . . . . . . . . . . . . . . 7
3.1. Signaling Model . . . . . . . . . . . . . . . . . . . . . 6 3.1. Signaling Model . . . . . . . . . . . . . . . . . . . . . 7
3.2. Session Descriptions and State Machine . . . . . . . . . 7 3.2. Session Descriptions and State Machine . . . . . . . . . 7
3.3. Session Description Format . . . . . . . . . . . . . . . 10 3.3. Session Description Format . . . . . . . . . . . . . . . 11
3.4. Session Description Control . . . . . . . . . . . . . . . 10 3.4. Session Description Control . . . . . . . . . . . . . . . 11
3.4.1. RtpTransceivers . . . . . . . . . . . . . . . . . . . 10 3.4.1. RtpTransceivers . . . . . . . . . . . . . . . . . . . 11
3.4.2. RtpSenders . . . . . . . . . . . . . . . . . . . . . 11 3.4.2. RtpSenders . . . . . . . . . . . . . . . . . . . . . 12
3.4.3. RtpReceivers . . . . . . . . . . . . . . . . . . . . 11 3.4.3. RtpReceivers . . . . . . . . . . . . . . . . . . . . 12
3.5. ICE . . . . . . . . . . . . . . . . . . . . . . . . . . . 11 3.5. ICE . . . . . . . . . . . . . . . . . . . . . . . . . . . 12
3.5.1. ICE Gathering Overview . . . . . . . . . . . . . . . 11 3.5.1. ICE Gathering Overview . . . . . . . . . . . . . . . 12
3.5.2. ICE Candidate Trickling . . . . . . . . . . . . . . . 12 3.5.2. ICE Candidate Trickling . . . . . . . . . . . . . . . 13
3.5.2.1. ICE Candidate Format . . . . . . . . . . . . . . 13 3.5.2.1. ICE Candidate Format . . . . . . . . . . . . . . 13
3.5.3. ICE Candidate Policy . . . . . . . . . . . . . . . . 13 3.5.3. ICE Candidate Policy . . . . . . . . . . . . . . . . 14
3.5.4. ICE Candidate Pool . . . . . . . . . . . . . . . . . 14 3.5.4. ICE Candidate Pool . . . . . . . . . . . . . . . . . 15
3.6. Video Size Negotiation . . . . . . . . . . . . . . . . . 15 3.6. Video Size Negotiation . . . . . . . . . . . . . . . . . 16
3.6.1. Creating an imageattr Attribute . . . . . . . . . . . 15 3.6.1. Creating an imageattr Attribute . . . . . . . . . . . 16
3.6.2. Interpreting an imageattr Attribute . . . . . . . . . 16 3.6.2. Interpreting an imageattr Attribute . . . . . . . . . 17
3.7. Simulcast . . . . . . . . . . . . . . . . . . . . . . . . 17 3.7. Simulcast . . . . . . . . . . . . . . . . . . . . . . . . 18
3.8. Interactions With Forking . . . . . . . . . . . . . . . . 18 3.8. Interactions With Forking . . . . . . . . . . . . . . . . 19
3.8.1. Sequential Forking . . . . . . . . . . . . . . . . . 19 3.8.1. Sequential Forking . . . . . . . . . . . . . . . . . 20
3.8.2. Parallel Forking . . . . . . . . . . . . . . . . . . 19 3.8.2. Parallel Forking . . . . . . . . . . . . . . . . . . 20
4. Interface . . . . . . . . . . . . . . . . . . . . . . . . . . 20 4. Interface . . . . . . . . . . . . . . . . . . . . . . . . . . 21
4.1. PeerConnection . . . . . . . . . . . . . . . . . . . . . 20 4.1. PeerConnection . . . . . . . . . . . . . . . . . . . . . 21
4.1.1. Constructor . . . . . . . . . . . . . . . . . . . . . 20 4.1.1. Constructor . . . . . . . . . . . . . . . . . . . . . 21
4.1.2. addTrack . . . . . . . . . . . . . . . . . . . . . . 22 4.1.2. addTrack . . . . . . . . . . . . . . . . . . . . . . 24
4.1.3. removeTrack . . . . . . . . . . . . . . . . . . . . . 23 4.1.3. removeTrack . . . . . . . . . . . . . . . . . . . . . 24
4.1.4. addTransceiver . . . . . . . . . . . . . . . . . . . 23 4.1.4. addTransceiver . . . . . . . . . . . . . . . . . . . 24
4.1.5. createDataChannel . . . . . . . . . . . . . . . . . . 23 4.1.5. createDataChannel . . . . . . . . . . . . . . . . . . 24
4.1.6. createOffer . . . . . . . . . . . . . . . . . . . . . 23 4.1.6. createOffer . . . . . . . . . . . . . . . . . . . . . 25
4.1.7. createAnswer . . . . . . . . . . . . . . . . . . . . 24 4.1.7. createAnswer . . . . . . . . . . . . . . . . . . . . 26
4.1.8. SessionDescriptionType . . . . . . . . . . . . . . . 25 4.1.8. SessionDescriptionType . . . . . . . . . . . . . . . 26
4.1.8.1. Use of Provisional Answers . . . . . . . . . . . 26 4.1.8.1. Use of Provisional Answers . . . . . . . . . . . 27
4.1.8.2. Rollback . . . . . . . . . . . . . . . . . . . . 27 4.1.8.2. Rollback . . . . . . . . . . . . . . . . . . . . 28
4.1.9. setLocalDescription . . . . . . . . . . . . . . . . . 28 4.1.9. setLocalDescription . . . . . . . . . . . . . . . . . 29
4.1.10. setRemoteDescription . . . . . . . . . . . . . . . . 28 4.1.10. setRemoteDescription . . . . . . . . . . . . . . . . 30
4.1.11. currentLocalDescription . . . . . . . . . . . . . . . 29 4.1.11. currentLocalDescription . . . . . . . . . . . . . . . 30
4.1.12. pendingLocalDescription . . . . . . . . . . . . . . . 29 4.1.12. pendingLocalDescription . . . . . . . . . . . . . . . 30
4.1.13. currentRemoteDescription . . . . . . . . . . . . . . 29 4.1.13. currentRemoteDescription . . . . . . . . . . . . . . 30
4.1.14. pendingRemoteDescription . . . . . . . . . . . . . . 29 4.1.14. pendingRemoteDescription . . . . . . . . . . . . . . 31
4.1.15. canTrickleIceCandidates . . . . . . . . . . . . . . . 30 4.1.15. canTrickleIceCandidates . . . . . . . . . . . . . . . 31
4.1.16. setConfiguration . . . . . . . . . . . . . . . . . . 30 4.1.16. setConfiguration . . . . . . . . . . . . . . . . . . 31
4.1.17. addIceCandidate . . . . . . . . . . . . . . . . . . . 31 4.1.17. addIceCandidate . . . . . . . . . . . . . . . . . . . 32
4.2. RtpTransceiver . . . . . . . . . . . . . . . . . . . . . 32 4.2. RtpTransceiver . . . . . . . . . . . . . . . . . . . . . 33
4.2.1. stop . . . . . . . . . . . . . . . . . . . . . . . . 32 4.2.1. stop . . . . . . . . . . . . . . . . . . . . . . . . 33
4.2.2. stopped . . . . . . . . . . . . . . . . . . . . . . . 32 4.2.2. stopped . . . . . . . . . . . . . . . . . . . . . . . 33
4.2.3. setDirection . . . . . . . . . . . . . . . . . . . . 32 4.2.3. setDirection . . . . . . . . . . . . . . . . . . . . 33
4.2.4. direction . . . . . . . . . . . . . . . . . . . . . . 32 4.2.4. direction . . . . . . . . . . . . . . . . . . . . . . 34
4.2.5. currentDirection . . . . . . . . . . . . . . . . . . 33 4.2.5. currentDirection . . . . . . . . . . . . . . . . . . 34
4.2.6. setCodecPreferences . . . . . . . . . . . . . . . . . 33 4.2.6. setCodecPreferences . . . . . . . . . . . . . . . . . 34
5. SDP Interaction Procedures . . . . . . . . . . . . . . . . . 33 5. SDP Interaction Procedures . . . . . . . . . . . . . . . . . 35
5.1. Requirements Overview . . . . . . . . . . . . . . . . . . 34 5.1. Requirements Overview . . . . . . . . . . . . . . . . . . 35
5.1.1. Usage Requirements . . . . . . . . . . . . . . . . . 34 5.1.1. Usage Requirements . . . . . . . . . . . . . . . . . 35
5.1.2. Profile Names and Interoperability . . . . . . . . . 34 5.1.2. Profile Names and Interoperability . . . . . . . . . 35
5.2. Constructing an Offer . . . . . . . . . . . . . . . . . . 35 5.2. Constructing an Offer . . . . . . . . . . . . . . . . . . 36
5.2.1. Initial Offers . . . . . . . . . . . . . . . . . . . 35 5.2.1. Initial Offers . . . . . . . . . . . . . . . . . . . 37
5.2.2. Subsequent Offers . . . . . . . . . . . . . . . . . . 42 5.2.2. Subsequent Offers . . . . . . . . . . . . . . . . . . 43
5.2.3. Options Handling . . . . . . . . . . . . . . . . . . 46 5.2.3. Options Handling . . . . . . . . . . . . . . . . . . 47
5.2.3.1. IceRestart . . . . . . . . . . . . . . . . . . . 46 5.2.3.1. IceRestart . . . . . . . . . . . . . . . . . . . 47
5.2.3.2. VoiceActivityDetection . . . . . . . . . . . . . 46 5.2.3.2. VoiceActivityDetection . . . . . . . . . . . . . 47
5.3. Generating an Answer . . . . . . . . . . . . . . . . . . 47 5.3. Generating an Answer . . . . . . . . . . . . . . . . . . 48
5.3.1. Initial Answers . . . . . . . . . . . . . . . . . . . 47 5.3.1. Initial Answers . . . . . . . . . . . . . . . . . . . 48
5.3.2. Subsequent Answers . . . . . . . . . . . . . . . . . 53 5.3.2. Subsequent Answers . . . . . . . . . . . . . . . . . 55
5.3.3. Options Handling . . . . . . . . . . . . . . . . . . 54 5.3.3. Options Handling . . . . . . . . . . . . . . . . . . 56
5.3.3.1. VoiceActivityDetection . . . . . . . . . . . . . 55 5.3.3.1. VoiceActivityDetection . . . . . . . . . . . . . 56
5.4. Modifying an Offer or Answer . . . . . . . . . . . . . . 55 5.4. Modifying an Offer or Answer . . . . . . . . . . . . . . 56
5.5. Processing a Local Description . . . . . . . . . . . . . 56 5.5. Processing a Local Description . . . . . . . . . . . . . 57
5.6. Processing a Remote Description . . . . . . . . . . . . . 56 5.6. Processing a Remote Description . . . . . . . . . . . . . 58
5.7. Parsing a Session Description . . . . . . . . . . . . . . 57 5.7. Parsing a Session Description . . . . . . . . . . . . . . 58
5.7.1. Session-Level Parsing . . . . . . . . . . . . . . . . 57 5.7.1. Session-Level Parsing . . . . . . . . . . . . . . . . 59
5.7.2. Media Section Parsing . . . . . . . . . . . . . . . . 59 5.7.2. Media Section Parsing . . . . . . . . . . . . . . . . 60
5.7.3. Semantics Verification . . . . . . . . . . . . . . . 61 5.7.3. Semantics Verification . . . . . . . . . . . . . . . 63
5.8. Applying a Local Description . . . . . . . . . . . . . . 63 5.8. Applying a Local Description . . . . . . . . . . . . . . 64
5.9. Applying a Remote Description . . . . . . . . . . . . . . 64 5.9. Applying a Remote Description . . . . . . . . . . . . . . 66
5.10. Applying an Answer . . . . . . . . . . . . . . . . . . . 68 5.10. Applying an Answer . . . . . . . . . . . . . . . . . . . 69
6. Processing RTP/RTCP . . . . . . . . . . . . . . . . . . . . . 70 6. Processing RTP/RTCP . . . . . . . . . . . . . . . . . . . . . 72
7. Examples . . . . . . . . . . . . . . . . . . . . . . . . . . 70 7. Examples . . . . . . . . . . . . . . . . . . . . . . . . . . 72
7.1. Simple Example . . . . . . . . . . . . . . . . . . . . . 71 7.1. Simple Example . . . . . . . . . . . . . . . . . . . . . 73
7.2. Detailed Example . . . . . . . . . . . . . . . . . . . . 76 7.2. Detailed Example . . . . . . . . . . . . . . . . . . . . 77
7.3. Early Transport Warmup Example . . . . . . . . . . . . . 85 7.3. Early Transport Warmup Example . . . . . . . . . . . . . 87
8. Security Considerations . . . . . . . . . . . . . . . . . . . 93 8. Security Considerations . . . . . . . . . . . . . . . . . . . 94
9. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 94 9. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 95
10. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 94 10. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 95
11. References . . . . . . . . . . . . . . . . . . . . . . . . . 94 11. References . . . . . . . . . . . . . . . . . . . . . . . . . 95
11.1. Normative References . . . . . . . . . . . . . . . . . . 94 11.1. Normative References . . . . . . . . . . . . . . . . . . 95
11.2. Informative References . . . . . . . . . . . . . . . . . 98 11.2. Informative References . . . . . . . . . . . . . . . . . 100
Appendix A. Appendix A . . . . . . . . . . . . . . . . . . . . . 100 Appendix A. Appendix A . . . . . . . . . . . . . . . . . . . . . 102
Appendix B. Change log . . . . . . . . . . . . . . . . . . . . . 101 Appendix B. Change log . . . . . . . . . . . . . . . . . . . . . 103
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 110 Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 112
1. Introduction 1. Introduction
This document describes how the W3C WEBRTC RTCPeerConnection This document describes how the W3C WEBRTC RTCPeerConnection
interface [W3C.WD-webrtc-20140617] is used to control the setup, interface [W3C.webrtc] is used to control the setup, management and
management and teardown of a multimedia session. teardown of a multimedia session.
1.1. General Design of JSEP 1.1. General Design of JSEP
The thinking behind WebRTC call setup has been to fully specify and The thinking behind WebRTC call setup has been to fully specify and
control the media plane, but to leave the signaling plane up to the control the media plane, but to leave the signaling plane up to the
application as much as possible. The rationale is that different application as much as possible. The rationale is that different
applications may prefer to use different protocols, such as the applications may prefer to use different protocols, such as the
existing SIP or Jingle call signaling protocols, or something custom existing SIP call signaling protocol, or something custom to the
to the particular application, perhaps for a novel use case. In this particular application, perhaps for a novel use case. In this
approach, the key information that needs to be exchanged is the approach, the key information that needs to be exchanged is the
multimedia session description, which specifies the necessary multimedia session description, which specifies the necessary
transport and media configuration information necessary to establish transport and media configuration information necessary to establish
the media plane. 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) that allows for full
control of the signaling state machine from Javascript. As described control of the signaling state machine from JavaScript. As described
above, JSEP assumes a model in which a Javascript application above, JSEP assumes a model in which a JavaScript application
executes inside a runtime containing WebRTC APIs (the "JSEP executes inside a runtime containing WebRTC APIs (the "JSEP
implementation"). The JSEP implementation is almost entirely 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 ICE state
machine. The combination of the JSEP implementation and the machine. The combination of the JSEP implementation and the
Javascript application is referred to throughout this document as a JavaScript application is referred to throughout this document as a
"JSEP endpoint". "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 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 MCU. This distinction is invisible to the JSEP
endpoint; it just follows the instructions it is given via the API. 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
skipping to change at page 5, line 16 skipping to change at page 5, line 16
createAnswer() API to generate an appropriate answer, applies it createAnswer() API to generate an appropriate answer, applies it
using the setLocalDescription() API, and sends the answer back to the using 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 [RFC5245], JSEP decouples the ICE state machine from Regarding ICE [RFC5245], JSEP decouples the ICE state machine from
the overall signaling state machine, as the ICE state machine must the overall signaling state machine, as the ICE state machine must
remain in the JSEP implementation, because only the implementation remain in the JSEP implementation, because only the implementation
has the necessary knowledge of candidates and other transport info. has the necessary knowledge of candidates and other transport
Performing this separation also provides additional flexibility; in information. Performing this separation provides additional
protocols that decouple session descriptions from transport, such as flexibility in protocols that decouple session descriptions from
Jingle, the session description can be sent immediately and the transport. For instance, in traditional SIP, each offer or answer is
transport information can be sent when available. In protocols that self-contained, including both the session descriptions and the
don't, such as SIP, the information can be used in the aggregated transport information. However, [I-D.ietf-mmusic-trickle-ice-sip]
form. Sending transport information separately can allow for faster allows SIP to be used with trickle ICE [I-D.ietf-ice-trickle], in
ICE and DTLS startup, since ICE checks can start as soon as any which the session description can be sent immediately and the
transport information is available rather than waiting for all of it. transport information can be sent when available. Sending transport
information separately can allow for faster ICE and DTLS startup,
since ICE checks can start as soon as any transport information is
available rather than waiting for all of it. JSEP's decoupling of
the ICE and signaling state machines allows it to accommodate either
model.
Through its abstraction of signaling, the JSEP approach does require Through its abstraction of signaling, the JSEP approach does require
the application to be aware of the signaling process. While the the application to be aware of the signaling process. While the
application does not need to understand the contents of session application does not need to understand the contents of session
descriptions to set up a call, the application must call the right descriptions to set up a call, the application must call the right
APIs at the right times, convert the session descriptions and ICE APIs at the right times, convert the session descriptions and ICE
information into the defined messages of its chosen signaling information into the defined messages of its chosen signaling
protocol, and perform the reverse conversion on the messages it protocol, and perform the reverse conversion on the messages it
receives from the other side. receives from the other side.
One way to mitigate this is to provide a Javascript library that One way to mitigate this is to provide a JavaScript library that
hides this complexity from the developer; said library would hides this complexity from the developer; said library would
implement a given signaling protocol along with its state machine and implement a given signaling protocol along with its state machine and
serialization code, presenting a higher level call-oriented interface serialization code, presenting a higher level call-oriented interface
to the application developer. For example, libraries exist to adapt to the application developer. For example, libraries exist to adapt
the JSEP API into an API suitable for a SIP or XMPP. Thus, JSEP the JSEP API into an API suitable for a SIP or XMPP. Thus, JSEP
provides greater control for the experienced developer without provides greater control for the experienced developer without
forcing any additional complexity on the novice developer. 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. In to have to understand and handle concepts like signaling glare (see
addition, it prevented the application from driving the state machine [RFC3264], Section 4).
to a desired state, as is needed in the page reload case.
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 a feeling that requiring directly. This was rejected based on a feeling 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 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, as well as creating a large
API surface that needed to be agreed upon and documented. In API surface that needed to be agreed upon and documented. In
skipping to change at page 7, line 26 skipping to change at page 7, line 37
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 user agent needs specific In order to establish the media plane, the JSEP implementation needs
parameters to indicate what to transmit to the remote side, as well specific parameters to indicate what to transmit to the remote side,
as how to handle the media that is received. These parameters are as well as how to handle the media that is received. These
determined by the exchange of session descriptions in offers and parameters are determined by the exchange of session descriptions in
answers, and there are certain details to this process that must be offers and answers, and there are certain details to this process
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; for example, the However, not all parameters follow this rule; for example, the
fingerprints [I-D.ietf-mmusic-4572-update] sent to a remote party are fingerprints [RFC8122] sent to a remote party are calculated based on
calculated based on the local certificate(s) offered; the remote the local certificate(s) offered; the remote party MUST either accept
party MUST either accept these parameters or reject them altogether, these parameters or reject them altogether, with no option to choose
with no option to choose different values. different values.
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 only known only after an
offer and an answer have been exchanged, it is possible for the offer and an answer have been exchanged, the offerer may receive ICE
offerer to receive media after they have sent an offer and before checks, and possibly media (e.g., in the case of a re-offer after a
they have received an answer. To properly process incoming media in connection has been established) before it receives an answer. To
this case, the offerer's media handler must be aware of the details properly process incoming media in this case, the offerer's media
of the offer before the answer arrives. handler must be aware of the details of the offer before the answer
arrives.
Therefore, in order to handle session descriptions properly, the user Therefore, in order to handle session descriptions properly, the JSEP
agent needs: implementation needs:
1. To know if a session description pertains to the local or remote 1. To know if a session description pertains to the local or remote
side. side.
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
skipping to change at page 9, line 4 skipping to change at page 9, line 15
during call setup. Note that the final answer itself may be during call setup. Note that the final answer itself may be
different than any received provisional answers. 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 media stack
level, a new offer can be generated at any point. For example, when level, a new offer can be generated at any point. For example, when
using SIP for signaling, if one offer is sent, then cancelled using a using SIP for signaling, if one offer is sent, then cancelled 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 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 |
+---------------+ | +---------------+ | +---------------+ | +---------------+ |
| |----/ | |----/ | |----/ | |----/
| | setLocal(PRANSWER) | | | have- | setLocal(PRANSWER) | have- |
| Remote-Offer |------------------- >| Local-Pranswer| | remote-offer |------------------- >| local-pranswer|
| | | | | | | |
| | | | | | | |
+---------------+ +---------------+ +---------------+ +---------------+
^ | | ^ | |
| | setLocal(ANSWER) | | | setLocal(ANSWER) |
setRemote(OFFER) | | setRemote(OFFER) | |
| V setLocal(ANSWER) | | V setLocal(ANSWER) |
+---------------+ | +---------------+ |
| | | | | |
| |<---------------------------+ | |<---------------------------+
| Stable | | stable |
| |<---------------------------+ | |<---------------------------+
| | | | | |
+---------------+ setRemote(ANSWER) | +---------------+ setRemote(ANSWER) |
^ | | ^ | |
| | setLocal(OFFER) | | | setLocal(OFFER) |
setRemote(ANSWER) | | setRemote(ANSWER) | |
| V | | V |
+---------------+ +---------------+ +---------------+ +---------------+
| | | | | | | |
| | setRemote(PRANSWER) | | | 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 SDP syntax for their internal
representation. While this format is not optimal for manipulation representation. While this format is not optimal for manipulation
from Javascript, it is widely accepted, and frequently updated with from JavaScript, it is widely accepted, and frequently updated with
new features; any alternate encoding of session descriptions would new features; any alternate encoding of session descriptions would
have to keep pace with the changes to SDP, at least until the time have to keep pace with the changes to SDP, at least until the time
that this new encoding eclipsed SDP in popularity. that this new encoding eclipsed SDP in popularity.
However, to simplify Javascript processing, and provide for future However, to provide for future flexibility, the SDP syntax is
flexibility, the SDP syntax is encapsulated within a encapsulated within a SessionDescription object, which can be
SessionDescription object, which can be constructed from SDP, and be constructed from SDP, and be serialized out to SDP. If future
serialized out to SDP. If future specifications agree on a JSON specifications agree on a JSON format for session descriptions, we
format for session descriptions, we could easily enable this object could easily enable this object to generate and consume that JSON.
to generate and consume that JSON.
Other methods may be added to SessionDescription in the future to
simplify handling of SessionDescriptions from Javascript. In the
meantime, Javascript libraries can be used to perform these
manipulations.
Note that 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 which 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/Answer 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 RtpSender
and an RtpReceiver, which an application can use to control the and an RtpReceiver, which an application can use to control the
sending and receiving of RTP media. The application may also modify sending and receiving of RTP media. The application may also modify
the RtpTransceiver directly, for instance, by stopping it. 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 a m= section,
or if RtpTransceivers have been stopped and disassociated from m= or if RtpTransceivers have been stopped and disassociated from m=
sections. An RtpTransceiver is said to be associated with an m= sections. An RtpTransceiver is said to be associated with an m=
section if its mid property is non-null; otherwise it is said to be section if its mid property is non-null; otherwise it is said to be
disassociated. The associated m= section is determined using a disassociated. The associated m= section is determined using a
mapping between transceivers and m= section indices, formed when mapping between transceivers and m= section indices, formed when
creating an offer or applying a remote offer. An RtpTransceiver is creating an offer or applying a remote offer.
never associated with more than one m= section, and once a session
description is applied, a m= section is always associated with An RtpTransceiver is never associated with more than one m= section,
exactly one RtpTransceiver. and once a session description is applied, a m= section is always
associated with exactly one RtpTransceiver. However, in certain
cases where a m= section has been rejected, as discussed in
Section 5.2.2 below, that m= section will be "recycled" and
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
skipping to change at page 12, line 20 skipping to change at page 13, line 16
application that gathering is occurring through an event. Then, when application that gathering is occurring through an event. Then, when
each new ICE candidate becomes available, the ICE agent will supply each new ICE candidate becomes available, the ICE agent will supply
it to the application via an additional event; these candidates will it to the application via an additional event; these candidates will
also automatically be added to the current and/or pending local also automatically be added to the current and/or pending local
session description. Finally, when all candidates have been session description. Finally, when all candidates have been
gathered, an event will be dispatched to signal that the gathering gathered, an event will be dispatched to signal that the gathering
process is complete. 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 to
use their existing candidates. Also, when bundling is active, use their existing candidates. Also, if an m= section is bundled
candidates are only gathered (and exchanged) for the m= sections (either by a successful bundle negotiation or by being marked as
referenced in BUNDLE-tags, as described in bundle-only), then candidates will be gathered and exchanged for that
m= section if and only if its MID is a BUNDLE-tag, as described in
[I-D.ietf-mmusic-sdp-bundle-negotiation]. [I-D.ietf-mmusic-sdp-bundle-negotiation].
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 [I-D.ietf-ice-trickle]. This process allows the callee to begin
acting upon the call and setting up the ICE (and perhaps DTLS) acting upon the call and setting up the ICE (and perhaps DTLS)
connections immediately, without having to wait for the caller to connections immediately, without having to wait for the caller to
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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
[RFC5245], Section 15.1. For example: [RFC5245], Section 15.1. Note that this field does not contain an
"a=" prefix, as indicated in the following 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 ufrag
it is associated with, as defined in [RFC5245], Section 15.4. This it is associated with, as defined in [RFC5245], Section 15.4. This
value is used to determine which session description (and thereby value is used to determine which session description (and thereby
which gathering phase) this IceCandidate belongs to, which helps which gathering phase) this IceCandidate belongs to, which helps
resolve ambiguities during ICE restarts. If this field is absent in resolve ambiguities during ICE restarts. If this field is absent in
a received IceCandidate (perhaps when communicating with a non-JSEP a received IceCandidate (perhaps when communicating with a non-JSEP
endpoint), the most recently received session description is assumed. endpoint), the most recently received session description is assumed.
skipping to change at page 14, line 35 skipping to change at page 15, line 31
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
this is where the app specifies the number of media streams, and the local description indicates the number of ICE components which
thereby ICE components, for which to gather candidates. However, to will be needed and for which candidates must be gathered. However,
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 goes to gather the needed ICE candidates, it SHOULD
start by checking if any candidates are available in the pool. If start by checking if any candidates are available in the pool. If
there are candidates in the pool, they SHOULD be handed to the 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
skipping to change at page 15, line 21 skipping to change at page 16, line 17
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.
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. maximum set by policy. By specifying these limits in an
"a=imageattr" attribute, JSEP endpoints can attempt to ensure that
the remote sender transmits video at an acceptable resolution.
However, when communicating with a non-JSEP endpoint that does not
understand this attribute, any signaled limits may be exceeded, and
the JSEP implementation MUST handle this gracefully, e.g., by
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 intersect any known local limits (e.g.,
hardware decoder capababilities, local policy) to determine the hardware decoder capababilities, 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
degenerate case of no permitted resolutions, the "a=imageattr"
attribute MUST be omitted, and the m= section MUST be marked as
sendonly/inactive, as appropriate.
Otherwise, an "a=imageattr" attribute is created with "recv" Otherwise, an "a=imageattr" attribute is created with "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. If the intersection is the null set, i.e., maximum x= and y= values.
the degenerate case of no permitted resolutions, this MUST be
represented by x=0 and y=0 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
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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 an imageattr Attribute 3.6.2. Interpreting an imageattr Attribute
[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 sender This specification prescribes more specific behavior. When a
of a given MediaStreamTrack, which is producing video of a certain MediaStreamTrack, which is producing video of a certain resolution
resolution, receives an "a=imageattr recv" attribute, it MUST check (the "track resolution"), is attached to a RtpSender, which is
to see if the original resolution meets the size criteria specified encoding the track video at the same or lower resolution(s) (the
in the attribute, and adapt the resolution accordingly by scaling (if "encoder resolutions"), and a remote description is applied that
appropriate). Note that when considering a MediaStreamTrack that is references the sender and contains valid "a=imageattr recv"
producing rotated video, the unrotated resolution MUST be used. This attributes, it MUST follow the rules below to ensure the sender does
is required regardless of whether the receiver supports performing not transmit a resolution that would exceed the size criteria
receive-side rotation (e.g., through CVO [TS26.114]), as it specified in the attributes. These rules MUST be followed as long as
significantly simplifies the matching logic. the attributes remain present in the remote description, including
cases in which the track changes its resolution, or is replaced with
For the purposes of resolution negotiation, only size limits are a different track.
considered. Any other values, e.g. picture or sample aspect ratio,
MUST be ignored.
When communicating with a non-JSEP endpoint, multiple relevant
"a=imageattr recv" attributes may be present in a received m=
section. If this occurs, attributes other than the one with the
highest "q=" value MUST be ignored. If multiple attributes have the
same "q=" value, those that appear after the first such attribute in
the m= section MUST be ignored.
If an "a=imageattr recv" attribute references a different video Depending on how the RtpSender is configured, it may be producing a
payload type than what has been selected for sending the single encoding at a certain resolution, or, if simulcast Section 3.7
MediaStreamTrack, it MUST be ignored. has been negotiated, multiple encodings, each at their own specific
resolution. In addition, depending on the configuration, each
encoding may have the flexibility to reduce resolution when needed,
or may be locked to a specific output resolution.
If the original resolution matches the size limits in the attribute, For each encoding being produced by the RtpSender, the following
the track MUST be transmitted untouched. rules are applied to determine what should be transmitted:
If the original resolution exceeds the size limits in the attribute, o First, the most suitable "a=imageattr recv" attribute is selected.
the sender SHOULD apply downscaling to the output of the This is performed by taking the attribute with the highest "q="
MediaStreamTrack in order to satisfy the limits. Downscaling MUST value from the set of attributes that reference the media format
NOT change the track aspect ratio. that has been selected for the specified encoding. If multiple
attributes have the same "q=" value, the one that appears first in
the m= section is used. Note that while JSEP endpoints will
include at most one "a=imageattr recv" attribute per media format,
JSEP endpoints may receive session descriptions from non-JSEP
endpoints with m= sections that contain multiple such attributes.
If the original resolution is less than the size limits in the o If there is an applicable "a=imageattr recv" attribute for the
attribute, upscaling is needed, but this may not be appropriate in encoding, the limits from the attribute are then compared to the
all cases. To address this concern, the application can set an encoder resolution. Only the specific limits mentioned below are
upscaling policy for each sent track. For this case, if upscaling is considered; any other values, such as picture aspect ratio, MUST
permitted by policy, the sender SHOULD apply upscaling in order to be ignored. Note that when considering a MediaStreamTrack that is
provide the desired resolution. Otherwise, the sender MUST NOT apply producing rotated video, the unrotated resolution MUST be used for
upscaling. The sender SHOULD NOT upscale in other cases, even if the the checks. This is required regardless of whether the receiver
policy permits it. Upscaling MUST NOT change the track aspect ratio. supports performing receive-side rotation (e.g., through CVO
[TS26.114]), as it significantly simplifies the matching logic.
If there is no appropriate and permitted scaling mechanism that o If the attribute includes a "sar=" (sample aspect ratio) value set
allows the received size limits to be satisfied, the sender MUST NOT to something other than "1.0", indicating the receiver wants to
transmit the track. receive non-square pixels, this cannot be satisfied and the sender
MUST NOT transmit the encoding.
If the attribute includes a "sar=" (sample aspect ratio) value set to o If the encoder resolution exceeds the maximum size permitted by
something other than "1.0", indicating the receiver wants to receive the attribute, and the encoder is allowed to adjust its
non-square pixels, this cannot be satisfied and the sender MUST NOT resolution, the encoder SHOULD apply downscaling in order to
transmit the track. satisfy the limits, although the downscaling MUST NOT change the
picture aspect ratio of the encoding. For example, if the encoder
resolution is 1280x720, and the attribute specified a maximum of
640x480, the expected output resolution would be 640x360. If
downscaling cannot be applied, the encoding MUST NOT be
transmitted, and an error SHOULD be surfaced to the application.
In the special case of receiving a maximum resolution of [0, 0], as o If the encoder resolution is less than the minimum size permitted
described above, the sender MUST NOT transmit the track. by the attribute, the encoding MUST NOT be transmitted, and an
error SHOULD be surfaced to the application; the encoder MUST NOT
apply upscaling. JSEP implementations SHOULD avoid this situation
by allowing receipt of arbitrarily small resolutions, perhaps via
fallback to a software decoder.
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 to
allow applications to send simulcasted media but only to receive a 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.
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"a=rid" attribute for each encoding, as specified in "a=rid" attribute for each encoding, as specified in
[I-D.ietf-mmusic-rid], Section 4; the use of RID identifiers allows [I-D.ietf-mmusic-rid], Section 4; the use of RID identifiers allows
the individual encodings to be disambiguated even though they are all the individual encodings to be disambiguated 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 of 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:
o Start exchanging media with a given remote peer, but keep all the o Start exchanging media with a given remote peer, but keep all the
resources reserved in the offer. resources reserved in the offer.
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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 Section 3.1 of possibilities for handling this are described in [RFC3960],
[RFC3960]. Most SIP devices today only support exchanging media with Section 3.1. Most SIP devices today only support exchanging media
a single device at a time, and do not try to mix multiple early media with a single device at a time, and do not try to mix multiple early
audio sources, as that could result in a confusing situation. For media audio sources, as that could result in a confusing situation.
example, consider having a European ringback tone mixed together with For example, consider having a European ringback tone mixed together
the North American ringback tone - the resulting sound would not be with the North American ringback tone - the resulting sound would not
like either tone, and would confuse the user. If the signaling 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
and making sure that the same local media streams have been added and making sure that the same local media streams have been added
into this new PeerConnection. Then the new PeerConnection object into this new PeerConnection. Then the new PeerConnection object
would produce a SDP offer that could be used by the signaling to would produce a SDP offer that could be used by the signaling to
perform the UPDATE strategy discussed in [RFC3960]. 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
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"rollback" with empty contents to either setLocalDescription or "rollback" with empty contents to either setLocalDescription or
setRemoteDescription, depending on which was most recently used (i.e. setRemoteDescription, depending on which was most recently used (i.e.
if the new offer was supplied to setLocalDescription, the rollback if the new offer was supplied to setLocalDescription, the rollback
should be done using setLocalDescription as well). should be done using setLocalDescription as well).
4.1.9. setLocalDescription 4.1.9. 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, or final answer; offers and answers are an offer, provisional answer, final answer, or rollback; offers and
checked differently, using the various rules that exist for each SDP answers are checked differently, using the various rules that exist
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 until PeerConnection enters the "have-local-offer" state, and continues
setRemoteDescription is called with either a final answer, at which until setRemoteDescription is called with either a final answer, at
point the PeerConnection can fully adopt the pending local which point the PeerConnection can fully adopt the pending local
description, or a rollback, which results in a revert to the current description, or a rollback, which results in a revert to the current
local description. 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.
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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 future
calls to createOffer and createAnswer. calls to createOffer and createAnswer.
When creating offers, the transceiver direction is directly reflected When creating offers, the transceiver direction is directly reflected
in the output, even for reoffers. 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 the Section 5.3 section 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
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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. This includes codecs formats that would otherwise not be present.
that were not negotiated in a previous offer/answer exchange that
included the transceiver.
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.
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profile and MUST indicate this profile for each data m= line they profile and MUST indicate this profile for each data m= line they
produce in an offer. Because ICE can select either UDP [RFC5245] or produce in an offer. Because ICE can select either UDP [RFC5245] or
TCP [RFC6544] transport depending on network conditions, this TCP [RFC6544] transport depending on network conditions, this
advertisement is consistent with ICE eventually selecting either advertisement is consistent with ICE eventually selecting either
either UDP or TCP. either UDP or TCP.
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,TCP)/TLS/RTP/SAVPF". In order to willingness to support "UDP/TLS/RTP/SAVPF" or "TCP/TLS/RTP/SAVPF".
simplify compatibility with such endpoints, JSEP implementations MUST In order to simplify compatibility with such endpoints, JSEP
follow the following rules when processing the media m= sections in implementations MUST follow the following rules when processing the
an offer: media m= sections in a received offer:
o The profile in any "m=" line in any answer MUST exactly match the o Any profile in the offer matching one of the following MUST be
profile provided in the offer. accepted:
o Any profile matching the following patterns MUST be accepted: * "RTP/AVP" (Defined in [RFC4566], Section 8.2.2)
"RTP/[S]AVP[F]" and "(UDP/TCP)/TLS/RTP/SAVP[F]"
* "RTP/AVPF" (Defined in [RFC4585], Section 9)
* "RTP/SAVP" (Defined in [RFC3711], Section 12)
* "RTP/SAVPF" (Defined in [RFC5124], Section 6)
* "TCP/DTLS/RTP/SAVP" (Defined in [RFC7850], Section 3.4)
* "TCP/DTLS/RTP/SAVPF" (Defined in [RFC7850], Section 3.5)
* "UDP/TLS/RTP/SAVP" (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
match the profile provided in the offer.
o Because DTLS-SRTP is REQUIRED, the choice of SAVP or AVP has no o 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" attribute. 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 o 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 present, assume AVPF timing, i.e., a default value of "trr- is present, assume AVPF timing, i.e., a default value of "trr-
int=0". Otherwise, assume that AVPF is being used in an AVP int=0". Otherwise, assume that AVPF is being used in an AVP
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The value of the <nettype> <addrtype> <unicast-address> tuple The value of the <nettype> <addrtype> <unicast-address> tuple
SHOULD be set to a non-meaningful address, such as IN IP4 0.0.0.0, SHOULD be set to a non-meaningful address, such as IN IP4 0.0.0.0,
to prevent leaking the local address in this field. As mentioned to prevent leaking the local address in this field. As mentioned
in [RFC4566], the entire o= line needs to be unique, but selecting in [RFC4566], the entire o= line needs to be unique, but selecting
a random number for <sess-id> is sufficient to accomplish this. a random number for <sess-id> is sufficient to accomplish this.
o The third SDP line MUST be a "s=" line, as specified in [RFC4566], o The third SDP line MUST be a "s=" line, as specified 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, having the same meaningless value "o=" and "s=" are meaningless in JSEP, having the same meaningless
seems clearer. value seems clearer.
o Session Information ("i="), URI ("u="), Email Address ("e="), o 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 o 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; o 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. "t=0
0". 0".
o An "a=ice-options" line with the "trickle" option MUST be added, o An "a=ice-options" line with the "trickle" option MUST be added,
as specified in [I-D.ietf-ice-trickle], Section 4. as specified in [I-D.ietf-ice-trickle], Section 4.
o If WebRTC identity is being used, an "a=identity" line as o If WebRTC identity is being used, an "a=identity" line as
described in [I-D.ietf-rtcweb-security-arch], Section 5. described in [I-D.ietf-rtcweb-security-arch], 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 [RFC4566],
Section 5.14. An m= section is generated for each RtpTransceiver Section 5.14. An m= section is generated for each RtpTransceiver
that has been added to the PeerConnection, excluding any stopped that has been added to the PeerConnection, excluding any stopped
RtpTransceivers. This is done in the order the RtpTransceivers were RtpTransceivers. This is done in the order the RtpTransceivers were
added to the PeerConnection. added to the PeerConnection.
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 generate a unique set of ICE credentials and gather its own unique
set of ICE candidates. Bundle-only m= sections MUST NOT contain any set of ICE candidates. Bundle-only m= sections MUST NOT contain any
ICE credentials and MUST NOT gather any candidates. ICE 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 all the certificate(s) that have
been specified for the PeerConnection; as a result, they MUST all been specified for the PeerConnection; as a result, they MUST all
have the same [I-D.ietf-mmusic-4572-update] fingerprint value(s), or have the same [RFC8122] fingerprint value(s), or these value(s) MUST
these value(s) MUST be session-level attributes. be session-level attributes.
Each m= section should be generated as specified in [RFC4566], Each m= section should 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 The port value is set to the port of the default ICE candidate for o If the m= section is marked as bundle-only, then the port value
this m= section, but given that no candidates are available yet, MUST be set to 0. Otherwise, the port value is set to the port of
the "dummy" port value of 9 (Discard) MUST be used, as indicated the default ICE candidate for this m= section, but given that no
in [I-D.ietf-ice-trickle], Section 5.1. candidates are available yet, the "dummy" port value of 9
(Discard) MUST be used, as indicated in [I-D.ietf-ice-trickle],
Section 5.1.
o To properly indicate use of DTLS, the <proto> field MUST be set to o 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, o 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 The media formats in the answer MAY include codecs present in the
offer that were discarded in a previous offer/answer exchange.
This is necessary for compatibility with third- party call control
and SIP use cases.
o Unless excluded by the above restrictions, the media formats MUST o 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
[I-D.ietf-rtcweb-audio] (see Section 3) and [RFC7874], Section 3, and [RFC7742], Section 5.
[I-D.ietf-rtcweb-video] (see 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 specified
in [RFC4566], Section 5.7. Again, as no candidates are available in [RFC4566], Section 5.7. Again, as no candidates are available
yet, the "c=" line must contain the "dummy" value "IN IP4 0.0.0.0", yet, the "c=" line must contain the "dummy" value "IN IP4 0.0.0.0",
as defined in [I-D.ietf-ice-trickle], Section 5.1. as defined in [I-D.ietf-ice-trickle], Section 5.1.
[I-D.ietf-mmusic-sdp-mux-attributes] groups SDP attributes into [I-D.ietf-mmusic-sdp-mux-attributes] groups SDP attributes into
different categories. To avoid unnecessary duplication when different categories. To avoid unnecessary duplication when
bundling, Section 8.1 of [I-D.ietf-mmusic-sdp-bundle-negotiation] bundling, attributes of category IDENTICAL or TRANSPORT MUST NOT be
specifies that attributes of category IDENTICAL or TRANSPORT should repeated in bundled m= sections, repeating the guidance from
not be repeated in bundled m= sections. [I-D.ietf-mmusic-sdp-bundle-negotiation], Section 8.1. This includes
m= sections for which bundling has been negotiated and is still
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 o 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
[I-D.ietf-mmusic-sdp-bundle-negotiation], Section 14. Note that [I-D.ietf-mmusic-sdp-bundle-negotiation], Section 14. Note that
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limitations on the size of images which can be decoded, an limitations on the size of images which 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 o 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
[I-D.ietf-rtcweb-rtp-usage], Section 5.2. Any header extensions [I-D.ietf-rtcweb-rtp-usage], Section 5.2. Any header extensions
that require encryption MUST be specified as indicated in that require encryption MUST be specified as indicated in
[RFC6904], Section 4. [RFC6904], Section 4.
o For each supported RTCP feedback mechanism, an "a=rtcp-fb" o For each supported RTCP feedback mechanism, an "a=rtcp-fb" line,
mechanism, as specified in [RFC4585], Section 4.2. The list of as specified in [RFC4585], Section 4.2. The list of RTCP feedback
RTCP feedback mechanisms that SHOULD/MUST be supported is mechanisms that SHOULD/MUST be supported is specified in
specified in [I-D.ietf-rtcweb-rtp-usage], Section 5.1. [I-D.ietf-rtcweb-rtp-usage], Section 5.1.
o If the RtpTransceiver has a sendrecv or sendonly direction: o 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. If a line, as specified in [I-D.ietf-mmusic-msid], Section 2. If a
MediaStreamTrack is attached to the transceiver's RtpSender, MediaStreamTrack is attached to the transceiver's RtpSender,
the "a=msid" lines MUST use that track's ID. If no the "a=msid" lines MUST use that track's ID. If no
MediaStreamTrack is attached, a valid ID MUST be generated, in MediaStreamTrack is attached, a valid ID MUST be generated, in
the same way that the implementation generates IDs for local the same way that the implementation generates IDs for local
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o If the bundle policy for this PeerConnection is set to "max- o If 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 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 which either have a
unique address or which are associated with the bundle-tag. (In unique address or which are associated with the bundle-tag. (In
initial offers, this means those "m=" sections which do not contain initial offers, this means those "m=" sections which do not contain
an "a=bundle-only" attribute.) an "a=bundle-only" attribute.)
o "a=ice-ufrag" and "a=ice-pwd" lines, as specified in [RFC5245], o "a=ice-ufrag" and "a=ice-pwd" lines, as specified in [RFC5245],
Section 15.4. Section 15.4.
o An "a=fingerprint" line for each of the endpoint's certificates, o For each desired digest algorithm, one or more "a=fingerprint"
as specified in [RFC4572], Section 5; the digest algorithm used lines for each of the endpoint's certificates, as specified in
for the fingerprint MUST match that used in the certificate [RFC8122], Section 5.
signature.
o An "a=setup" line, as specified in [RFC4145], Section 4, and o 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=dtls-id" line, as specified in [I-D.ietf-mmusic-dtls-sdp] o An "a=tls-id" line, as specified in [I-D.ietf-mmusic-dtls-sdp],
Section 5.2. Section 5.2.
o An "a=rtcp" line, as specified in [RFC3605], Section 2.1, o 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 dummy 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. o 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- o 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 [I-D.ietf-mmusic-mux-exclusive],
skipping to change at page 41, line 4 skipping to change at page 42, line 23
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" o "a=ice-ufrag"
o "a=ice-pwd" o "a=ice-pwd"
o "a=fingerprint" o "a=fingerprint"
o "a=setup" o "a=setup"
o "a=dtls-id" o "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 implementation
offers all m= sections as one bundle group. However, whether the m= offers all m= sections as one bundle group. However, whether the m=
sections are bundle-only or not depends on the bundle policy. 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 which SDP permits to either be at 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 promotes readability, especially if one of a set of identical. This assists development and debugging by making it
initially identical attributes is subsequently changed. easier to understand individual media sections, especially if one of
a set of initially identical attributes is subsequently changed.
However, implementations MAY choose to aggregate attributes at the
session level and JSEP implementations MUST be prepared to receive
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 [I-D.ietf-rtcweb-rtp-usage],
and MUST NOT be included: and MUST NOT be included:
o "a=crypto" o "a=crypto"
o "a=key-mgmt" o "a=key-mgmt"
skipping to change at page 42, line 11 skipping to change at page 43, line 31
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 initial 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 should be followed, subject to the
following restriction: following restriction:
o The fields of the "o=" line MUST stay the same except for the o 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
skipping to change at page 42, line 37 skipping to change at page 44, line 8
Note that if the application creates an offer by reading Note that if the application creates an offer by reading
currentLocalDescription instead of calling createOffer, the returned currentLocalDescription instead of calling createOffer, the returned
SDP may be different than when setLocalDescription was originally SDP may be different than when setLocalDescription was originally
called, due to the addition of gathered ICE candidates, but the called, due to the addition of gathered ICE candidates, but the
<session-version> will not have changed. There are no known <session-version> will not have changed. There are no known
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 initial offer was applied using setLocalDescription, but an If the previous offer was applied using setLocalDescription, but a
answer from the remote side has not yet been applied, meaning the corresponding answer from the remote side has not yet been applied,
PeerConnection is still in the "local-offer" state, an offer is meaning the PeerConnection is still in the "have-local-offer" state,
generated by following the steps in the "stable" state above, along an offer is generated by following the steps in the "stable" state
with these exceptions: above, along with these exceptions:
o The "s=" and "t=" lines MUST stay the same. o The "s=" and "t=" lines MUST stay the same.
o If any RtpTransceiver has been added, and there exists an m= o 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 m=
section were being added to the session description, placed at the section were being added to the session description (including a
same index as the m= section with a zero port. new MID value), and placing it at the same index as the m= section
with a zero port.
o If an RtpTransceiver is stopped and is not associated with an m= o 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= o If an RtpTransceiver has been stopped and is associated with an m=
section, and the m= section is not being recycled as described section, and the m= section is not being recycled as described
above, an m= section MUST be generated for it with the port set to above, an m= section MUST be generated for it with the port set to
skipping to change at page 44, line 16 skipping to change at page 45, line 37
Section 9.3. If the m= section is bundled into another m= Section 9.3. If the m= section is bundled into another m=
section, both "a=candidate" and "a=end-of-candidates" MUST be section, both "a=candidate" and "a=end-of-candidates" MUST be
omitted. omitted.
o For RtpTransceivers that are still present, the "a=rid" lines MUST o 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 o For RtpTransceivers that are still present, any "a=simulcast" line
MUST stay the same. MUST stay the same.
o If any RtpTransceiver has been stopped, the port MUST be set to If the previous offer was applied using setLocalDescription, and a
zero and all "a=msid" lines MUST be removed. corresponding answer from the remote side has been applied using
setRemoteDescription, meaning the PeerConnection is in the "have-
o If any RtpTransceiver has been added, and there exists a m= remote-pranswer" or "stable" states, an offer is generated based on
section with a zero port in the current local description or the the negotiated session descriptions by following the steps mentioned
current remote description, that m= section MUST be recycled by for the "have-local-offer" state above.
generating a m= section for the added RtpTransceiver as if the m=
section were being added to session description, except that
instead of adding it, the generated m= section replaces the m=
section with a zero port. The new m= section MUST contain a new
MID.
If the initial offer was applied using setLocalDescription, and an
answer from the remote side has been applied using
setRemoteDescription, meaning the PeerConnection is in the "remote-
pranswer" or "stable" states, an offer is generated based on the
negotiated session descriptions by following the steps mentioned for
the "local-offer" state above.
In addition, for each non-recycled, non-rejected m= section in the In addition, for each existing, non-recycled, non-rejected m= section
new offer, the following adjustments are made based on the contents in the new offer, the following adjustments are made based on the
of the corresponding m= section in the current remote description, if contents of the corresponding m= section in the current local or
any: remote description, as appropriate:
o The m= line and corresponding "a=rtpmap" and "a=fmtp" lines MUST o The m= line and corresponding "a=rtpmap" and "a=fmtp" lines MUST
only include codecs present in the most recent answer which have only include media formats which have not been excluded by the
not been excluded by the codec preferences of the associated codec preferences of the associated transceiver, and MUST include
transceiver. Note that non-JSEP endpoints are not subject to all currently available formats. Media formats that were
these restrictions, and might offer media formats that were not previously offered but are no longer available (e.g., a shared
present in the most recent answer, as specified in [RFC3264], hardware codec) MAY be excluded.
Section 8. Therefore, JSEP implementations MUST be prepared to
receive such offers.
o Unless codec preferences have been set for the associated o 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 in
the same order as in the current local description. the same order as in the most recent answer. Any media formats
that were not present in the most recent answer MUST be added
after all existing formats.
o The RTP header extensions MUST only include those that are present o 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 extensions MUST only include those that are o The RTCP feedback mechanisms MUST only include those that are
present in the most recent answer. present in the most recent answer, except for the case of format-
specific mechanisms that are referencing a newly-added media
format.
o The "a=rtcp" line MUST only be added if the most recent answer did o The "a=rtcp" line MUST NOT be added if the most recent answer
not include an "a=rtcp-mux" line. included an "a=rtcp-mux" line.
o The "a=rtcp-mux" line MUST only be added if present in the most o The "a=rtcp-mux" line MUST be the same as that in the most recent
recent answer. answer.
o The "a=rtcp-mux-only" line MUST NOT be added. o The "a=rtcp-mux-only" line MUST NOT be added.
o The "a=rtcp-rsize" line MUST only be added if present in the most o The "a=rtcp-rsize" line MUST NOT be added unless present in the
recent answer. most recent answer.
o An "a=bundle-only" line MUST NOT be added, as indicated in o An "a=bundle-only" line MUST NOT be added, as indicated in
[I-D.ietf-mmusic-sdp-bundle-negotiation], Section 6. Instead, [I-D.ietf-mmusic-sdp-bundle-negotiation], Section 6. Instead,
JSEP implementations MUST simply omit parameters in the IDENTICAL JSEP implementations MUST simply omit parameters in the IDENTICAL
and TRANSPORT categories for bundled m= sections, as described in and TRANSPORT categories for bundled m= sections, as described in
[I-D.ietf-mmusic-sdp-bundle-negotiation], Section 8.1. [I-D.ietf-mmusic-sdp-bundle-negotiation], Section 8.1.
o Note that if media m= sections are bundled into a data m= section, o Note that if media m= sections are bundled into a data m= section,
then certain TRANSPORT and IDENTICAL attributes may appear in the then certain TRANSPORT and IDENTICAL attributes may appear in the
data m= section even if they would otherwise only be appropriate data m= section even if they would otherwise only be appropriate
skipping to change at page 46, line 27 skipping to change at page 47, line 37
pwd attributes, as specified in [RFC5245], Section 9.1.1.1. If this pwd attributes, as specified in [RFC5245], Section 9.1.1.1. If this
option is specified on an initial offer, it has no effect (since a option is specified on an initial offer, it has no effect (since a
new ICE ufrag and pwd are already generated). Similarly, if the ICE new ICE ufrag and pwd are already generated). Similarly, if the ICE
configuration has changed, this option has no effect, since new ufrag configuration has changed, this option has no effect, since new ufrag
and pwd attributes will be generated automatically. This option is and pwd attributes will be generated automatically. This option is
primarily useful for reestablishing connectivity in cases where primarily useful for reestablishing connectivity in cases where
failures are detected by the application. failures are detected by the application.
5.2.3.2. VoiceActivityDetection 5.2.3.2. VoiceActivityDetection
Silence suppression, also known as discontinuous transmission
("DTX"), can reduce the bandwidth used for audio by switching to a
special encoding when voice activity is not detected, at the cost of
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
the audio it receives by including comfort noise ("CN") codecs for the audio it receives by including comfort noise ("CN") codecs for
each offered audio codec, as specified in [RFC3389], Section 5.1, each offered audio codec, as specified in [RFC3389], Section 5.1,
except for codecs that have their own internal silence suppression except for codecs that have their own internal silence suppression
support. For codecs that have their own internal silence suppression support. For codecs that have their own internal silence suppression
support, the appropriate fmtp parameters for that codec MUST be support, the appropriate fmtp parameters for that codec MUST be
specified to indicate that silence suppression for received audio is specified to indicate that silence suppression for received audio is
desired. For example, when using the Opus codec [RFC6716], the desired. For example, when using the Opus codec [RFC6716], the
"usedtx=1" parameter, specified in [RFC7587], would be used in the "usedtx=1" parameter, specified in [RFC7587], would be used in the
offer. This option allows the endpoint to significantly reduce the offer.
amount of audio bandwidth it receives, at the cost of some fidelity,
depending on the quality of the remote VAD algorithm.
If the "VoiceActivityDetection" option is specified, with a value of If the "VoiceActivityDetection" option is specified, with a value of
"false", the JSEP implementation MUST NOT emit "CN" codecs. For "false", the JSEP implementation MUST NOT emit "CN" codecs. For
codecs that have their own internal silence suppression support, the codecs that have their own internal silence suppression support, the
appropriate fmtp parameters for that codec MUST be specified to appropriate fmtp parameters for that codec MUST be specified to
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. would be specified in the offer. In addition, the implementation
MUST NOT use silence suppression for media it generates, regardless
Note that setting the "VoiceActivityDetection" parameter when of whether the "CN" codecs or related fmtp parameters appear in the
generating an offer is a request to receive audio with silence peer's description. The impact of these rules is that silence
suppression. It has no impact on whether the local endpoint does suppression in JSEP depends on mutual agreement of both sides, which
silence suppression for the audio it sends. 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 [I-D.ietf-rtcweb-rtp-usage]. The exact
skipping to change at page 49, line 31 skipping to change at page 50, line 43
The next step is to go through each offered m= section. Each offered The next step is to go through each offered m= section. Each offered
m= section will have an associated RtpTransceiver, as described in m= section will have an associated RtpTransceiver, as described in
Section 5.9. If there are more RtpTransceivers than there are m= Section 5.9. If there are more RtpTransceivers than there are m=
sections, the unmatched RtpTransceivers will need to be associated in sections, the unmatched RtpTransceivers will need to be associated in
a subsequent 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 in
[RFC3264], Section 6., and further processing for this m= section can [RFC3264], Section 6, and further processing for this m= section can
be skipped: be skipped:
o The associated RtpTransceiver has been stopped. o The associated RtpTransceiver has been stopped.
o No supported codec is present in the offer. o None of the offered media formats are supported and, if
applicable, allowed by codec preferences.
o The bundle policy is "max-bundle", and this is not the first m= o 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= o 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.
Otherwise, each m= section in the answer should then be generated as Otherwise, each m= section in the answer should 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
skipping to change at page 50, line 11 skipping to change at page 51, line 25
o The port value would normally be set to the port of the default o 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 "dummy" port value of 9 (Discard) MUST be
used, as indicated in [I-D.ietf-ice-trickle], Section 5.1. used, as indicated in [I-D.ietf-ice-trickle], Section 5.1.
o The <proto> field MUST be set to exactly match the <proto> field o 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, o 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,
MUST exclude any codecs not present in the codec preferences or regardless of what was offered, and MUST exclude any codecs not
not present in the offer. Note that non-JSEP endpoints are not present in the codec preferences.
subject to this restriction, and might add media formats in the
answer that are not present in the offer, as specified in
[RFC3264], Section 6.1. Therefore, JSEP implementations MUST be
prepared to receive such answers.
o Unless excluded by the above restrictions, the media formats MUST o Otherwise, the media formats on the m= line MUST be generated in
include the mandatory audio/video codecs as specified in the same order as those offered in the current remote description,
[I-D.ietf-rtcweb-audio] (see Section 3) and excluding any currently unsupported formats. Any currently
[I-D.ietf-rtcweb-video] (see Section 5). available media formats that are not present in the current remote
description MUST be added after all existing formats.
o In either case, the media formats in the answer MUST include at
least one format that is present in the offer, but MAY include
formats that are locally supported but not present in the offer,
as mentioned in [RFC3264], Section 6.1. If no common format
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 specified
in [RFC4566], Section 5.7. Again, as no candidates are available in [RFC4566], Section 5.7. Again, as no candidates are available
yet, the "c=" line must contain the "dummy" value "IN IP4 0.0.0.0", yet, the "c=" line must contain the "dummy" value "IN IP4 0.0.0.0",
as defined in [I-D.ietf-ice-trickle], Section 5.1. as defined in [I-D.ietf-ice-trickle], Section 5.1.
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 use
the same ICE credentials and candidates; all m= sections not being the same ICE credentials and candidates; all m= sections not being
bundled must use unique ICE credentials and candidates. Each m= bundled must use unique ICE credentials and candidates. Each m=
section MUST contain the following attributes (which are of attribute section MUST contain the following attributes (which are of attribute
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"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 o 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 [I-D.ietf-rtcweb-rtp-usage], Section 5.2. Any header
extensions that require encryption MUST be specified as indicated extensions that require encryption MUST be specified as indicated
in [RFC6904], Section 4. in [RFC6904], Section 4.
o For each supported RTCP feedback mechanism that is present in the o For each supported RTCP feedback mechanism that is present in the
offer, an "a=rtcp-fb" mechanism, 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 [I-D.ietf-rtcweb-rtp-usage],
Section 5.1. Section 5.1.
o If the RtpTransceiver has a sendrecv or sendonly direction: o 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. If a line, as specified in [I-D.ietf-mmusic-msid], Section 2. If a
MediaStreamTrack is attached to the transceiver's RtpSender, MediaStreamTrack is attached to the transceiver's RtpSender,
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MediaStream ID, as specified in [I-D.ietf-mmusic-msid], MediaStream ID, as specified in [I-D.ietf-mmusic-msid],
Section 3. The track ID MUST be selected as described above. Section 3. The track ID MUST be selected as described above.
Each m= section which is not bundled into another m= section, MUST Each m= section which 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 [RFC5245], o "a=ice-ufrag" and "a=ice-pwd" lines, as specified in [RFC5245],
Section 15.4. Section 15.4.
o An "a=fingerprint" line for each of the endpoint's certificates, o For each desired digest algorithm, one or more "a=fingerprint"
as specified in [RFC4572], Section 5; the digest algorithm used lines for each of the endpoint's certificates, as specified in
for the fingerprint MUST match that used in the certificate [RFC8122], Section 5.
signature.
o An "a=setup" line, as specified in [RFC4145], Section 4, and o 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"; the The role value in the answer MUST be "active" or "passive". When
"active" role is RECOMMENDED. the offer contains the "actpass" value, as will always be the case
with JSEP endpoints, the answerer SHOULD use the "active" role.
Offers from non-JSEP endpoints MAY send other values for
"a=setup", in which case the answer MUST use a value consistent
with the value in the offer.
o An "a=dtls-id" line, as specified in [I-D.ietf-mmusic-dtls-sdp] o An "a=tls-id" line, as specified in [I-D.ietf-mmusic-dtls-sdp],
Section 5.3. Section 5.3.
o If present in the offer, an "a=rtcp-mux" line, as specified in o 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 dummy value "9
IN IP4 0.0.0.0" (because no candidates have yet been gathered). 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 o If present in the offer, an "a=rtcp-rsize" line, as specified in
[RFC5506], Section 5. [RFC5506], Section 5.
skipping to change at page 53, line 4 skipping to change at page 54, line 24
"a=max-message-size" line, as defined in [I-D.ietf-mmusic-sctp-sdp], "a=max-message-size" line, as defined in [I-D.ietf-mmusic-sctp-sdp],
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" o "a=ice-ufrag"
o "a=ice-pwd" o "a=ice-pwd"
o "a=fingerprint" o "a=fingerprint"
o "a=setup" o "a=setup"
o "a=dtls-id" o "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= section,
then certain TRANSPORT and IDENTICAL attributes may also appear in then certain TRANSPORT and IDENTICAL attributes may also appear in
the data m= section even if they would otherwise only be appropriate the data m= section even if they would otherwise only be appropriate
for a media m= section (e.g., "a=rtcp-mux"). 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 the all mid identifiers from the offered
skipping to change at page 53, line 35 skipping to change at page 55, line 11
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 initial 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 should be followed,
subject to the following restriction: subject to the following restriction:
o The fields of the "o=" line MUST stay the same except for the o 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
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o The "s=" and "t=" lines MUST stay the same. o The "s=" and "t=" lines MUST stay the same.
o Each "m=" and c=" line MUST be filled in with the port and address o Each "m=" and c=" line MUST be filled in with the port and address
of the default candidate for the m= section, as described in of the default candidate for the m= section, as described in
[RFC5245], Section 4.3. Note, however, that the m= line protocol [RFC5245], Section 4.3. Note, however, that the m= line protocol
need not match the default candidate, because this protocol value need not match the default candidate, because this protocol value
must instead match what was supplied in the offer, as described must instead match what was supplied in the offer, as described
above. above.
o Unless codec preferences have been set for the associated
transceiver, the media formats on the m= line MUST be generated in
the same order as in the current local description.
o Each "a=ice-ufrag" and "a=ice-pwd" line MUST stay the same, unless o 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 [RFC5245], Section 9.2.1.1. If must be created as specified in [RFC5245], Section 9.2.1.1. If
the m= section is bundled into another m= section, it still MUST the m= section is bundled into another m= section, it still MUST
NOT contain any ICE credentials. NOT contain any ICE credentials.
o Each "a=setup" line MUST use an "active" or "passive" role value o 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
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 o If the m= section is not bundled into another m= section and RTCP
multiplexing is not active, an "a=rtcp" attribute line MUST be multiplexing is not active, an "a=rtcp" attribute line MUST be
filled in with the port and address of the default RTCP candidate. filled in with the port and address of the default RTCP candidate.
If no RTCP candidates have yet been gathered, dummy values MUST be If no RTCP candidates have yet been gathered, dummy values MUST be
used, as described in the initial answer section above. used, as described in the initial answer section above.
o If the m= section is not bundled into another m= section, for each o If the m= section is not bundled into another m= section, for each
candidate that has been gathered during the most recent gathering candidate that has been gathered during the most recent gathering
phase (see Section 3.5.1), an "a=candidate" line MUST be added, as phase (see Section 3.5.1), an "a=candidate" line MUST be added, as
defined in [RFC5245], Section 4.3., paragraph 3. If candidate defined in [RFC5245], Section 4.3., paragraph 3. If candidate
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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 should
be generated as if VoiceActivityDetection was set to false. This is be generated as if VoiceActivityDetection was set to false. This is
done on a per-codec basis (e.g., if the offerer somehow offered 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"). 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
not offer it, making silence suppression support bilateral even with
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
options or RtpTransceiver APIs MUST be used. options or RtpTransceiver APIs MUST be used.
Note that the application MAY modify the SDP to reduce the Note that the application MAY modify the SDP to reduce the
capabilities in the offer it sends to the far side (post- capabilities in the offer it sends to the far side (post-
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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 First, the type of the SessionDescription is checked against the o If the description is of type "rollback", follow the processing
current state of the PeerConnection: defined in Section 4.1.8.2 and skip the processing described in
the rest of this section.
o Otherwise, the type of the SessionDescription is checked against
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 o 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 o 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 o Next, the SessionDescription is parsed into a data structure, as
described in the Section 5.7 section below. If parsing fails for described in Section 5.7 below.
any reason, processing MUST stop and an error MUST be returned.
o Finally, the parsed SessionDescription is applied as described in o Finally, the parsed SessionDescription is applied as described in
the Section 5.8 section below. Section 5.8 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 First, the type of the SessionDescription is checked against the o If the description is of type "rollback", follow the processing
current state of the PeerConnection: defined in Section 4.1.8.2 and skip the processing described in
the rest of this section.
o Otherwise, the type of the SessionDescription is checked against
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 o 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 o Next, the SessionDescription is parsed into a data structure, as
described in the Section 5.7 section below. If parsing fails for described in Section 5.7 below. If parsing fails for any reason,
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 o Finally, the parsed SessionDescription is applied as described in
the Section 5.9 section below. Section 5.9 below.
5.7. Parsing a Session Description 5.7. Parsing a Session Description
When a SessionDescription of any type is supplied to setLocal/
RemoteDescription, the implementation must parse it and reject it if
it is invalid. The exact details of this process are explained
below.
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 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 [RFC4566]
and [RFC3264] semantics, and then either parse and store or discard and [RFC3264] semantics, and then either parse and store or discard
the provided value, as described below. the provided value, as described below.
skipping to change at page 57, line 41 skipping to change at page 59, line 16
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.
5.7.1. Session-Level Parsing 5.7.1. 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 the
same type (typically "a=") can occur in any order, and their ordering same type (typically "a=") can occur in any order.
is not meaningful.
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 not The "c=" line MUST be checked for syntax but its value is only
used. This supersedes the guidance in [RFC5245], Section 6.1, to used for ICE mismatch detection, as defined in [RFC5245],
use "ice-mismatch" to indicate mismatches between "c=" and the Section 6.1. Note that JSEP implementations should never
candidate lines; because JSEP always uses ICE, "ice-mismatch" is encounter this condition because ICE is required for WebRTC.
not useful in this context.
The "i=", "u=", "e=", "p=", "t=", "r=", "z=", and "k=" lines are The "i=", "u=", "e=", "p=", "t=", "r=", "z=", and "k=" lines are
not used by this specification; they MUST be checked for syntax not used by this specification; they MUST be checked for syntax
but their values are not 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.
skipping to change at page 58, line 41 skipping to change at page 60, line 15
o If present, a single "a=ice-ufrag" line is parsed as specified in o If present, a single "a=ice-ufrag" line is parsed as specified in
[RFC5245], Section 15.4, and the ufrag value is stored. [RFC5245], Section 15.4, and the ufrag value is stored.
o If present, a single "a=ice-pwd" line is parsed as specified in o If present, a single "a=ice-pwd" line is parsed as specified in
[RFC5245], Section 15.4, and the password value is stored. [RFC5245], Section 15.4, and the password value is stored.
o If present, a single "a=ice-options" line is parsed as specified o If present, a single "a=ice-options" line is parsed as specified
in [RFC5245], Section 15.5, and the set of specified options is in [RFC5245], Section 15.5, and the set of specified options is
stored. stored.
o Any "a=fingerprint" lines are parsed as specified in [RFC4572], o 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 o 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=dtls-id" line is parsed as specified in o If present, a single "a=tls-id" line is parsed as specified in
[I-D.ietf-mmusic-dtls-sdp] Section 5, and the dtls-id value is [I-D.ietf-mmusic-dtls-sdp] Section 5, and the tls-id value is
stored. stored.
o Any "a=identity" lines are parsed and the identity values stored o Any "a=identity" lines are parsed and the identity values stored
for subsequent verification, as specified for subsequent verification, as specified
[I-D.ietf-rtcweb-security-arch], Section 5. [I-D.ietf-rtcweb-security-arch], Section 5.
o Any "a=extmap" lines are parsed as specified in [RFC5285], o Any "a=extmap" lines are parsed as specified in [RFC5285],
Section 5, and their values are stored. Section 5, and their values are stored.
As required by [RFC4566], Section 5.13, unknown attribute lines MUST Other attributes that are not relevant to JSEP may also be present,
be ignored. and implementations SHOULD process any that they recognize. As
required by [RFC4566], Section 5.13, unknown attribute lines MUST be
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.
5.7.2. Media Section Parsing 5.7.2. 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.
skipping to change at page 60, line 12 skipping to change at page 61, line 36
o Any "a=candidate" attributes MUST be parsed as specified in o Any "a=candidate" attributes MUST be parsed as specified in
[RFC5245], Section 15.1, and their values stored. [RFC5245], Section 15.1, and their values stored.
o Any "a=remote-candidates" attributes MUST be parsed as specified o Any "a=remote-candidates" attributes MUST be parsed as specified
in [RFC5245], Section 15.2, but their values are ignored. in [RFC5245], Section 15.2, but their values are ignored.
o If present, a single "a=end-of-candidates" attribute MUST be o If present, a single "a=end-of-candidates" attribute MUST be
parsed as specified in [I-D.ietf-ice-trickle], Section 8.2, and parsed as specified in [I-D.ietf-ice-trickle], Section 8.2, and
its presence or absence flagged and stored. its presence or absence flagged and stored.
o Any "a=fingerprint" lines are parsed as specified in [RFC4572], o 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 the If the "m=" proto value indicates use of RTP, as described in
Section 5.1.2 section above, the following attribute lines MUST be Section 5.1.2 above, the following attribute lines MUST be processed:
processed:
o The "m=" fmt value MUST be parsed as specified in [RFC4566], o 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 o 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 o 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 o 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") o 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" or "a=ssrc-group" attributes MUST be parsed as o Any "a=ssrc" attributes MUST be parsed as specified in [RFC5576],
specified in [RFC5576], Sections 4.1-4.2, and their values stored. Section 4.1, and their values stored.
o Any "a=extmap" attributes MUST be parsed as specified in o 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 o 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 o 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.
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protocol value stored. protocol value stored.
o An "a=sctp-port" attribute MUST be present, and it MUST be parsed o 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 [I-D.ietf-mmusic-sctp-sdp], Section 5.2, and the
value stored. value stored.
o If present, a single "a=max-message-size" attribute MUST be parsed o 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 [I-D.ietf-mmusic-sctp-sdp], Section 6, and the
value stored. Otherwise, use the specified default. value stored. Otherwise, use the specified default.
As required by [RFC4566], Section 5.13, unknown attribute lines MUST Other attributes that are not relevant to JSEP may also be present,
be ignored. and implementations SHOULD process any that they recognize. As
required by [RFC4566], Section 5.13, unknown attribute lines MUST be
ignored.
5.7.3. Semantics Verification 5.7.3. Semantics Verification
Assuming parsing completes successfully, the parsed description is Assuming parsing completes successfully, the parsed description is
then evaluated to ensure internal consistency as well as proper 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 o For each m= section, valid values for each of the mandatory-to-use
features enumerated in Section 5.1.1 MUST be present. These 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 either be 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 [RFC5245], Section 15.4. limits specified in [RFC5245], Section 15.4.
* dtls-id value, which MUST be set according to * tls-id value, which MUST be set according to
[I-D.ietf-mmusic-dtls-sdp] Section 5. If this is a re-offer [I-D.ietf-mmusic-dtls-sdp], Section 5. If this is a re-offer
and the dtls-id value is different from that presently in use, and the tls-id value is different from that presently in use,
the DTLS connection is not being continued and the remote the DTLS connection is not being continued and the remote
description MUST be part of an ICE restart, together with new description MUST be part of an ICE restart, together with new
ufrag and password values. If this is an answer, the dtls-id ufrag and password values. If this is an answer, the tls-id
value, if present, MUST be the same as in the offer. value, if present, MUST be the same as in the offer.
* DTLS setup value, which MUST be set according to the rules * 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 o All RID values 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 o Each m= section is also checked to ensure prohibited features are
not used. If this is a local description, the "ice-lite" not used.
attribute MUST NOT be specified.
o If the RTP/RTCP multiplexing policy is "require", each m= section o If the RTP/RTCP multiplexing policy is "require", each m= section
MUST contain an "a=rtcp-mux" attribute. If an "m=" section MUST contain an "a=rtcp-mux" attribute. If an m= section contains
contains an "a=rtcp-mux-only" attribute then that section MUST an "a=rtcp-mux-only" attribute then that section MUST also contain
also contain an "a=rtcp-mux" attribute. an "a=rtcp-mux" attribute.
o If this m= section was present in the previous answer then the
state 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 o 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 the
associated offer. associated offer.
o For each m= section, the media type and protocol values MUST o For each m= section, the media type and protocol values MUST
skipping to change at page 63, line 17 skipping to change at page 64, line 47
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.
Next, m= sections are processed. For each m= section, the following Next, m= sections are processed. For each m= section, the following
steps MUST be performed; if any parameters are out of bounds, or steps MUST be performed; if any parameters are out of bounds, or
cannot be applied, processing MUST stop and an error MUST be cannot be applied, processing MUST stop and an error MUST be
returned. returned.
o If this m= section is new, begin gathering candidates for it, as o If this m= section is new, begin gathering candidates for it, as
defined in [RFC5245], Section 4.1.1, unless it has been marked as defined in [RFC5245], Section 4.1.1, unless it is definitively
bundle-only. being bundled (either this is an offer and the m= section is
marked bundle-only, or it is an answer and the m= section is
bundled into into another m= section.)
o Or, if the ICE ufrag and password values have changed, and it has o Or, if the ICE ufrag and password values have changed, trigger the
not been marked as bundle-only, trigger the ICE agent to start an ICE agent to start an ICE restart, and begin gathering new
ICE restart, and begin gathering new candidates for the m= section candidates for the m= section as described in [RFC5245],
as described in [RFC5245], Section 9.1.1.1. If this description Section 9.1.1.1. If this description is an answer, also start
is an answer, also start checks on that media section as defined checks on that media section as defined in [RFC5245],
in [RFC5245], Section 9.3.1.1. Section 9.3.1.1.
o If the m= section proto value indicates use of RTP: o 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= section,
(which will only happen when applying an offer), find one and find one and associate it with this m= section according to the
associate it with this m= section according to the following following steps. Note that this situation will only occur when
steps: applying an offer.
+ Find the RtpTransceiver that corresponds to this m= section, + Find the RtpTransceiver that corresponds to this m= section,
using the mapping between transceivers and m= section using the mapping between transceivers and m= section
indices established when creating the offer. indices established when creating the offer.
+ Set the value of this RtpTransceiver's mid property to the + 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. If RTCP mux is not indicated, but was Section 5.1.3.
previously negotiated, i.e., the RTCP ICE component no longer
exists, this MUST result in an error.
* 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 section 6 of between the extension ID and URI, as described in [RFC5285],
[RFC5285]. If any indicated RTP header extension is not Section 6.
supported, this MUST result in an error.
* 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
[I-D.ietf-mmusic-sdp-bundle-negotiation], Section 14. [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. If any indicated media format is not [RFC3264], Section 6.1. In addition, prepare to demux RTP
supported, this MUST result in an error. streams intended for this m= section based on the media formats
supported by this m= section, as described in
[I-D.ietf-mmusic-sdp-bundle-negotiation], Section 10.2.
* 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. If any type, as described in [RFC4588], Sections 8.6 and 8.7.
referenced primary payload types are not present, this MUST
result in an error.
* If the directional attribute is of type "sendrecv" or * If the directional attribute is of type "sendrecv" or
"recvonly", enable receipt and decoding of media. "recvonly", 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 the Section 5.10 section below. follow the processing defined in Section 5.10 below.
5.9. Applying a Remote Description 5.9. 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 canTrickle
property to be true. Otherwise, set this property to false. 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 o For any specified "CT" bandwidth value, set this as the limit for
the maximum total bitrate for all m= sections, as specified in the maximum total bitrate for all m= sections, as specified in
Section 5.8 of [RFC4566]. Within this overall limit, the [RFC4566], Section 5.8. Within this overall limit, the
implementation can dynamically decide how to best allocate 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 o 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 o Any "AS" bandwidth value MUST be ignored, as the meaning of this
construct at the session level is not well defined. 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 PeerConnection state is "have-local-offer", and the ICE o If the ICE ufrag or password changed from the previous remote
ufrag or password changed from the previous remote description, description: [RFC5245].
then an ICE restart is needed, as described in Section 9.1.1.1 of
[RFC5245]. If the description is of type "offer", note that an
ICE restart is needed. If the description is of type "answer" or
"pranswer" and the current local description is also an ICE
restart, then signal the ICE agent to begin checks as described in
Section 9.3.1.1 of [RFC5245]. An answerer MUST change the ufrag
and password in an answer if and only if ICE is restarting, as
described in Section 9.2.1.1 of [RFC5245].
o If the PeerConnection state is "have-remote-pranswer", and the ICE * If the description is of type "offer", note that an ICE restart
ufrag or password changed from the previous provisional answer, is needed, as described in [RFC5245], Section 9.1.1.1 .
then signal the ICE agent to discard any previous ICE check list
state for the m= section and begin checks as if this were the * If the description is of type "answer" or "pranswer", then
first answer. However, such an answer MAY only change the ICE check to see if the current local description is an ICE
ufrag or password if the local offer is starting or restarting ICE restart, and if not, generate an error. It the PeerConnection
for the m= section. state is "have-remote-pranswer", and the ICE ufrag or password
changed from the previous provisional answer, then signal the
ICE agent to discard any previous ICE check list state for the
m= section. Finally, signal the ICE agent to begin checks as
described in [RFC5245], Section 9.3.1.1.
o If the current local description indicates an ICE restart, and
either the ICE ufrag or password has not changed from the previous
remote description, as prescribed by [RFC5245], Section 9.2.1.1,
generate an error.
o Configure the ICE components associated with this media section to o 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 o Pair any supplied ICE candidates with any gathered local
candidates, as described in Section 5.7 of [RFC5245] and start candidates, as described in [RFC5245], Section 5.7, 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- o 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 [I-D.ietf-ice-trickle],
Section 11. Section 11.
o If the m= section proto value indicates use of RTP: o 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 dissociate 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
skipping to change at page 66, line 39 skipping to change at page 68, line 19
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. error. Note that RTX payload types may refer to primary
payload types which are not supported by the local media
implementation, in which case, the RTX payload type MUST also
be 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,
prepare to demux RTP streams intended for this m= section using
that SSRC, as described in
[I-D.ietf-mmusic-sdp-bundle-negotiation], Section 10.2.
* 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" 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 using this formula: value 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 50 is based on 50 packets per second, the 40 is based on an
estimate of total header size, the 1000 changes the unit from estimate of total header size, the 1000 changes the unit from
kbps to bps (as required by TIAS), and the 0.95 is to allocate kbps to bps (as required by TIAS), and the 0.95 is to allocate
5% to RTCP. "TIAS" is used in preference to "AS" because it 5% to RTCP. "TIAS" is used in preference to "AS" because it
provides more accurate control of bandwidth. 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, enable DTX for all + For each specified "CN" media format, configure silence
supported media formats with the same clockrate, as suppression for all supported media formats with the same
described in [RFC3389], Section 5, except for formats that clockrate, as described in [RFC3389], Section 5, except for
have their own internal DTX mechanisms. DTX for such formats that have their own internal silence suppression
formats (e.g., Opus) is controlled via fmtp parameters, as mechanisms. Silence suppression for such formats (e.g.,
discussed in Section 5.2.3.2. Opus) is controlled via fmtp parameters, as discussed in
Section 5.2.3.2.
+ For each specified "telephone-event" media format, enable + For each specified "telephone-event" media format, enable
DTMF transmission for all supported media formats with the DTMF transmission for all supported media formats with the
same clockrate, as described in [RFC4733], Section 2.5.1.2. same clockrate, as described in [RFC4733], Section 2.5.1.2.
If the application attempts to transmit DTMF when using a If the application attempts to transmit DTMF when using a
media format that does not have a corresponding telephone- media format that does not have a corresponding telephone-
event format, this MUST result in an error. event format, this MUST result in an error.
+ For any specified "ptime" value, configure the available + For any specified "ptime" value, configure the available
media formats to use the specified packet size. If the media formats to use the specified packet size when sending.
specified size is not supported for a media format, use the If the specified size is not supported for a media format,
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 the Section 5.10 section below. follow the processing defined in Section 5.10 below.
5.10. Applying an Answer 5.10. 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 o If the m= section has been rejected (i.e. port is set to zero in
the answer), stop any reception or transmission of media for this the answer), stop any reception or transmission of media for this
section, and, unless a non-rejected m= section is bundled with section, and, unless a non-rejected m= section is bundled with
this m= section, discard any associated ICE components, as this m= section, discard any associated ICE components, as
described in Section 9.2.1.3 of [RFC5245]. described in [RFC5245], Section 9.2.1.3.
o If the remote DTLS fingerprint has been changed or the dtls-id has o If the remote DTLS fingerprint has been changed or the tls-id has
changed, tear down the DTLS connection. This includes the case changed, tear down the DTLS connection. This includes the case
when the PeerConnection state is "have-remote-pranswer". If a when the PeerConnection state is "have-remote-pranswer". If a
DTLS connection needs to be torn down but the answer does not DTLS connection needs to be torn down but the answer does not
indicate an ICE restart or, in the case of "have-remote-pranswer", indicate an ICE restart or, in the case of "have-remote-pranswer",
new ICE credentials, an error MUST be generated. If an ICE new ICE credentials, an error MUST be generated. If an ICE
restart is performed without a change in dtls-id or fingerprint, restart is performed without a change in tls-id or fingerprint,
then the same DTLS connection is continued over the new ICE then the same DTLS connection is continued over the new ICE
channel. channel.
o If no valid DTLS connection exists, prepare to start a DTLS o 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: o If the m= section proto value indicates use of RTP:
* If the m= section references any media formats, RTP header * If the m= section references RTCP feedback mechanisms that were
extensions, or RTCP feedback mechanisms that were not present not present in the corresponding m= section in the offer, this
in the corresponding m= section in the offer, this indicates a indicates a negotiation problem and MUST result in an error.
negotiation problem and MUST result in an error. However, new media formats and new RTP header extension values
are permitted in the answer, as described in [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 the
RTCP transmission for this m= section to use reduced-size RTCP, RTCP transmission for this m= section to use reduced-size RTCP,
as specified in [RFC5506]. as specified in [RFC5506].
* If the directional attribute in the answer is of type * If the directional attribute in the answer indicates that the
"sendrecv" or "sendonly", choose the media format to send as JSEP implementation should be sending media ("sendonly" for
the most preferred media format from the remote description local answers, "recvonly" for remote answers, or "sendrecv" for
that is also present in the answer, as described in [RFC3264], either type of answer), choose the media format to send as the
Sections 6.1 and 7, and start transmitting RTP media once the most preferred media format from the remote description that is
underlying transport layers have been established. If an SSRC also locally supported, as discussed in [RFC3264], Sections 6.1
has not already been chosen for this outgoing RTP stream, and 7, and start transmitting RTP media using that format once
the underlying transport layers have been established. If an
SSRC has not already been chosen for this outgoing RTP stream,
choose a random one. If media is already being transmitted, choose a random one. If media is already being transmitted,
the same SSRC SHOULD be used unless the clockrate of the new the same SSRC SHOULD be used unless the clockrate of the new
codec is different, in which case a new SSRC MUST be chosen, as codec is different, in which case a new SSRC MUST be chosen, as
specified in [RFC7160], Section 3.1. specified in [RFC7160], Section 3.1.
* 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 RID header extension has been
negotiated, and RID values are specified, include the RID negotiated, and RID values are specified, include the RID
header extension in the outgoing RTP streams, as indicated in header extension in the outgoing RTP streams, as indicated in
[I-D.ietf-mmusic-rid], Section 4. [I-D.ietf-mmusic-rid], Section 4.
* If the m= section is of type audio, and silence suppression was
configured for the send media format as a result of processing
the remote description, and is also enabled for that format in
the local description, use silence suppression for outgoing
media, in accordance with the guidance in Section 5.2.3.2. If
these conditions are not met, silence 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 number of Source RTP
Streams as specified in [I-D.ietf-mmusic-sdp-simulcast], Streams as specified in [I-D.ietf-mmusic-sdp-simulcast],
Section 6.2.2. 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 random SSRC for each Source RTP
Stream, and start or continue transmitting RTX/FEC packets as Stream, and start or continue transmitting RTX/FEC packets as
needed. needed.
skipping to change at page 69, line 47 skipping to change at page 71, line 49
discussed in [I-D.ietf-rtcweb-fec], Section 3.2. Note that discussed in [I-D.ietf-rtcweb-fec], Section 3.2. Note that
unlike RTX or FEC media formats, the "red" format is unlike RTX or FEC media formats, the "red" format is
transmitted on the Source RTP Stream, not the Redundancy RTP transmitted on the 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 the rules and recommendations from [RFC8108] Section 5.4.1, to
[I-D.ietf-avtcore-rtp-multi-stream], Section 5.4.1 to select select which SSRC to use.
which SSRC to use.
* If the directional attribute is of type "recvonly" or * If the directional attribute in the answer indicates that the
"inactive", stop transmitting all RTP media, but continue JSEP implementation should not be sending media ("recvonly" for
sending RTCP, as described in [RFC3264], Section 5.1. local answers, "sendonly" for remote answers, or "inactive" for
either type of answer) stop transmitting all RTP media, but
continue sending RTCP, as described in [RFC3264], Section 5.1.
o If the m= section proto value indicates use of SCTP: o 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 a SCTP
association over the associated ICE component and DTLS association over the associated ICE component and DTLS
skipping to change at page 70, line 35 skipping to change at page 72, line 37
ICE components in each bundle, and begin muxing these m= sections ICE components in each bundle, and begin muxing these m= sections
accordingly, as described in accordingly, as described in
[I-D.ietf-mmusic-sdp-bundle-negotiation], Section 8.2. [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]. When section is defined in [I-D.ietf-mmusic-sdp-bundle-negotiation],
not bundling, the proper m= section is clear from the ICE component Section 10.2. When not bundling, the proper m= section is clear from
over which the RTP/RTCP is received. the ICE component over which the RTP/RTCP is received.
Once the proper m= section(s) are known, RTP/RTCP is delivered to the Once the proper m= section(s) are known, RTP/RTCP is delivered to the
RtpTransceiver(s) associated with the m= section(s) and further RtpTransceiver(s) associated with the m= section(s) and further
processing of the RTP/RTCP is done at the RtpTransceiver level. This processing of the RTP/RTCP is done at the RtpTransceiver level. This
includes using RID [I-D.ietf-mmusic-rid] to distinguish between includes using RID [I-D.ietf-mmusic-rid] to distinguish between
multiple Encoded Streams, as well as determine which Source RTP multiple Encoded Streams, as well as determine which Source RTP
stream should be repaired by a given Redundancy RTP stream. 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 format in 72 columns, some of the lines in SDP have been split into
multiple lines, where leading whitespace indicates that a line is a multiple lines, where leading whitespace indicates that a line is a
continuation of the previous line. In addition, some blank lines continuation of the previous line. In addition, some blank lines
have been added to improve readability but are not valid in SDP. have been added to improve readability but are not valid in SDP.
More examples of SDP for WebRTC call flows can be found in More examples of SDP for WebRTC call flows, including examples with
[I-D.nandakumar-rtcweb-sdp]. IPv6 addresses, can be found in [I-D.ietf-rtcweb-sdp].
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 audio
/ video call between two JSEP endpoints without using trickle ICE. / video call between two JSEP endpoints without using trickle ICE.
The example in the following section provides a more detailed example The example in the following section provides a more detailed example
of what could happen in a JSEP session. 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 Alice's JS to Bob's JS are assumed Bob's endpoint. The messages from the JavaScript application in
to flow over some signaling protocol via a web server. The JS on Alice's browser to the JavaScript in Bob's browser, abbreviated as
both Alice's side and Bob's side waits for all candidates before AliceJS and BobJS respectively, are assumed to flow over some
sending the offer or answer, so the offers and answers are complete; signaling protocol via a web server. The JavaScript on both Alice's
trickle ICE is not used. Both Alice and Bob are using the default side and Bob's side waits for all candidates before sending the offer
bundle policy of "balanced", and the default RTCP mux policy of or answer, so the offers and answers are complete; trickle ICE is not
"require". used. The user agents (JSEP implementations) in Alice and Bob's
browsers, abbreviated as AliceUA and BobUA respectively, are using
the default bundle policy of "balanced", and the default 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
skipping to change at page 73, line 19 skipping to change at page 75, line 19
m=audio 10100 UDP/TLS/RTP/SAVPF 96 0 8 97 98 m=audio 10100 UDP/TLS/RTP/SAVPF 96 0 8 97 98
c=IN IP4 203.0.113.100 c=IN IP4 203.0.113.100
a=mid:a1 a=mid:a1
a=sendrecv a=sendrecv
a=rtpmap:96 opus/48000/2 a=rtpmap:96 opus/48000/2
a=rtpmap:0 PCMU/8000 a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000 a=rtpmap:8 PCMA/8000
a=rtpmap:97 telephone-event/8000 a=rtpmap:97 telephone-event/8000
a=rtpmap:98 telephone-event/48000 a=rtpmap:98 telephone-event/48000
a=fmtp:97 0-15
a=fmtp:98 0-15
a=maxptime:120 a=maxptime:120
a=extmap:1 urn:ietf:params:rtp-hdrext:sdes:mid a=extmap:1 urn:ietf:params:rtp-hdrext:sdes:mid
a=extmap:2 urn:ietf:params:rtp-hdrext:ssrc-audio-level a=extmap:2 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=msid:47017fee-b6c1-4162-929c-a25110252400 a=msid:47017fee-b6c1-4162-929c-a25110252400
f83006c5-a0ff-4e0a-9ed9-d3e6747be7d9 f83006c5-a0ff-4e0a-9ed9-d3e6747be7d9
a=ice-ufrag:ETEn a=ice-ufrag:ETEn
a=ice-pwd:OtSK0WpNtpUjkY4+86js7ZQl a=ice-pwd:OtSK0WpNtpUjkY4+86js7ZQl
a=fingerprint:sha-256 a=fingerprint:sha-256
19:E2:1C:3B:4B:9F:81:E6:B8:5C:F4:A5:A8:D8:73:04: 19:E2:1C:3B:4B:9F:81:E6:B8:5C:F4:A5:A8:D8:73:04:
BB:05:2F:70:9F:04:A9:0E:05:E9:26:33:E8:70:88:A2 BB:05:2F:70:9F:04:A9:0E:05:E9:26:33:E8:70:88:A2
a=setup:actpass a=setup:actpass
a=dtls-id:1 a=tls-id:1
a=rtcp:10101 IN IP4 203.0.113.100 a=rtcp:10101 IN IP4 203.0.113.100
a=rtcp-mux a=rtcp-mux
a=rtcp-rsize a=rtcp-rsize
a=candidate:1 1 udp 2113929471 203.0.113.100 10100 typ host a=candidate:1 1 udp 2113929471 203.0.113.100 10100 typ host
a=candidate:1 2 udp 2113929470 203.0.113.100 10101 typ host a=candidate:1 2 udp 2113929470 203.0.113.100 10101 typ host
a=end-of-candidates a=end-of-candidates
m=video 10102 UDP/TLS/RTP/SAVPF 100 101 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 rtx/90000 a=rtpmap:101 H264/90000
a=fmtp:101 apt=100 a=fmtp:101 packetization-mode=1;profile-level-id=42e01f
a=rtpmap:102 rtx/90000
a=fmtp:102 apt=100
=rtpmap:103 rtx/90000
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=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
f30bdb4a-5db8-49b5-bcdc-e0c9a23172e0 f30bdb4a-5db8-49b5-bcdc-e0c9a23172e0
a=ice-ufrag:BGKk a=ice-ufrag:BGKk
a=ice-pwd:mqyWsAjvtKwTGnvhPztQ9mIf a=ice-pwd:mqyWsAjvtKwTGnvhPztQ9mIf
a=fingerprint:sha-256 a=fingerprint:sha-256
19:E2:1C:3B:4B:9F:81:E6:B8:5C:F4:A5:A8:D8:73:04: 19:E2:1C:3B:4B:9F:81:E6:B8:5C:F4:A5:A8:D8:73:04:
BB:05:2F:70:9F:04:A9:0E:05:E9:26:33:E8:70:88:A2 BB:05:2F:70:9F:04:A9:0E:05:E9:26:33:E8:70:88:A2
a=setup:actpass a=setup:actpass
a=dtls-id:1 a=tls-id:1
a=rtcp:10103 IN IP4 203.0.113.100 a=rtcp:10103 IN IP4 203.0.113.100
a=rtcp-mux a=rtcp-mux
a=rtcp-rsize a=rtcp-rsize
a=candidate:1 1 udp 2113929471 203.0.113.100 10102 typ host a=candidate:1 1 udp 2113929471 203.0.113.100 10102 typ host
a=candidate:1 2 udp 2113929470 203.0.113.100 10103 typ host a=candidate:1 2 udp 2113929470 203.0.113.100 10103 typ host
a=end-of-candidates a=end-of-candidates
The SDP for |answer-A1| looks like: The SDP for |answer-A1| looks like:
v=0 v=0
skipping to change at page 75, line 22 skipping to change at page 76, line 43
m=audio 10200 UDP/TLS/RTP/SAVPF 96 0 8 97 98 m=audio 10200 UDP/TLS/RTP/SAVPF 96 0 8 97 98
c=IN IP4 203.0.113.200 c=IN IP4 203.0.113.200
a=mid:a1 a=mid:a1
a=sendrecv a=sendrecv
a=rtpmap:96 opus/48000/2 a=rtpmap:96 opus/48000/2
a=rtpmap:0 PCMU/8000 a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000 a=rtpmap:8 PCMA/8000
a=rtpmap:97 telephone-event/8000 a=rtpmap:97 telephone-event/8000
a=rtpmap:98 telephone-event/48000 a=rtpmap:98 telephone-event/48000
a=fmtp:97 0-15
a=fmtp:98 0-15
a=maxptime:120 a=maxptime:120
a=extmap:1 urn:ietf:params:rtp-hdrext:sdes:mid a=extmap:1 urn:ietf:params:rtp-hdrext:sdes:mid
a=extmap:2 urn:ietf:params:rtp-hdrext:ssrc-audio-level a=extmap:2 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=msid:61317484-2ed4-49d7-9eb7-1414322a7aae a=msid:61317484-2ed4-49d7-9eb7-1414322a7aae
5a7b57b8-f043-4bd1-a45d-09d4dfa31226 5a7b57b8-f043-4bd1-a45d-09d4dfa31226
a=ice-ufrag:6sFv a=ice-ufrag:6sFv
a=ice-pwd:cOTZKZNVlO9RSGsEGM63JXT2 a=ice-pwd:cOTZKZNVlO9RSGsEGM63JXT2
a=fingerprint:sha-256 a=fingerprint:sha-256
6B:8B:F0:65:5F:78:E2:51:3B:AC:6F:F3:3F:46:1B:35: 6B:8B:F0:65:5F:78:E2:51:3B:AC:6F:F3:3F:46:1B:35:
DC:B8:5F:64:1A:24:C2:43:F0:A1:58:D0:A1:2C:19:08 DC:B8:5F:64:1A:24:C2:43:F0:A1:58:D0:A1:2C:19:08
a=setup:active a=setup:active
a=dtls-id:1 a=tls-id:1
a=rtcp-mux a=rtcp-mux
a=rtcp-rsize a=rtcp-rsize
a=candidate:1 1 udp 2113929471 203.0.113.200 10200 typ host a=candidate:1 1 udp 2113929471 203.0.113.200 10200 typ host
a=end-of-candidates a=end-of-candidates
m=video 10200 UDP/TLS/RTP/SAVPF 100 101 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 rtx/90000 a=rtpmap:101 H264/90000
a=fmtp:101 apt=100 a=fmtp:101 packetization-mode=1;profile-level-id=42e01f
a=rtpmap:102 rtx/90000
a=fmtp:102 apt=100
=rtpmap:103 rtx/90000
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=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
4ea4d4a1-2fda-4511-a9cc-1b32c2e59552 4ea4d4a1-2fda-4511-a9cc-1b32c2e59552
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
skipping to change at page 79, line 21 skipping to change at page 80, line 21
m=audio 9 UDP/TLS/RTP/SAVPF 96 0 8 97 98 m=audio 9 UDP/TLS/RTP/SAVPF 96 0 8 97 98
c=IN IP4 0.0.0.0 c=IN IP4 0.0.0.0
a=mid:a1 a=mid:a1
a=sendrecv a=sendrecv
a=rtpmap:96 opus/48000/2 a=rtpmap:96 opus/48000/2
a=rtpmap:0 PCMU/8000 a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000 a=rtpmap:8 PCMA/8000
a=rtpmap:97 telephone-event/8000 a=rtpmap:97 telephone-event/8000
a=rtpmap:98 telephone-event/48000 a=rtpmap:98 telephone-event/48000
a=fmtp:97 0-15
a=fmtp:98 0-15
a=maxptime:120 a=maxptime:120
a=extmap:1 urn:ietf:params:rtp-hdrext:sdes:mid a=extmap:1 urn:ietf:params:rtp-hdrext:sdes:mid
a=extmap:2 urn:ietf:params:rtp-hdrext:ssrc-audio-level a=extmap:2 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=msid:57017fee-b6c1-4162-929c-a25110252400 a=msid:57017fee-b6c1-4162-929c-a25110252400
e83006c5-a0ff-4e0a-9ed9-d3e6747be7d9 e83006c5-a0ff-4e0a-9ed9-d3e6747be7d9
a=ice-ufrag:ATEn a=ice-ufrag:ATEn
a=ice-pwd:AtSK0WpNtpUjkY4+86js7ZQl a=ice-pwd:AtSK0WpNtpUjkY4+86js7ZQl
a=fingerprint:sha-256 a=fingerprint:sha-256
29:E2:1C:3B:4B:9F:81:E6:B8:5C:F4:A5:A8:D8:73:04: 29:E2:1C:3B:4B:9F:81:E6:B8:5C:F4:A5:A8:D8:73:04:
BB:05:2F:70:9F:04:A9:0E:05:E9:26:33:E8:70:88:A2 BB:05:2F:70:9F:04:A9:0E:05:E9:26:33:E8:70:88:A2
a=setup:actpass a=setup:actpass
a=dtls-id:1 a=tls-id:1
a=rtcp-mux a=rtcp-mux
a=rtcp-mux-only a=rtcp-mux-only
a=rtcp-rsize a=rtcp-rsize
m=application 0 UDP/DTLS/SCTP webrtc-datachannel m=application 0 UDP/DTLS/SCTP webrtc-datachannel
c=IN IP4 0.0.0.0 c=IN IP4 0.0.0.0
a=mid:d1 a=mid:d1
a=sctp-port:5000 a=sctp-port:5000
a=max-message-size:65536 a=max-message-size:65536
a=bundle-only a=bundle-only
skipping to change at page 81, line 21 skipping to change at page 82, line 21
m=audio 9 UDP/TLS/RTP/SAVPF 96 0 8 97 98 m=audio 9 UDP/TLS/RTP/SAVPF 96 0 8 97 98
c=IN IP4 0.0.0.0 c=IN IP4 0.0.0.0
a=mid:a1 a=mid:a1
a=sendrecv a=sendrecv
a=rtpmap:96 opus/48000/2 a=rtpmap:96 opus/48000/2
a=rtpmap:0 PCMU/8000 a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000 a=rtpmap:8 PCMA/8000
a=rtpmap:97 telephone-event/8000 a=rtpmap:97 telephone-event/8000
a=rtpmap:98 telephone-event/48000 a=rtpmap:98 telephone-event/48000
a=fmtp:97 0-15
a=fmtp:98 0-15
a=maxptime:120 a=maxptime:120
a=extmap:1 urn:ietf:params:rtp-hdrext:sdes:mid a=extmap:1 urn:ietf:params:rtp-hdrext:sdes:mid
a=extmap:2 urn:ietf:params:rtp-hdrext:ssrc-audio-level a=extmap:2 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=msid:71317484-2ed4-49d7-9eb7-1414322a7aae a=msid:71317484-2ed4-49d7-9eb7-1414322a7aae
6a7b57b8-f043-4bd1-a45d-09d4dfa31226 6a7b57b8-f043-4bd1-a45d-09d4dfa31226
a=ice-ufrag:7sFv a=ice-ufrag:7sFv
a=ice-pwd:dOTZKZNVlO9RSGsEGM63JXT2 a=ice-pwd:dOTZKZNVlO9RSGsEGM63JXT2
a=fingerprint:sha-256 a=fingerprint:sha-256
7B:8B:F0:65:5F:78:E2:51:3B:AC:6F:F3:3F:46:1B:35: 7B:8B:F0:65:5F:78:E2:51:3B:AC:6F:F3:3F:46:1B:35:
DC:B8:5F:64:1A:24:C2:43:F0:A1:58:D0:A1:2C:19:08 DC:B8:5F:64:1A:24:C2:43:F0:A1:58:D0:A1:2C:19:08
a=setup:active a=setup:active
a=dtls-id:1 a=tls-id:1
a=rtcp-mux a=rtcp-mux
a=rtcp-mux-only a=rtcp-mux-only
a=rtcp-rsize a=rtcp-rsize
m=application 9 UDP/DTLS/SCTP webrtc-datachannel m=application 9 UDP/DTLS/SCTP webrtc-datachannel
c=IN IP4 0.0.0.0 c=IN IP4 0.0.0.0
a=mid:d1 a=mid:d1
a=sctp-port:5000 a=sctp-port:5000
a=max-message-size:65536 a=max-message-size:65536
skipping to change at page 82, line 44 skipping to change at page 84, line 7
m=audio 12200 UDP/TLS/RTP/SAVPF 96 0 8 97 98 m=audio 12200 UDP/TLS/RTP/SAVPF 96 0 8 97 98
c=IN IP4 192.0.2.200 c=IN IP4 192.0.2.200
a=mid:a1 a=mid:a1
a=sendrecv a=sendrecv
a=rtpmap:96 opus/48000/2 a=rtpmap:96 opus/48000/2
a=rtpmap:0 PCMU/8000 a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000 a=rtpmap:8 PCMA/8000
a=rtpmap:97 telephone-event/8000 a=rtpmap:97 telephone-event/8000
a=rtpmap:98 telephone-event/48000 a=rtpmap:98 telephone-event/48000
a=fmtp:97 0-15
a=fmtp:98 0-15
a=maxptime:120 a=maxptime:120
a=extmap:1 urn:ietf:params:rtp-hdrext:sdes:mid a=extmap:1 urn:ietf:params:rtp-hdrext:sdes:mid
a=extmap:2 urn:ietf:params:rtp-hdrext:ssrc-audio-level a=extmap:2 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=msid:71317484-2ed4-49d7-9eb7-1414322a7aae a=msid:71317484-2ed4-49d7-9eb7-1414322a7aae
6a7b57b8-f043-4bd1-a45d-09d4dfa31226 6a7b57b8-f043-4bd1-a45d-09d4dfa31226
a=ice-ufrag:7sFv a=ice-ufrag:7sFv
a=ice-pwd:dOTZKZNVlO9RSGsEGM63JXT2 a=ice-pwd:dOTZKZNVlO9RSGsEGM63JXT2
a=fingerprint:sha-256 a=fingerprint:sha-256
7B:8B:F0:65:5F:78:E2:51:3B:AC:6F:F3:3F:46:1B:35: 7B:8B:F0:65:5F:78:E2:51:3B:AC:6F:F3:3F:46:1B:35:
DC:B8:5F:64:1A:24:C2:43:F0:A1:58:D0:A1:2C:19:08 DC:B8:5F:64:1A:24:C2:43:F0:A1:58:D0:A1:2C:19:08
a=setup:actpass a=setup:actpass
a=dtls-id:1 a=tls-id:1
a=rtcp-mux a=rtcp-mux
a=rtcp-mux-only a=rtcp-mux-only
a=rtcp-rsize a=rtcp-rsize
a=candidate:1 1 udp 2113929471 203.0.113.200 10200 typ host a=candidate:1 1 udp 2113929471 203.0.113.200 10200 typ host
a=candidate:1 1 udp 1845494015 198.51.100.200 11200 typ srflx a=candidate:1 1 udp 1845494015 198.51.100.200 11200 typ srflx
raddr 203.0.113.200 rport 10200 raddr 203.0.113.200 rport 10200
a=candidate:1 1 udp 255 192.0.2.200 12200 typ relay a=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
a=end-of-candidates a=end-of-candidates
m=application 12200 UDP/DTLS/SCTP webrtc-datachannel m=application 12200 UDP/DTLS/SCTP webrtc-datachannel
c=IN IP4 192.0.2.200 c=IN IP4 192.0.2.200
a=mid:d1 a=mid:d1
a=sctp-port:5000 a=sctp-port:5000
a=max-message-size:65536 a=max-message-size:65536
m=video 12200 UDP/TLS/RTP/SAVPF 100 101 102 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 rtx/90000 a=rtpmap:101 H264/90000
a=fmtp:101 apt=100 a=fmtp:101 packetization-mode=1;profile-level-id=42e01f
a=rtpmap:102 flexfec/90000 a=rtpmap:102 rtx/90000
a=fmtp:102 apt=100
=rtpmap:103 rtx/90000
a=fmtp:103 apt=101
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=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
5ea4d4a1-2fda-4511-a9cc-1b32c2e59552 5ea4d4a1-2fda-4511-a9cc-1b32c2e59552
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 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 rtx/90000 a=rtpmap:101 H264/90000
a=fmtp:101 apt=100 a=fmtp:101 packetization-mode=1;profile-level-id=42e01f
a=rtpmap:102 flexfec/90000 a=rtpmap:102 rtx/90000
a=fmtp:102 apt=100
=rtpmap:103 rtx/90000
a=fmtp:103 apt=101
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=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
6ea4d4a1-2fda-4511-a9cc-1b32c2e59552 6ea4d4a1-2fda-4511-a9cc-1b32c2e59552
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
skipping to change at page 84, line 38 skipping to change at page 86, line 10
m=audio 12100 UDP/TLS/RTP/SAVPF 96 0 8 97 98 m=audio 12100 UDP/TLS/RTP/SAVPF 96 0 8 97 98
c=IN IP4 192.0.2.100 c=IN IP4 192.0.2.100
a=mid:a1 a=mid:a1
a=sendrecv a=sendrecv
a=rtpmap:96 opus/48000/2 a=rtpmap:96 opus/48000/2
a=rtpmap:0 PCMU/8000 a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000 a=rtpmap:8 PCMA/8000
a=rtpmap:97 telephone-event/8000 a=rtpmap:97 telephone-event/8000
a=rtpmap:98 telephone-event/48000 a=rtpmap:98 telephone-event/48000
a=fmtp:97 0-15
a=fmtp:98 0-15
a=maxptime:120 a=maxptime:120
a=extmap:1 urn:ietf:params:rtp-hdrext:sdes:mid a=extmap:1 urn:ietf:params:rtp-hdrext:sdes:mid
a=extmap:2 urn:ietf:params:rtp-hdrext:ssrc-audio-level a=extmap:2 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=msid:57017fee-b6c1-4162-929c-a25110252400 a=msid:57017fee-b6c1-4162-929c-a25110252400
e83006c5-a0ff-4e0a-9ed9-d3e6747be7d9 e83006c5-a0ff-4e0a-9ed9-d3e6747be7d9
a=ice-ufrag:ATEn a=ice-ufrag:ATEn
a=ice-pwd:AtSK0WpNtpUjkY4+86js7ZQl a=ice-pwd:AtSK0WpNtpUjkY4+86js7ZQl
a=fingerprint:sha-256 a=fingerprint:sha-256
29:E2:1C:3B:4B:9F:81:E6:B8:5C:F4:A5:A8:D8:73:04: 29:E2:1C:3B:4B:9F:81:E6:B8:5C:F4:A5:A8:D8:73:04:
BB:05:2F:70:9F:04:A9:0E:05:E9:26:33:E8:70:88:A2 BB:05:2F:70:9F:04:A9:0E:05:E9:26:33:E8:70:88:A2
a=setup:passive a=setup:passive
a=dtls-id:1 a=tls-id:1
a=rtcp-mux a=rtcp-mux
a=rtcp-mux-only a=rtcp-mux-only
a=rtcp-rsize a=rtcp-rsize
a=candidate:1 1 udp 2113929471 203.0.113.100 10100 typ host a=candidate:1 1 udp 2113929471 203.0.113.100 10100 typ host
a=candidate:1 1 udp 1845494015 198.51.100.100 11100 typ srflx a=candidate:1 1 udp 1845494015 198.51.100.100 11100 typ srflx
raddr 203.0.113.100 rport 10100 raddr 203.0.113.100 rport 10100
a=candidate:1 1 udp 255 192.0.2.100 12100 typ relay a=candidate:1 1 udp 255 192.0.2.100 12100 typ relay
raddr 198.51.100.100 rport 11100 raddr 198.51.100.100 rport 11100
a=end-of-candidates a=end-of-candidates
m=application 12100 UDP/DTLS/SCTP webrtc-datachannel m=application 12100 UDP/DTLS/SCTP webrtc-datachannel
c=IN IP4 192.0.2.100 c=IN IP4 192.0.2.100
a=mid:d1 a=mid:d1
a=sctp-port:5000 a=sctp-port:5000
a=max-message-size:65536 a=max-message-size:65536
m=video 12100 UDP/TLS/RTP/SAVPF 100 101 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 rtx/90000 a=rtpmap:101 H264/90000
a=fmtp:101 apt=100 a=fmtp:101 packetization-mode=1;profile-level-id=42e01f
a=rtpmap:102 rtx/90000
a=fmtp:102 apt=100
=rtpmap:103 rtx/90000
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=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 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 rtx/90000 a=rtpmap:101 H264/90000
a=fmtp:101 apt=100 a=fmtp:101 packetization-mode=1;profile-level-id=42e01f
a=rtpmap:102 rtx/90000
a=fmtp:102 apt=100
=rtpmap:103 rtx/90000
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=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 4.1.8.1. 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 as
skipping to change at page 88, line 22 skipping to change at page 89, line 43
m=audio 9 UDP/TLS/RTP/SAVPF 96 0 8 97 98 m=audio 9 UDP/TLS/RTP/SAVPF 96 0 8 97 98
c=IN IP4 0.0.0.0 c=IN IP4 0.0.0.0
a=mid:a1 a=mid:a1
a=sendrecv a=sendrecv
a=rtpmap:96 opus/48000/2 a=rtpmap:96 opus/48000/2
a=rtpmap:0 PCMU/8000 a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000 a=rtpmap:8 PCMA/8000
a=rtpmap:97 telephone-event/8000 a=rtpmap:97 telephone-event/8000
a=rtpmap:98 telephone-event/48000 a=rtpmap:98 telephone-event/48000
a=fmtp:97 0-15
a=fmtp:98 0-15
a=maxptime:120 a=maxptime:120
a=extmap:1 urn:ietf:params:rtp-hdrext:sdes:mid a=extmap:1 urn:ietf:params:rtp-hdrext:sdes:mid
a=extmap:2 urn:ietf:params:rtp-hdrext:ssrc-audio-level a=extmap:2 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=msid:bbce3ba6-abfc-ac63-d00a-e15b286f8fce a=msid:bbce3ba6-abfc-ac63-d00a-e15b286f8fce
e80098db-7159-3c06-229a-00df2a9b3dbc e80098db-7159-3c06-229a-00df2a9b3dbc
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=dtls-id:1 a=tls-id:1
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 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 rtx/90000 a=rtpmap:101 H264/90000
a=fmtp:101 apt=100 a=fmtp:101 packetization-mode=1;profile-level-id=42e01f
a=rtpmap:102 rtx/90000
a=fmtp:102 apt=100
=rtpmap:103 rtx/90000
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=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
ac701365-eb06-42df-cc93-7f22bc308789 ac701365-eb06-42df-cc93-7f22bc308789
a=bundle-only a=bundle-only
|offer-C1-candidate-1| looks like: |offer-C1-candidate-1| looks like:
ufrag 4ZcD ufrag 4ZcD
index 0 index 0
mid a1 mid a1
attr candidate:1 1 udp 255 192.0.2.100 12100 typ relay attr candidate:1 1 udp 255 192.0.2.100 12100 typ relay
raddr 0.0.0.0 rport 0 raddr 0.0.0.0 rport 0
The SDP for |answer-C1| looks like: The SDP for |answer-C1| looks like:
skipping to change at page 90, line 22 skipping to change at page 91, line 18
m=audio 9 UDP/TLS/RTP/SAVPF 96 0 8 97 98 m=audio 9 UDP/TLS/RTP/SAVPF 96 0 8 97 98
c=IN IP4 0.0.0.0 c=IN IP4 0.0.0.0
a=mid:a1 a=mid:a1
a=sendonly a=sendonly
a=rtpmap:96 opus/48000/2 a=rtpmap:96 opus/48000/2
a=rtpmap:0 PCMU/8000 a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000 a=rtpmap:8 PCMA/8000
a=rtpmap:97 telephone-event/8000 a=rtpmap:97 telephone-event/8000
a=rtpmap:98 telephone-event/48000 a=rtpmap:98 telephone-event/48000
a=fmtp:97 0-15
a=fmtp:98 0-15
a=maxptime:120 a=maxptime:120
a=extmap:1 urn:ietf:params:rtp-hdrext:sdes:mid a=extmap:1 urn:ietf:params:rtp-hdrext:sdes:mid
a=extmap:2 urn:ietf:params:rtp-hdrext:ssrc-audio-level a=extmap:2 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=msid:751f239e-4ae0-c549-aa3d-890de772998b a=msid:751f239e-4ae0-c549-aa3d-890de772998b
04b5a445-82cc-c9e8-9ffe-c24d0ef4b0ff 04b5a445-82cc-c9e8-9ffe-c24d0ef4b0ff
a=ice-ufrag:TpaA a=ice-ufrag:TpaA
a=ice-pwd:t2Ouhc67y8JcCaYZxUUTgKw/ a=ice-pwd:t2Ouhc67y8JcCaYZxUUTgKw/
a=fingerprint:sha-256 a=fingerprint:sha-256
A2:F3:A5:6D:4C:8C:1E:B2:62:10:4A:F6:70:61:C4:FC: A2:F3:A5:6D:4C:8C:1E:B2:62:10:4A:F6:70:61:C4:FC:
3C:E0:01:D6:F3:24:80:74:DA:7C:3E:50:18:7B:CE:4D 3C:E0:01:D6:F3:24:80:74:DA:7C:3E:50:18:7B:CE:4D
a=setup:active a=setup:active
a=dtls-id:1 a=tls-id:1
a=rtcp-mux a=rtcp-mux
a=rtcp-mux-only a=rtcp-mux-only
a=rtcp-rsize a=rtcp-rsize
m=video 9 UDP/TLS/RTP/SAVPF 100 101 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 rtx/90000 a=rtpmap:101 H264/90000
a=fmtp:101 apt=100 a=fmtp:101 packetization-mode=1;profile-level-id=42e01f
a=rtpmap:102 rtx/90000
a=fmtp:102 apt=100
=rtpmap:103 rtx/90000
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=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
39292672-c102-d075-f580-5826f31ca958 39292672-c102-d075-f580-5826f31ca958
|answer-C1-candidate-1| looks like: |answer-C1-candidate-1| looks like:
ufrag TpaA ufrag TpaA
index 0 index 0
skipping to change at page 91, line 32 skipping to change at page 92, line 34
m=audio 12200 UDP/TLS/RTP/SAVPF 96 0 8 97 98 m=audio 12200 UDP/TLS/RTP/SAVPF 96 0 8 97 98
c=IN IP4 192.0.2.200 c=IN IP4 192.0.2.200
a=mid:a1 a=mid:a1
a=sendrecv a=sendrecv
a=rtpmap:96 opus/48000/2 a=rtpmap:96 opus/48000/2
a=rtpmap:0 PCMU/8000 a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000 a=rtpmap:8 PCMA/8000
a=rtpmap:97 telephone-event/8000 a=rtpmap:97 telephone-event/8000
a=rtpmap:98 telephone-event/48000 a=rtpmap:98 telephone-event/48000
a=fmtp:97 0-15
a=fmtp:98 0-15
a=maxptime:120 a=maxptime:120
a=extmap:1 urn:ietf:params:rtp-hdrext:sdes:mid a=extmap:1 urn:ietf:params:rtp-hdrext:sdes:mid
a=extmap:2 urn:ietf:params:rtp-hdrext:ssrc-audio-level a=extmap:2 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=msid:751f239e-4ae0-c549-aa3d-890de772998b a=msid:751f239e-4ae0-c549-aa3d-890de772998b
04b5a445-82cc-c9e8-9ffe-c24d0ef4b0ff 04b5a445-82cc-c9e8-9ffe-c24d0ef4b0ff
a=ice-ufrag:TpaA a=ice-ufrag:TpaA
a=ice-pwd:t2Ouhc67y8JcCaYZxUUTgKw/ a=ice-pwd:t2Ouhc67y8JcCaYZxUUTgKw/
a=fingerprint:sha-256 a=fingerprint:sha-256
A2:F3:A5:6D:4C:8C:1E:B2:62:10:4A:F6:70:61:C4:FC: A2:F3:A5:6D:4C:8C:1E:B2:62:10:4A:F6:70:61:C4:FC:
3C:E0:01:D6:F3:24:80:74:DA:7C:3E:50:18:7B:CE:4D 3C:E0:01:D6:F3:24:80:74:DA:7C:3E:50:18:7B:CE:4D
a=setup:actpass a=setup:actpass
a=dtls-id:1 a=tls-id:1
a=rtcp-mux a=rtcp-mux
a=rtcp-mux-only a=rtcp-mux-only
a=rtcp-rsize a=rtcp-rsize
a=candidate:1 1 udp 255 192.0.2.200 12200 typ relay a=candidate:1 1 udp 255 192.0.2.200 12200 typ relay
raddr 0.0.0.0 rport 0 raddr 0.0.0.0 rport 0
a=end-of-candidates a=end-of-candidates
m=video 12200 UDP/TLS/RTP/SAVPF 100 101
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 rtx/90000 a=rtpmap:101 H264/90000
a=fmtp:101 apt=100 a=fmtp:101 packetization-mode=1;profile-level-id=42e01f
a=rtpmap:102 rtx/90000
a=fmtp:102 apt=100
=rtpmap:103 rtx/90000
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=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
39292672-c102-d075-f580-5826f31ca958 39292672-c102-d075-f580-5826f31ca958
The SDP for |answer-C2| looks like: The SDP for |answer-C2| looks like:
v=0 v=0
o=- 1070771854436052752 2 IN IP4 0.0.0.0 o=- 1070771854436052752 2 IN IP4 0.0.0.0
skipping to change at page 92, line 37 skipping to change at page 93, line 49
m=audio 12100 UDP/TLS/RTP/SAVPF 96 0 8 97 98 m=audio 12100 UDP/TLS/RTP/SAVPF 96 0 8 97 98
c=IN IP4 192.0.2.100 c=IN IP4 192.0.2.100
a=mid:a1 a=mid:a1
a=sendrecv a=sendrecv
a=rtpmap:96 opus/48000/2 a=rtpmap:96 opus/48000/2
a=rtpmap:0 PCMU/8000 a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000 a=rtpmap:8 PCMA/8000
a=rtpmap:97 telephone-event/8000 a=rtpmap:97 telephone-event/8000
a=rtpmap:98 telephone-event/48000 a=rtpmap:98 telephone-event/48000
a=fmtp:97 0-15
a=fmtp:98 0-15
a=maxptime:120 a=maxptime:120
a=extmap:1 urn:ietf:params:rtp-hdrext:sdes:mid a=extmap:1 urn:ietf:params:rtp-hdrext:sdes:mid
a=extmap:2 urn:ietf:params:rtp-hdrext:ssrc-audio-level a=extmap:2 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=msid:bbce3ba6-abfc-ac63-d00a-e15b286f8fce a=msid:bbce3ba6-abfc-ac63-d00a-e15b286f8fce
e80098db-7159-3c06-229a-00df2a9b3dbc e80098db-7159-3c06-229a-00df2a9b3dbc
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:passive a=setup:passive
a=dtls-id:1 a=tls-id:1
a=rtcp-mux a=rtcp-mux
a=rtcp-mux-only a=rtcp-mux-only
a=rtcp-rsize a=rtcp-rsize
a=candidate:1 1 udp 255 192.0.2.100 12100 typ relay a=candidate:1 1 udp 255 192.0.2.100 12100 typ relay
raddr 0.0.0.0 rport 0 raddr 0.0.0.0 rport 0
a=end-of-candidates a=end-of-candidates
m=video 12100 UDP/TLS/RTP/SAVPF 100 101 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 rtx/90000 a=rtpmap:101 H264/90000
a=fmtp:101 apt=100 a=fmtp:101 packetization-mode=1;profile-level-id=42e01f
a=rtpmap:102 rtx/90000
a=fmtp:102 apt=100
=rtpmap:103 rtx/90000
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=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
ac701365-eb06-42df-cc93-7f22bc308789 ac701365-eb06-42df-cc93-7f22bc308789
8. Security Considerations 8. Security Considerations
The IETF has published separate documents The IETF has published separate documents
[I-D.ietf-rtcweb-security-arch] [I-D.ietf-rtcweb-security] describing [I-D.ietf-rtcweb-security-arch] [I-D.ietf-rtcweb-security] describing
skipping to change at page 94, line 26 skipping to change at page 95, line 45
Adam Bergkvist, Dan Burnett, Ben Campbell, Alissa Cooper, Richard Adam Bergkvist, Dan Burnett, Ben Campbell, Alissa Cooper, Richard
Ejzak, Stefan Hakansson, Ted Hardie, Christer Holmberg Andrew Hutton, Ejzak, Stefan Hakansson, Ted Hardie, Christer Holmberg Andrew Hutton,
Randell Jesup, Matthew Kaufman, Anant Narayanan, Adam Roach, Neil Randell Jesup, Matthew Kaufman, Anant Narayanan, Adam Roach, Neil
Stratford, Martin Thomson, Sean Turner, and Magnus Westerlund all Stratford, Martin Thomson, Sean Turner, and Magnus Westerlund all
provided valuable feedback on this proposal. provided valuable feedback on this proposal.
11. References 11. References
11.1. Normative References 11.1. Normative References
[I-D.ietf-avtcore-rtp-multi-stream]
Lennox, J., Westerlund, M., Wu, Q., and C. Perkins,
"Sending Multiple RTP Streams in a Single RTP Session",
draft-ietf-avtcore-rtp-multi-stream-11 (work in progress),
December 2015.
[I-D.ietf-avtext-rid] [I-D.ietf-avtext-rid]
Roach, A., Nandakumar, S., and P. Thatcher, "RTP Stream Roach, A., Nandakumar, S., and P. Thatcher, "RTP Stream
Identifier (RID) Source Description (SDES)", draft-ietf- Identifier Source Description (SDES)", draft-ietf-avtext-
avtext-rid-00 (work in progress), February 2016. rid-09 (work in progress), October 2016.
[I-D.ietf-ice-trickle] [I-D.ietf-ice-trickle]
Ivov, E., Rescorla, E., Uberti, J., and P. Saint-Andre, Ivov, E., Rescorla, E., Uberti, J., and P. Saint-Andre,
"Trickle ICE: Incremental Provisioning of Candidates for "Trickle ICE: Incremental Provisioning of Candidates for
the Interactive Connectivity Establishment (ICE) the Interactive Connectivity Establishment (ICE)
Protocol". Protocol", draft-ietf-ice-trickle-12 (work in progress),
June 2017.
[I-D.ietf-mmusic-4572-update]
Holmberg, C., "Updates to RFC 4572", draft-ietf-mmusic-
4572-update-05 (work in progress), June 2016.
[I-D.ietf-mmusic-dtls-sdp] [I-D.ietf-mmusic-dtls-sdp]
Holmberg, C. and R. Shpount, "Using the SDP Offer/Answer Holmberg, C. and R. Shpount, "Using the SDP Offer/Answer
Mechanism for DTLS", draft-ietf-mmusic-dtls-sdp-14 (work Mechanism for DTLS", draft-ietf-mmusic-dtls-sdp-26 (work
in progress), July 2016. in progress), June 2017.
[I-D.ietf-mmusic-msid] [I-D.ietf-mmusic-msid]
Alvestrand, H., "Cross Session Stream Identification in Alvestrand, H., "WebRTC MediaStream Identification in the
the Session Description Protocol", draft-ietf-mmusic- Session Description Protocol", draft-ietf-mmusic-msid-16
msid-01 (work in progress), August 2013. (work in progress), February 2017.
[I-D.ietf-mmusic-mux-exclusive] [I-D.ietf-mmusic-mux-exclusive]
Holmberg, C., "Indicating Exclusive Support of RTP/RTCP Holmberg, C., "Indicating Exclusive Support of RTP/RTCP
Multiplexing using SDP", draft-ietf-mmusic-mux- Multiplexing using SDP", draft-ietf-mmusic-mux-
exclusive-08 (work in progress), June 2016. exclusive-12 (work in progress), May 2017.
[I-D.ietf-mmusic-rid] [I-D.ietf-mmusic-rid]
Thatcher, P., Zanaty, M., Nandakumar, S., Burman, B., Thatcher, P., Zanaty, M., Nandakumar, S., Burman, B.,
Roach, A., and B. Campen, "RTP Payload Format Roach, A., and B. Campen, "RTP Payload Format
Constraints", draft-ietf-mmusic-rid-04 (work in progress), Restrictions", draft-ietf-mmusic-rid-10 (work in
February 2016. progress), March 2017.
[I-D.ietf-mmusic-sctp-sdp] [I-D.ietf-mmusic-sctp-sdp]
Loreto, S. and G. Camarillo, "Stream Control Transmission Holmberg, C., Shpount, R., Loreto, S., and G. Camarillo,
Protocol (SCTP)-Based Media Transport in the Session "Session Description Protocol (SDP) Offer/Answer
Description Protocol (SDP)", draft-ietf-mmusic-sctp-sdp-04 Procedures For Stream Control Transmission Protocol (SCTP)
(work in progress), June 2013. 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] [I-D.ietf-mmusic-sdp-bundle-negotiation]
Holmberg, C., Alvestrand, H., and C. Jennings, Holmberg, C., Alvestrand, H., and C. Jennings,
"Multiplexing Negotiation Using Session Description "Negotiating Media Multiplexing Using the Session
Protocol (SDP) Port Numbers", draft-ietf-mmusic-sdp- Description Protocol (SDP)", draft-ietf-mmusic-sdp-bundle-
bundle-negotiation-04 (work in progress), June 2013. negotiation-38 (work in progress), April 2017.
[I-D.ietf-mmusic-sdp-mux-attributes] [I-D.ietf-mmusic-sdp-mux-attributes]
Nandakumar, S., "A Framework for SDP Attributes when Nandakumar, S., "A Framework for SDP Attributes when
Multiplexing", draft-ietf-mmusic-sdp-mux-attributes-01 Multiplexing", draft-ietf-mmusic-sdp-mux-attributes-16
(work in progress), February 2014. (work in progress), December 2016.
[I-D.ietf-mmusic-sdp-simulcast] [I-D.ietf-mmusic-sdp-simulcast]
Burman, B., Westerlund, M., Nandakumar, S., and M. Zanaty, Burman, B., Westerlund, M., Nandakumar, S., and M. Zanaty,
"Using Simulcast in SDP and RTP Sessions", draft-ietf- "Using Simulcast in SDP and RTP Sessions", draft-ietf-
mmusic-sdp-simulcast-04 (work in progress), February 2016. mmusic-sdp-simulcast-08 (work in progress), March 2017.
[I-D.ietf-rtcweb-audio]
Valin, J. and C. Bran, "WebRTC Audio Codec and Processing
Requirements", draft-ietf-rtcweb-audio-02 (work in
progress), August 2013.
[I-D.ietf-rtcweb-fec] [I-D.ietf-rtcweb-fec]
Uberti, J., "WebRTC Forward Error Correction Uberti, J., "WebRTC Forward Error Correction
Requirements", draft-ietf-rtcweb-fec-00 (work in Requirements", draft-ietf-rtcweb-fec-05 (work in
progress), February 2015. progress), May 2017.
[I-D.ietf-rtcweb-rtp-usage] [I-D.ietf-rtcweb-rtp-usage]
Perkins, C., Westerlund, M., and J. Ott, "Web Real-Time Perkins, C., Westerlund, M., and J. Ott, "Web Real-Time
Communication (WebRTC): Media Transport and Use of RTP", Communication (WebRTC): Media Transport and Use of RTP",
draft-ietf-rtcweb-rtp-usage-09 (work in progress), draft-ietf-rtcweb-rtp-usage-26 (work in progress), March
September 2013. 2016.
[I-D.ietf-rtcweb-security] [I-D.ietf-rtcweb-security]
Rescorla, E., "Security Considerations for WebRTC", draft- Rescorla, E., "Security Considerations for WebRTC", draft-
ietf-rtcweb-security-06 (work in progress), January 2014. ietf-rtcweb-security-08 (work in progress), February 2015.
[I-D.ietf-rtcweb-security-arch] [I-D.ietf-rtcweb-security-arch]
Rescorla, E., "WebRTC Security Architecture", draft-ietf- Rescorla, E., "WebRTC Security Architecture", draft-ietf-
rtcweb-security-arch-09 (work in progress), February 2014. rtcweb-security-arch-12 (work in progress), June 2016.
[I-D.ietf-rtcweb-video]
Roach, A., "WebRTC Video Processing and Codec
Requirements", draft-ietf-rtcweb-video-00 (work in
progress), July 2014.
[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, March 1997. Requirement Levels", BCP 14, RFC 2119,
DOI 10.17487/RFC2119, March 1997,
<http://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,
June 2002. DOI 10.17487/RFC3261, June 2002,
<http://www.rfc-editor.org/info/rfc3261>.
[RFC3264] Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model [RFC3264] Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model
with Session Description Protocol (SDP)", RFC 3264, June with Session Description Protocol (SDP)", RFC 3264,
2002. DOI 10.17487/RFC3264, June 2002,
<http://www.rfc-editor.org/info/rfc3264>.
[RFC3552] Rescorla, E. and B. Korver, "Guidelines for Writing RFC [RFC3552] Rescorla, E. and B. Korver, "Guidelines for Writing RFC
Text on Security Considerations", BCP 72, RFC 3552, July Text on Security Considerations", BCP 72, RFC 3552,
2003. DOI 10.17487/RFC3552, July 2003,
<http://www.rfc-editor.org/info/rfc3552>.
[RFC3605] Huitema, C., "Real Time Control Protocol (RTCP) attribute [RFC3605] Huitema, C., "Real Time Control Protocol (RTCP) attribute
in Session Description Protocol (SDP)", RFC 3605, October in Session Description Protocol (SDP)", RFC 3605,
2003. DOI 10.17487/RFC3605, October 2003,
<http://www.rfc-editor.org/info/rfc3605>.
[RFC3711] Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K.
Norrman, "The Secure Real-time Transport Protocol (SRTP)",
RFC 3711, DOI 10.17487/RFC3711, March 2004,
<http://www.rfc-editor.org/info/rfc3711>.
[RFC3890] Westerlund, M., "A Transport Independent Bandwidth [RFC3890] Westerlund, M., "A Transport Independent Bandwidth
Modifier for the Session Description Protocol (SDP)", RFC Modifier for the Session Description Protocol (SDP)",
3890, DOI 10.17487/RFC3890, September 2004, RFC 3890, DOI 10.17487/RFC3890, September 2004,
<http://www.rfc-editor.org/info/rfc3890>. <http://www.rfc-editor.org/info/rfc3890>.
[RFC4145] Yon, D. and G. Camarillo, "TCP-Based Media Transport in [RFC4145] Yon, D. and G. Camarillo, "TCP-Based Media Transport in
the Session Description Protocol (SDP)", RFC 4145, the Session Description Protocol (SDP)", RFC 4145,
September 2005. DOI 10.17487/RFC4145, September 2005,
<http://www.rfc-editor.org/info/rfc4145>.
[RFC4566] Handley, M., Jacobson, V., and C. Perkins, "SDP: Session [RFC4566] Handley, M., Jacobson, V., and C. Perkins, "SDP: Session
Description Protocol", RFC 4566, July 2006. Description Protocol", RFC 4566, DOI 10.17487/RFC4566,
July 2006, <http://www.rfc-editor.org/info/rfc4566>.
[RFC4572] Lennox, J., "Connection-Oriented Media Transport over the
Transport Layer Security (TLS) Protocol in the Session
Description Protocol (SDP)", RFC 4572, July 2006.
[RFC4585] Ott, J., Wenger, S., Sato, N., Burmeister, C., and J. Rey, [RFC4585] Ott, J., Wenger, S., Sato, N., Burmeister, C., and J. Rey,
"Extended RTP Profile for Real-time Transport Control "Extended RTP Profile for Real-time Transport Control
Protocol (RTCP)-Based Feedback (RTP/AVPF)", RFC 4585, July Protocol (RTCP)-Based Feedback (RTP/AVPF)", RFC 4585,
2006. DOI 10.17487/RFC4585, July 2006,
<http://www.rfc-editor.org/info/rfc4585>.
[RFC5124] Ott, J. and E. Carrara, "Extended Secure RTP Profile for
Real-time Transport Control Protocol (RTCP)-Based Feedback
(RTP/SAVPF)", RFC 5124, DOI 10.17487/RFC5124, February
2008, <http://www.rfc-editor.org/info/rfc5124>.
[RFC5245] Rosenberg, J., "Interactive Connectivity Establishment [RFC5245] Rosenberg, J., "Interactive Connectivity Establishment
(ICE): A Protocol for Network Address Translator (NAT) (ICE): A Protocol for Network Address Translator (NAT)
Traversal for Offer/Answer Protocols", RFC 5245, April Traversal for Offer/Answer Protocols", RFC 5245,
2010. DOI 10.17487/RFC5245, April 2010,
<http://www.rfc-editor.org/info/rfc5245>.
[RFC5285] Singer, D. and H. Desineni, "A General Mechanism for RTP [RFC5285] Singer, D. and H. Desineni, "A General Mechanism for RTP
Header Extensions", RFC 5285, July 2008. Header Extensions", RFC 5285, DOI 10.17487/RFC5285, July
2008, <http://www.rfc-editor.org/info/rfc5285>.
[RFC5761] Perkins, C. and M. Westerlund, "Multiplexing RTP Data and [RFC5761] Perkins, C. and M. Westerlund, "Multiplexing RTP Data and
Control Packets on a Single Port", RFC 5761, April 2010. Control Packets on a Single Port", RFC 5761,
DOI 10.17487/RFC5761, April 2010,
<http://www.rfc-editor.org/info/rfc5761>.
[RFC5888] Camarillo, G. and H. Schulzrinne, "The Session Description [RFC5888] Camarillo, G. and H. Schulzrinne, "The Session Description
Protocol (SDP) Grouping Framework", RFC 5888, June 2010. Protocol (SDP) Grouping Framework", RFC 5888,
DOI 10.17487/RFC5888, June 2010,
<http://www.rfc-editor.org/info/rfc5888>.
[RFC6236] Johansson, I. and K. Jung, "Negotiation of Generic Image [RFC6236] Johansson, I. and K. Jung, "Negotiation of Generic Image
Attributes in the Session Description Protocol (SDP)", RFC Attributes in the Session Description Protocol (SDP)",
6236, May 2011. RFC 6236, DOI 10.17487/RFC6236, May 2011,
<http://www.rfc-editor.org/info/rfc6236>.
[RFC6347] Rescorla, E. and N. Modadugu, "Datagram Transport Layer [RFC6347] Rescorla, E. and N. Modadugu, "Datagram Transport Layer
Security Version 1.2", RFC 6347, January 2012. Security Version 1.2", RFC 6347, DOI 10.17487/RFC6347,
January 2012, <http://www.rfc-editor.org/info/rfc6347>.
[RFC6716] Valin, JM., Vos, K., and T. Terriberry, "Definition of the [RFC6716] Valin, JM., Vos, K., and T. Terriberry, "Definition of the
Opus Audio Codec", RFC 6716, DOI 10.17487/RFC6716, Opus Audio Codec", RFC 6716, DOI 10.17487/RFC6716,
September 2012, <http://www.rfc-editor.org/info/rfc6716>. September 2012, <http://www.rfc-editor.org/info/rfc6716>.
[RFC6904] Lennox, J., "Encryption of Header Extensions in the Secure [RFC6904] Lennox, J., "Encryption of Header Extensions in the Secure
Real-time Transport Protocol (SRTP)", RFC 6904, April Real-time Transport Protocol (SRTP)", RFC 6904,
2013. DOI 10.17487/RFC6904, April 2013,
<http://www.rfc-editor.org/info/rfc6904>.
[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, DOI 10.17487/ Clock Rates in an RTP Session", RFC 7160,
RFC7160, April 2014, DOI 10.17487/RFC7160, April 2014,
<http://www.rfc-editor.org/info/rfc7160>. <http://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, DOI for the Opus Speech and Audio Codec", RFC 7587,
10.17487/RFC7587, June 2015, DOI 10.17487/RFC7587, June 2015,
<http://www.rfc-editor.org/info/rfc7587>. <http://www.rfc-editor.org/info/rfc7587>.
[RFC7742] Roach, A., "WebRTC Video Processing and Codec
Requirements", RFC 7742, DOI 10.17487/RFC7742, March 2016,
<http://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,
<http://www.rfc-editor.org/info/rfc7850>. <http://www.rfc-editor.org/info/rfc7850>.
[RFC7941] Westerlund, M., Burman, B., Even, R., and M. Zanaty, "RTP [RFC7874] Valin, JM. and C. Bran, "WebRTC Audio Codec and Processing
Header Extension for the RTP Control Protocol (RTCP) Requirements", RFC 7874, DOI 10.17487/RFC7874, May 2016,
Source Description Items", RFC 7941, DOI 10.17487/RFC7941, <http://www.rfc-editor.org/info/rfc7874>.
August 2016, <http://www.rfc-editor.org/info/rfc7941>.
[RFC8108] Lennox, J., Westerlund, M., Wu, Q., and C. Perkins,
"Sending Multiple RTP Streams in a Single RTP Session",
RFC 8108, DOI 10.17487/RFC8108, March 2017,
<http://www.rfc-editor.org/info/rfc8108>.
[RFC8122] Lennox, J. and C. Holmberg, "Connection-Oriented Media
Transport over the Transport Layer Security (TLS) Protocol
in the Session Description Protocol (SDP)", RFC 8122,
DOI 10.17487/RFC8122, March 2017,
<http://www.rfc-editor.org/info/rfc8122>.
11.2. Informative References 11.2. Informative References
[I-D.ietf-avtext-lrr] [I-D.ietf-mmusic-trickle-ice-sip]
Lennox, J., Hong, D., Uberti, J., Homer, S., and M. Ivov, E., Stach, T., Marocco, E., and C. Holmberg, "A
Flodman, "The Layer Refresh Request (LRR) RTCP Feedback Session Initiation Protocol (SIP) usage for Trickle ICE",
Message", draft-ietf-avtext-lrr-03 (work in progress), draft-ietf-mmusic-trickle-ice-sip-07 (work in progress),
July 2016. March 2017.
[I-D.ietf-rtcweb-ip-handling] [I-D.ietf-rtcweb-ip-handling]
Uberti, J. and G. Shieh, "WebRTC IP Address Handling Uberti, J. and G. Shieh, "WebRTC IP Address Handling
Recommendations", draft-ietf-rtcweb-ip-handling-01 (work Requirements", draft-ietf-rtcweb-ip-handling-03 (work in
in progress), March 2016. progress), January 2017.
[I-D.nandakumar-rtcweb-sdp] [I-D.ietf-rtcweb-sdp]
Nandakumar, S. and C. Jennings, "SDP for the WebRTC", Nandakumar, S. and C. Jennings, "Annotated Example SDP for
draft-nandakumar-rtcweb-sdp-02 (work in progress), July WebRTC", draft-ietf-rtcweb-sdp-06 (work in progress),
2013. April 2017.
[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, September 2002. Comfort Noise (CN)", RFC 3389, DOI 10.17487/RFC3389,
September 2002, <http://www.rfc-editor.org/info/rfc3389>.
[RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V.
Jacobson, "RTP: A Transport Protocol for Real-Time
Applications", STD 64, RFC 3550, DOI 10.17487/RFC3550,
July 2003, <http://www.rfc-editor.org/info/rfc3550>.
[RFC3556] Casner, S., "Session Description Protocol (SDP) Bandwidth [RFC3556] Casner, S., "Session Description Protocol (SDP) Bandwidth
Modifiers for RTP Control Protocol (RTCP) Bandwidth", RFC Modifiers for RTP Control Protocol (RTCP) Bandwidth",
3556, July 2003. RFC 3556, DOI 10.17487/RFC3556, July 2003,
<http://www.rfc-editor.org/info/rfc3556>.
[RFC3611] Friedman, T., Caceres, R., and A. Clark, "RTP Control
Protocol Extended Reports (RTCP XR)", RFC 3611, DOI
10.17487/RFC3611, November 2003,
<http://www.rfc-editor.org/info/rfc3611>.
[RFC3960] Camarillo, G. and H. Schulzrinne, "Early Media and Ringing [RFC3960] Camarillo, G. and H. Schulzrinne, "Early Media and Ringing
Tone Generation in the Session Initiation Protocol (SIP)", Tone Generation in the Session Initiation Protocol (SIP)",
RFC 3960, December 2004. RFC 3960, DOI 10.17487/RFC3960, December 2004,
<http://www.rfc-editor.org/info/rfc3960>.
[RFC4568] Andreasen, F., Baugher, M., and D. Wing, "Session [RFC4568] Andreasen, F., Baugher, M., and D. Wing, "Session
Description Protocol (SDP) Security Descriptions for Media Description Protocol (SDP) Security Descriptions for Media
Streams", RFC 4568, July 2006. Streams", RFC 4568, DOI 10.17487/RFC4568, July 2006,
<http://www.rfc-editor.org/info/rfc4568>.
[RFC4588] Rey, J., Leon, D., Miyazaki, A., Varsa, V., and R. [RFC4588] Rey, J., Leon, D., Miyazaki, A., Varsa, V., and R.
Hakenberg, "RTP Retransmission Payload Format", RFC 4588, Hakenberg, "RTP Retransmission Payload Format", RFC 4588,
July 2006. DOI 10.17487/RFC4588, July 2006,
<http://www.rfc-editor.org/info/rfc4588>.
[RFC4733] Schulzrinne, H. and T. Taylor, "RTP Payload for DTMF [RFC4733] Schulzrinne, H. and T. Taylor, "RTP Payload for DTMF
Digits, Telephony Tones, and Telephony Signals", RFC 4733, Digits, Telephony Tones, and Telephony Signals", RFC 4733,
DOI 10.17487/RFC4733, December 2006, DOI 10.17487/RFC4733, December 2006,
<http://www.rfc-editor.org/info/rfc4733>. <http://www.rfc-editor.org/info/rfc4733>.
[RFC5104] Wenger, S., Chandra, U., Westerlund, M., and B. Burman,
"Codec Control Messages in the RTP Audio-Visual Profile
with Feedback (AVPF)", RFC 5104, DOI 10.17487/RFC5104,
February 2008, <http://www.rfc-editor.org/info/rfc5104>.
[RFC5506] Johansson, I. and M. Westerlund, "Support for Reduced-Size [RFC5506] Johansson, I. and M. Westerlund, "Support for Reduced-Size
Real-Time Transport Control Protocol (RTCP): Opportunities Real-Time Transport Control Protocol (RTCP): Opportunities
and Consequences", RFC 5506, April 2009. and Consequences", RFC 5506, DOI 10.17487/RFC5506, April
2009, <http://www.rfc-editor.org/info/rfc5506>.
[RFC5576] Lennox, J., Ott, J., and T. Schierl, "Source-Specific [RFC5576] Lennox, J., Ott, J., and T. Schierl, "Source-Specific
Media Attributes in the Session Description Protocol Media Attributes in the Session Description Protocol
(SDP)", RFC 5576, June 2009. (SDP)", RFC 5576, DOI 10.17487/RFC5576, June 2009,
<http://www.rfc-editor.org/info/rfc5576>.
[RFC5763] Fischl, J., Tschofenig, H., and E. Rescorla, "Framework [RFC5763] Fischl, J., Tschofenig, H., and E. Rescorla, "Framework
for Establishing a Secure Real-time Transport Protocol for Establishing a Secure Real-time Transport Protocol
(SRTP) Security Context Using Datagram Transport Layer (SRTP) Security Context Using Datagram Transport Layer
Security (DTLS)", RFC 5763, May 2010. Security (DTLS)", RFC 5763, DOI 10.17487/RFC5763, May
2010, <http://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, May 2010. Real-time Transport Protocol (SRTP)", RFC 5764,
DOI 10.17487/RFC5764, May 2010,
<http://www.rfc-editor.org/info/rfc5764>.
[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, DOI 10.17487/ Mixer Audio Level Indication", RFC 6464,
RFC6464, December 2011, DOI 10.17487/RFC6464, December 2011,
<http://www.rfc-editor.org/info/rfc6464>. <http://www.rfc-editor.org/info/rfc6464>.
[RFC6544] Rosenberg, J., Keranen, A., Lowekamp, B., and A. Roach, [RFC6544] Rosenberg, J., Keranen, A., Lowekamp, B., and A. Roach,
"TCP Candidates with Interactive Connectivity "TCP Candidates with Interactive Connectivity
Establishment (ICE)", RFC 6544, DOI 10.17487/RFC6544, Establishment (ICE)", RFC 6544, DOI 10.17487/RFC6544,
March 2012, <http://www.rfc-editor.org/info/rfc6544>. March 2012, <http://www.rfc-editor.org/info/rfc6544>.
[RFC7656] Lennox, J., Gross, K., Nandakumar, S., Salgueiro, G., and
B. Burman, Ed., "A Taxonomy of Semantics and Mechanisms
for Real-Time Transport Protocol (RTP) Sources", RFC 7656,
DOI 10.17487/RFC7656, November 2015,
<http://www.rfc-editor.org/info/rfc7656>.
[TS26.114] [TS26.114]
3GPP TS 26.114 V12.8.0, "3rd Generation Partnership 3GPP TS 26.114 V12.8.0, "3rd Generation Partnership
Project; Technical Specification Group Services and System Project; Technical Specification Group Services and System
Aspects; IP Multimedia Subsystem (IMS); Multimedia Aspects; IP Multimedia Subsystem (IMS); Multimedia
Telephony; Media handling and interaction (Release 12)", Telephony; Media handling and interaction (Release 12)",
December 2014, <http://www.3gpp.org/DynaReport/26114.htm>. December 2014, <http://www.3gpp.org/DynaReport/26114.htm>.
[W3C.WD-webrtc-20140617] [W3C.webrtc]
Bergkvist, A., Burnett, D., Narayanan, A., and C. Bergkvist, A., Burnett, D., Jennings, C., Narayanan, A.,
Jennings, "WebRTC 1.0: Real-time Communication Between Aboba, B., and T. Brandstetter, "WebRTC 1.0: Real-time
Browsers", World Wide Web Consortium WD WD-webrtc- Communication Between Browsers", World Wide Web Consortium
20140617, June 2014, WD WD-webrtc-20170515, May 2017,
<http://www.w3.org/TR/2011/WD-webrtc-20140617>. <https://www.w3.org/TR/2017/WD-webrtc-20170515/>.
Appendix A. Appendix A Appendix A. Appendix A
For the syntax validation performed in Section 5.7, the following For the syntax validation performed in Section 5.7, the following
list of ABNF definitions is used: list of ABNF definitions is used:
+------------------------+------------------------------------------+ +------------------------+------------------------------------------+
| Attribute | Reference | | Attribute | Reference |
+------------------------+------------------------------------------+ +------------------------+------------------------------------------+
| ptime | [RFC4566] Section 9 | | ptime | [RFC4566] Section 9 |
skipping to change at page 101, line 21 skipping to change at page 103, line 21
| recvonly | [RFC4566] Section 9 | | recvonly | [RFC4566] Section 9 |
| sendrecv | [RFC4566] Section 9 | | sendrecv | [RFC4566] Section 9 |
| sendonly | [RFC4566] Section 9 | | sendonly | [RFC4566] Section 9 |
| inactive | [RFC4566] Section 9 | | inactive | [RFC4566] Section 9 |
| framerate | [RFC4566] Section 9 | | framerate | [RFC4566] Section 9 |
| fmtp | [RFC4566] Section 9 | | fmtp | [RFC4566] Section 9 |
| quality | [RFC4566] Section 9 | | quality | [RFC4566] Section 9 |
| rtcp | [RFC3605] Section 2.1 | | rtcp | [RFC3605] Section 2.1 |
| setup | [RFC4145] Sections 3, 4, and 5 | | setup | [RFC4145] Sections 3, 4, and 5 |
| connection | [RFC4145] Sections 3, 4, and 5 | | connection | [RFC4145] Sections 3, 4, and 5 |
| fingerprint | [RFC4572] Section 5 | | fingerprint | [RFC8122] Section 5 |
| rtcp-fb | [RFC4585] Section 4.2 | | rtcp-fb | [RFC4585] Section 4.2 |
| candidate | [RFC5245] Section 15.1 | | candidate | [RFC5245] Section 15.1 |
| remote-candidates | [RFC5245] Section 15.2 | | remote-candidates | [RFC5245] Section 15.2 |
| ice-lite | [RFC5245] Section 15.3 | | ice-lite | [RFC5245] Section 15.3 |
| ice-ufrag | [RFC5245] Section 15.4 | | ice-ufrag | [RFC5245] Section 15.4 |
| ice-pwd | [RFC5245] Section 15.4 | | ice-pwd | [RFC5245] Section 15.4 |
| ice-options | [RFC5245] Section 15.5 | | ice-options | [RFC5245] Section 15.5 |
| extmap | [RFC5285] Section 7 | | extmap | [RFC5285] Section 7 |
| mid | [RFC5888] Section 4 and 5 | | mid | [RFC5888] Sections 4 and 5 |
| group | [RFC5888] Section 4 and 5 | | group | [RFC5888] Sections 4 and 5 |
| imageattr | [RFC6236] Section 3.1 | | imageattr | [RFC6236] Section 3.1 |
| extmap (encrypt | [RFC6904] Section 4 | | extmap (encrypt | [RFC6904] Section 4 |
| option) | | | option) | |
| msid | [I-D.ietf-mmusic-msid] Section 2 | | msid | [I-D.ietf-mmusic-msid] Section 2 |
| rid | [I-D.ietf-mmusic-rid] Section 10 | | rid | [I-D.ietf-mmusic-rid] Section 10 |
| simulcast | [I-D.ietf-mmusic-sdp-simulcast] Section | | simulcast | [I-D.ietf-mmusic-sdp-simulcast] Section |
| | 6.1 | | | 6.1 |
| dtls-id | [I-D.ietf-mmusic-dtls-sdp] Section 4 | | tls-id | [I-D.ietf-mmusic-dtls-sdp] Section 4 |
+------------------------+------------------------------------------+ +------------------------+------------------------------------------+
Table 1: SDP ABNF References Table 1: SDP ABNF References
Appendix B. Change log Appendix B. Change log
Note: This section will be removed by RFC Editor before publication. Note: This section will be removed by RFC Editor before publication.
Changes in draft-21:
o Change dtls-id to tls-id to match MMUSIC draft.
o Replace regular expression for proto field with a list and clarify
that the answer must exactly match the offer.
o Remove text about how to error check on setLocal because local
descriptions cannot be changed.
o Rework silence suppression support to always require that both
sides agree to silence suppression or none is used.
o Remove instructions to parse "a=ssrc-group".
o Allow the addition of new codecs in answers and in subsequent
offers.
o Clarify imageattr processing. Replace use of [x=0,y=0] with
direction indicators.
o Document when early media can occur.
o Fix ICE default port handling when bundle-only is used.
o Forbid duplicating IDENTICAL/TRANSPORT attributes when you are
bundling.
o Clarify the number of components to gather when bundle is
involved.
o Explicitly state that PTs and SSRCs are to be used for demuxing.
o Update guidance on "a=setup" line. This should now match the
MMUSIC draft.
o Update guidance on certificate/digest matching to conform to
RFC8122.
o Update examples.
Changes in draft-20: Changes in draft-20:
o Remove Appendix-B. o Remove Appendix-B.
Changes in draft-19: Changes in draft-19:
o Examples are now machine-generated for correctness, and use IETF- o Examples are now machine-generated for correctness, and use IETF-
approved example IP addresses. approved example IP addresses.
o Add early transport warmup example, and add missing attributes to o Add early transport warmup example, and add missing attributes to
 End of changes. 259 change blocks. 
671 lines changed or deleted 871 lines changed or added

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