draft-ietf-rtcweb-jsep-06.txt   draft-ietf-rtcweb-jsep-07.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: August 17, 2014 Cisco Expires: January 5, 2015 Cisco
February 13, 2014 E. Rescorla, Ed.
Mozilla
July 4, 2014
Javascript Session Establishment Protocol Javascript Session Establishment Protocol
draft-ietf-rtcweb-jsep-06 draft-ietf-rtcweb-jsep-07
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
This Internet-Draft is submitted in full conformance with the This Internet-Draft is submitted in full conformance with the
provisions of BCP 78 and BCP 79. provisions of BCP 78 and BCP 79.
Internet-Drafts are working documents of the Internet Engineering Internet-Drafts are working documents of the Internet Engineering
Task Force (IETF). Note that other groups may also distribute Task Force (IETF). Note that other groups may also distribute
working documents as Internet-Drafts. The list of current Internet- working documents as Internet-Drafts. The list of current Internet-
Drafts is at http://datatracker.ietf.org/drafts/current/. Drafts is at http://datatracker.ietf.org/drafts/current/.
Internet-Drafts are draft documents valid for a maximum of six months Internet-Drafts are draft documents valid for a maximum of six months
and may be updated, replaced, or obsoleted by other documents at any and may be updated, replaced, or obsoleted by other documents at any
time. It is inappropriate to use Internet-Drafts as reference time. It is inappropriate to use Internet-Drafts as reference
material or to cite them other than as "work in progress." material or to cite them other than as "work in progress."
This Internet-Draft will expire on August 17, 2014. This Internet-Draft will expire on January 5, 2015.
Copyright Notice Copyright Notice
Copyright (c) 2014 IETF Trust and the persons identified as the Copyright (c) 2014 IETF Trust and the persons identified as the
document authors. All rights reserved. document authors. All rights reserved.
This document is subject to BCP 78 and the IETF Trust's Legal This document is subject to BCP 78 and the IETF Trust's Legal
Provisions Relating to IETF Documents Provisions Relating to IETF Documents
(http://trustee.ietf.org/license-info) in effect on the date of (http://trustee.ietf.org/license-info) in effect on the date of
publication of this document. Please review these documents publication of this document. Please review these documents
carefully, as they describe your rights and restrictions with respect carefully, as they describe your rights and restrictions with respect
to this document. Code Components extracted from this document must to this document. Code Components extracted from this document must
include Simplified BSD License text as described in Section 4.e of include Simplified BSD License text as described in Section 4.e of
the Trust Legal Provisions and are provided without warranty as the Trust Legal Provisions and are provided without warranty as
described in the Simplified BSD License. described in the Simplified BSD License.
Table of Contents Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 3 1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . . 4
1.1. General Design of JSEP . . . . . . . . . . . . . . . . . 3 1.1. General Design of JSEP . . . . . . . . . . . . . . . . . . 4
1.2. Other Approaches Considered . . . . . . . . . . . . . . . 5 1.2. Other Approaches Considered . . . . . . . . . . . . . . . 5
2. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 6 2. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 6
3. Semantics and Syntax . . . . . . . . . . . . . . . . . . . . 6 3. Semantics and Syntax . . . . . . . . . . . . . . . . . . . . . 6
3.1. Signaling Model . . . . . . . . . . . . . . . . . . . . . 6 3.1. Signaling Model . . . . . . . . . . . . . . . . . . . . . 6
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 . . . . . . . . . . . . . . . . 9
3.4. ICE . . . . . . . . . . . . . . . . . . . . . . . . . . . 10 3.4. ICE . . . . . . . . . . . . . . . . . . . . . . . . . . . 10
3.4.1. ICE Candidate Trickling . . . . . . . . . . . . . . . 10 3.4.1. ICE Candidate Trickling . . . . . . . . . . . . . . . 10
3.4.1.1. ICE Candidate Format . . . . . . . . . . . . . . 11 3.4.1.1. ICE Candidate Format . . . . . . . . . . . . . . . 11
3.4.2. ICE Candidate Pool . . . . . . . . . . . . . . . . . 11 3.4.2. ICE Candidate Pool . . . . . . . . . . . . . . . . . . 11
3.5. Interactions With Forking . . . . . . . . . . . . . . . . 12 3.5. Interactions With Forking . . . . . . . . . . . . . . . . 12
3.5.1. Sequential Forking . . . . . . . . . . . . . . . . . 12 3.5.1. Sequential Forking . . . . . . . . . . . . . . . . . . 12
3.5.2. Parallel Forking . . . . . . . . . . . . . . . . . . 13 3.5.2. Parallel Forking . . . . . . . . . . . . . . . . . . . 13
3.6. Session Rehydration . . . . . . . . . . . . . . . . . . . 14 4. Interface . . . . . . . . . . . . . . . . . . . . . . . . . . 13
4. Interface . . . . . . . . . . . . . . . . . . . . . . . . . . 14 4.1. Methods . . . . . . . . . . . . . . . . . . . . . . . . . 14
4.1. Methods . . . . . . . . . . . . . . . . . . . . . . . . . 14 4.1.1. Constructor . . . . . . . . . . . . . . . . . . . . . 14
4.1.1. Constructor . . . . . . . . . . . . . . . . . . . . . 14 4.1.2. createOffer . . . . . . . . . . . . . . . . . . . . . 14
4.1.2. createOffer . . . . . . . . . . . . . . . . . . . . . 15 4.1.3. createAnswer . . . . . . . . . . . . . . . . . . . . . 15
4.1.3. createAnswer . . . . . . . . . . . . . . . . . . . . 16 4.1.4. SessionDescriptionType . . . . . . . . . . . . . . . . 16
4.1.4. SessionDescriptionType . . . . . . . . . . . . . . . 17 4.1.4.1. Use of Provisional Answers . . . . . . . . . . . . 17
4.1.4.1. Use of Provisional Answers . . . . . . . . . . . 18 4.1.4.2. Rollback . . . . . . . . . . . . . . . . . . . . . 18
4.1.4.2. Rollback . . . . . . . . . . . . . . . . . . . . 18 4.1.5. setLocalDescription . . . . . . . . . . . . . . . . . 18
4.1.5. setLocalDescription . . . . . . . . . . . . . . . . . 19 4.1.6. setRemoteDescription . . . . . . . . . . . . . . . . . 19
4.1.6. setRemoteDescription . . . . . . . . . . . . . . . . 20 4.1.7. localDescription . . . . . . . . . . . . . . . . . . . 19
4.1.7. localDescription . . . . . . . . . . . . . . . . . . 20 4.1.8. remoteDescription . . . . . . . . . . . . . . . . . . 20
4.1.8. remoteDescription . . . . . . . . . . . . . . . . . . 20 4.1.9. updateIce . . . . . . . . . . . . . . . . . . . . . . 20
4.1.9. updateIce . . . . . . . . . . . . . . . . . . . . . . 20 4.1.10. addIceCandidate . . . . . . . . . . . . . . . . . . . 20
4.1.10. addIceCandidate . . . . . . . . . . . . . . . . . . . 21 5. SDP Interaction Procedures . . . . . . . . . . . . . . . . . . 20
5. SDP Interaction Procedures . . . . . . . . . . . . . . . . . 21 5.1. Requirements Overview . . . . . . . . . . . . . . . . . . 21
5.1. Requirements Overview . . . . . . . . . . . . . . . . . . 21 5.1.1. Implementation Requirements . . . . . . . . . . . . . 21
5.1.1. Implementation Requirements . . . . . . . . . . . . . 21 5.1.2. Usage Requirements . . . . . . . . . . . . . . . . . . 22
5.1.2. Usage Requirements . . . . . . . . . . . . . . . . . 23 5.2. Constructing an Offer . . . . . . . . . . . . . . . . . . 22
5.2. Constructing an Offer . . . . . . . . . . . . . . . . . . 23 5.2.1. Initial Offers . . . . . . . . . . . . . . . . . . . . 22
5.2.1. Initial Offers . . . . . . . . . . . . . . . . . . . 23 5.2.2. Subsequent Offers . . . . . . . . . . . . . . . . . . 26
5.2.2. Subsequent Offers . . . . . . . . . . . . . . . . . . 27 5.2.3. Options Handling . . . . . . . . . . . . . . . . . . . 28
5.2.3. Constraints Handling . . . . . . . . . . . . . . . . 29 5.2.3.1. OfferToReceiveAudio . . . . . . . . . . . . . . . 28
5.2.3.1. OfferToReceiveAudio . . . . . . . . . . . . . . . 30 5.2.3.2. OfferToReceiveVideo . . . . . . . . . . . . . . . 29
5.2.3.2. OfferToReceiveVideo . . . . . . . . . . . . . . . 30 5.2.3.3. VoiceActivityDetection . . . . . . . . . . . . . . 29
5.2.3.3. VoiceActivityDetection . . . . . . . . . . . . . 30 5.2.3.4. IceRestart . . . . . . . . . . . . . . . . . . . . 29
5.2.3.4. IceRestart . . . . . . . . . . . . . . . . . . . 30 5.3. Generating an Answer . . . . . . . . . . . . . . . . . . . 29
5.3. Generating an Answer . . . . . . . . . . . . . . . . . . 31 5.3.1. Initial Answers . . . . . . . . . . . . . . . . . . . 30
5.3.1. Initial Answers . . . . . . . . . . . . . . . . . . . 31 5.3.2. Subsequent Answers . . . . . . . . . . . . . . . . . . 33
5.3.2. Subsequent Answers . . . . . . . . . . . . . . . . . 35 5.3.3. Options Handling . . . . . . . . . . . . . . . . . . . 33
5.3.3. Constraints Handling . . . . . . . . . . . . . . . . 35 5.4. Parsing an Offer . . . . . . . . . . . . . . . . . . . . . 33
5.5. Parsing an Answer . . . . . . . . . . . . . . . . . . . . 33
5.4. Parsing an Offer . . . . . . . . . . . . . . . . . . . . 35 5.6. Applying a Local Description . . . . . . . . . . . . . . . 33
5.5. Parsing an Answer . . . . . . . . . . . . . . . . . . . . 35 5.7. Applying a Remote Description . . . . . . . . . . . . . . 33
5.6. Applying a Local Description . . . . . . . . . . . . . . 35 6. Configurable SDP Parameters . . . . . . . . . . . . . . . . . 33
5.7. Applying a Remote Description . . . . . . . . . . . . . . 35 7. Security Considerations . . . . . . . . . . . . . . . . . . . 34
6. Configurable SDP Parameters . . . . . . . . . . . . . . . . . 35 8. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 35
7. Security Considerations . . . . . . . . . . . . . . . . . . . 36 9. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 35
8. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 36 10. References . . . . . . . . . . . . . . . . . . . . . . . . . . 35
9. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 36 10.1. Normative References . . . . . . . . . . . . . . . . . . . 35
10. References . . . . . . . . . . . . . . . . . . . . . . . . . 37 10.2. Informative References . . . . . . . . . . . . . . . . . . 38
10.1. Normative References . . . . . . . . . . . . . . . . . . 37 Appendix A. JSEP Implementation Examples . . . . . . . . . . . . 39
10.2. Informative References . . . . . . . . . . . . . . . . . 38 A.1. Example API Flows . . . . . . . . . . . . . . . . . . . . 39
Appendix A. JSEP Implementation Examples . . . . . . . . . . . . 40 A.1.1. Call using ROAP . . . . . . . . . . . . . . . . . . . 39
A.1. Example API Flows . . . . . . . . . . . . . . . . . . . . 40 A.1.2. Call using XMPP . . . . . . . . . . . . . . . . . . . 40
A.1.1. Call using ROAP . . . . . . . . . . . . . . . . . . . 40 A.1.3. Adding video to a call, using XMPP . . . . . . . . . . 42
A.1.2. Call using XMPP . . . . . . . . . . . . . . . . . . . 41 A.1.4. Simultaneous add of video streams, using XMPP . . . . 42
A.1.3. Adding video to a call, using XMPP . . . . . . . . . 43 A.1.5. Call using SIP . . . . . . . . . . . . . . . . . . . . 43
A.1.4. Simultaneous add of video streams, using XMPP . . . . 43 A.1.6. Handling early media (e.g. 1-800-GO FEDEX), using
A.1.5. Call using SIP . . . . . . . . . . . . . . . . . . . 44 SIP . . . . . . . . . . . . . . . . . . . . . . . . . 44
A.1.6. Handling early media (e.g. 1-800-GO FEDEX), using SIP 45 A.2. Example Session Descriptions . . . . . . . . . . . . . . . 45
A.2. Example Session Descriptions . . . . . . . . . . . . . . 46 A.2.1. createOffer . . . . . . . . . . . . . . . . . . . . . 45
A.2.1. createOffer . . . . . . . . . . . . . . . . . . . . . 46 A.2.2. createAnswer . . . . . . . . . . . . . . . . . . . . . 47
A.2.2. createAnswer . . . . . . . . . . . . . . . . . . . . 48 A.2.3. Call Flows . . . . . . . . . . . . . . . . . . . . . . 49
A.2.3. Call Flows . . . . . . . . . . . . . . . . . . . . . 50 Appendix B. Change log . . . . . . . . . . . . . . . . . . . . . 49
Appendix B. Change log . . . . . . . . . . . . . . . . . . . . . 50 Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 50
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 51
1. Introduction 1. Introduction
This document describes how the W3C WEBRTC RTCPeerConnection This document describes how the W3C WEBRTC RTCPeerConnection
interface[W3C.WD-webrtc-20111027] is used to control the setup, interface[W3C.WD-webrtc-20140617] is used to control the setup,
management and teardown of a multimedia session. management and 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 or Jingle call signaling protocols, or something custom
to the particular application, perhaps for a novel use case. In this to the 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.
The browser environment also has its own challenges that pose
problems for an embedded signaling state machine. One of these is
that the user may reload the web page at any time. If the browser is
fully in charge of the signaling state, this will result in the loss
of the call when this state is wiped by the reload. However, if the
state can be stored at the server, and pushed back down to the new
page, the call can be resumed with minimal interruption.
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. This control of the signaling state machine from Javascript. JSEP removes
mechanism effectively removes the browser almost completely from the the browser almost entirely from the core signaling flow, which is
core signaling flow; the only interface needed is a way for the instead handled by the Javascript making use of two interfaces: (1)
application to pass in the local and remote session descriptions passing in local and remote session descriptions and (2) interacting
negotiated by whatever signaling mechanism is used, and a way to with the ICE state machine.
interact with the ICE state machine.
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 browsers. Note though in many cases it will actually be between two browsers. Note though in many cases it will actually be
between a browser and some kind of server, such as a gateway or MCU. between a browser and some kind of server, such as a gateway or MCU.
This distinction is invisible to the browser; it just follows the This distinction is invisible to the browser; it just follows the
instructions it is given via the API. instructions it is given via the API.
JSEP's handling of session descriptions is simple and JSEP's handling of session descriptions is simple and
straightforward. Whenever an offer/answer exchange is needed, the straightforward. Whenever an offer/answer exchange is needed, the
initiating side creates an offer by calling a createOffer() API. The initiating side creates an offer by calling a createOffer() API. The
skipping to change at page 4, line 47 skipping to change at page 5, line 12
installs it using setRemoteDescription(), and initial setup is installs it using setRemoteDescription(), and initial setup is
complete. This process can be repeated for additional offer/answer complete. This process can be repeated for additional offer/answer
exchanges. 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 browser, because only the browser has the necessary remain in the browser, because only the browser has the necessary
knowledge of candidates and other transport info. Performing this knowledge of candidates and other transport info. Performing this
separation also provides additional flexibility; in protocols that separation also provides additional flexibility; in protocols that
decouple session descriptions from transport, such as Jingle, the decouple session descriptions from transport, such as Jingle, the
transport information can be sent separately; in protocols that session description can be sent immediately and the transport
don't, such as SIP, the information can be used in the aggregated information can be sent when available. In protocols that don't,
form. Sending transport information separately can allow for faster such as SIP, the information can be used in the aggregated form.
ICE and DTLS startup, since the necessary roundtrips can occur while Sending transport information separately can allow for faster ICE and
waiting for the remote side to accept the session. DTLS startup, since ICE checks can start as soon as any transport
information is available rather than waiting for all of it.
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, this library could easily to the application developer. For example, libraries exist to adapt
adapt the JSEP API into the API that was proposed for the ROAP the JSEP API into an API suitable for a SIP or XMPP. Thus, JSEP
signaling protocol [I-D.jennings-rtcweb-signaling], which would provides greater control for the experienced developer without
perform a ROAP call setup under the covers, interacting with the forcing any additional complexity on the novice developer.
application only when it needs a signaling message to be sent. In
the same fashion, one could also implement other popular signaling
protocols, including SIP or Jingle. This allow JSEP to provide
greater control for the experienced developer without forcing any
additional complexity on the novice developer.
1.2. Other Approaches Considered 1.2. Other Approaches Considered
One approach that was considered instead of JSEP was to include a One approach that was considered instead of JSEP was to include a
lightweight signaling protocol. Instead of providing session lightweight signaling protocol. Instead of providing session
descriptions to the API, the API would produce and consume messages descriptions to the API, the API would produce and consume messages
from this protocol. While providing a more high-level API, this put from this protocol. While providing a more high-level API, this put
more control of signaling within the browser, forcing the browser to more control of signaling within the browser, forcing the browser to
have to understand and handle concepts like signaling glare. In have to understand and handle concepts like signaling glare. In
addition, it prevented the application from driving the state machine addition, it prevented the application from driving the state machine
skipping to change at page 6, line 17 skipping to change at page 6, line 26
generating offers and answers out of the browser. Instead of generating offers and answers out of the browser. Instead of
providing createOffer/createAnswer methods within the browser, this providing createOffer/createAnswer methods within the browser, this
approach would instead expose a getCapabilities API which would approach would instead expose a getCapabilities API which would
provide the application with the information it needed in order to provide the application with the information it needed in order to
generate its own session descriptions. This increases the amount of generate its own session descriptions. This increases the amount of
work that the application needs to do; it needs to know how to work that the application needs to do; it needs to know how to
generate session descriptions from capabilities, and especially how generate session descriptions from capabilities, and especially how
to generate the correct answer from an arbitrary offer and the to generate the correct answer from an arbitrary offer and the
supported capabilities. While this could certainly be addressed by supported capabilities. While this could certainly be addressed by
using a library like the one mentioned above, it basically forces the using a library like the one mentioned above, it basically forces the
use of said library even for a simple example. Providing createOffer use of said library even for a simple example. Providing
/createAnswer avoids this problem, but still allows applications to createOffer/createAnswer avoids this problem, but still allows
generate their own offers/answers (to a large extent) if they choose, applications to generate their own offers/answers (to a large extent)
using the description generated by createOffer as an indication of if they choose, using the description generated by createOffer as an
the browser's capabilities. indication of the browser's capabilities.
2. Terminology 2. Terminology
The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT", The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
"SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
document are to be interpreted as described in [RFC2119]. document are to be interpreted as described in [RFC2119].
3. Semantics and Syntax 3. Semantics and Syntax
3.1. Signaling Model 3.1. Signaling Model
skipping to change at page 7, line 35 skipping to change at page 7, line 38
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 SRTP However, not all parameters follow this rule; for example, the SRTP
parameters [RFC4568] sent to a remote party indicate what the local parameters [RFC4568] sent to a remote party indicate what the local
side will use to encrypt, and thereby what the remote party should side will use to encrypt, and thereby what the remote party should
expect to receive; the remote party will have to accept these expect to receive; the remote party will have to accept these
parameters, with no option to choose a different value. parameters, with no option to choose a different value. [[OPEN
ISSUE: This is not correct because we removed SDES
(https://github.com/rtcweb-wg/jsep/issues/10)]]
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, a offer may propose multiple offers versus answers. For example, a offer may propose multiple
SRTP configurations, but an answer may only contain a single SRTP SRTP configurations, but an answer may only contain a single SRTP
configuration. configuration. [[OPEN ISSUE: See issue 10 above.]]
Lastly, while the exact media parameters are only known only after a 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, it is possible for the
offerer to receive media after they have sent an offer and before offerer to receive media after they have sent an offer and before
they have received an answer. To properly process incoming media in they have received an answer. To properly process incoming media in
this case, the offerer's media handler must be aware of the details this case, the offerer's media handler must be aware of the details
of the offer before the answer arrives. 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 user
agent needs: agent 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 a setLocalDescription and a setRemoteDescription methods and having session description objects
setRemoteDescription method and having session description objects
contain a type field indicating the type of session description being contain a type field indicating the type of session description being
supplied. This satisfies the requirements listed above for both the supplied. This satisfies the requirements listed above for both the
offerer, who first calls setLocalDescription(sdp [offer]) and then offerer, who first calls setLocalDescription(sdp [offer]) and then
later setRemoteDescription(sdp [answer]), as well as for the later setRemoteDescription(sdp [answer]), as well as for the
answerer, who first calls setRemoteDescription(sdp [offer]) and then answerer, who first calls setRemoteDescription(sdp [offer]) and then
later setLocalDescription(sdp [answer]). later setLocalDescription(sdp [answer]).
JSEP also allows for an answer to be treated as provisional by the JSEP also allows for an answer to be treated as provisional by the
application. Provisional answers provide a way for an answerer to application. Provisional answers provide a way for an answerer to
communicate initial session parameters back to the offerer, in order communicate initial session parameters back to the offerer, in order
skipping to change at page 8, line 35 skipping to change at page 8, line 35
specified later. This concept of a final answer is important to the specified later. This concept of a final answer is important to the
offer/answer model; when such an answer is received, any extra offer/answer model; when such an answer is received, any extra
resources allocated by the caller can be released, now that the exact resources allocated by the caller can be released, now that the exact
session configuration is known. These "resources" can include things session configuration is known. These "resources" can include things
like extra ICE components, TURN candidates, or video decoders. like extra ICE components, TURN candidates, or video decoders.
Provisional answers, on the other hand, do no such deallocation Provisional answers, on the other hand, do no such deallocation
results; as a result, multiple dissimilar provisional answers can be results; as a result, multiple dissimilar provisional answers can be
received and applied during call setup. received and applied during call setup.
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 from the media offer can be outstanding for a given session, but at the media stack
stack level, a new offer can be generated at any point. For example, level, a new offer can be generated at any point. For example, when
when using SIP for signaling, if one offer is sent, then cancelled using SIP for signaling, if one offer is sent, then cancelled using a
using a SIP CANCEL, another offer can be generated even though no SIP CANCEL, another offer can be generated even though no answer was
answer was received for the first offer. To support this, the JSEP received for the first offer. To support this, the JSEP media layer
media layer can provide an offer whenever the Javascript application can provide an offer via the createOffer() method whenever the
needs one for the signaling. The answerer can send back zero or more Javascript application needs one for the signaling. The answerer can
provisional answers, and finally end the offer-answer exchange by send back zero or more provisional answers, and finally end the
sending a final answer. The state machine for this is as follows: offer-answer exchange by sending a final answer. The state machine
for this is as follows:
setRemote(OFFER) setLocal(PRANSWER) setRemote(OFFER) setLocal(PRANSWER)
/-----\ /-----\ /-----\ /-----\
| | | | | | | |
v | v | v | v |
+---------------+ | +---------------+ | +---------------+ | +---------------+ |
| |----/ | |----/ | |----/ | |----/
| | setLocal(PRANSWER) | | | | setLocal(PRANSWER) | |
| 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) | | | | setRemote(PRANSWER) | |
| 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
In the WebRTC specification, session descriptions are formatted as In the WebRTC specification, session descriptions are formatted as
SDP messages. While this format is not optimal for manipulation from SDP messages. While this format is not optimal for manipulation from
Javascript, it is widely accepted, and frequently updated with new Javascript, it is widely accepted, and frequently updated with new
features. Any alternate encoding of session descriptions would have features. Any alternate encoding of session descriptions would have
to keep pace with the changes to SDP, at least until the time that to keep pace with the changes to SDP, at least until the time that
skipping to change at page 10, line 31 skipping to change at page 10, line 26
to generate and consume that JSON. to generate and consume that JSON.
Other methods may be added to SessionDescription in the future to Other methods may be added to SessionDescription in the future to
simplify handling of SessionDescriptions from Javascript. In the simplify handling of SessionDescriptions from Javascript. In the
meantime, Javascript libraries can be used to perform these meantime, Javascript libraries can be used to perform these
manipulations. manipulations.
Note that most applications should be able to treat the Note that 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. The W3C API will provide appropriate APIs to allow the change them. The W3C WebRTC API specification will provide
application to control various session parameters, which will provide appropriate APIs to allow the application to control various session
the necessary information to the browser about what sort of parameters, which will provide the necessary information to the
SessionDescription to produce. browser about what sort of SessionDescription to produce.
3.4. ICE 3.4. ICE
When a new ICE candidate is available, the ICE Agent will notify the When a new ICE candidate is available, the ICE Agent will notify the
application via a callback; these candidates will automatically be application via a callback; these candidates will automatically be
added to the local session description. When all candidates have added to the local session description. When all candidates have
been gathered, the callback will also be invoked to signal that the been gathered, the callback will also be invoked to signal that the
gathering process is complete. gathering process is complete.
3.4.1. ICE Candidate Trickling 3.4.1. 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-mmusic-trickle-ice]. This process allows the callee to in [I-D.ietf-mmusic-trickle-ice]. This process allows the callee to
begin acting upon the call and setting up the ICE (and perhaps DTLS) begin 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
gather all possible candidates. This results in faster call startup gather all possible candidates. This results in faster media setup
in cases where gathering is not performed prior to initiating the in cases where gathering is not performed prior to initiating the
call. call.
JSEP supports optional candidate trickling by providing APIs that JSEP supports optional candidate trickling by providing APIs that
provide control and feedback on the ICE candidate gathering process. provide control and feedback on the ICE candidate gathering process.
Applications that support candidate trickling can send the initial Applications that support candidate trickling can send the initial
offer immediately and send individual candidates when they get the offer immediately and send individual candidates when they get the
notified of a new candidate; applications that do not support this notified of a new candidate; applications that do not support this
feature can simply wait for the indication that gathering is feature can simply wait for the indication that gathering is
complete, and then create and send their offer, with all the complete, and then create and send their offer, with all the
skipping to change at page 11, line 28 skipping to change at page 11, line 24
using the new remote candidates for connectivity checks. using the new remote candidates for connectivity checks.
3.4.1.1. ICE Candidate Format 3.4.1.1. ICE Candidate Format
As with session descriptions, the syntax of the IceCandidate object As with session descriptions, the syntax of the IceCandidate object
provides some abstraction, but can be easily converted to and from provides some abstraction, but can be easily converted to and from
the SDP candidate lines. the SDP candidate lines.
The candidate lines are the only SDP information that is contained The candidate lines are the only SDP information that is contained
within IceCandidate, as they represent the only information needed within IceCandidate, as they represent the only information needed
that is not present in the initial offer (i.e. for trickle that is not present in the initial offer (i.e., for trickle
candidates). This information is carried with the same syntax as the candidates). This information is carried with the same syntax as the
"candidate-attribute" field defined for ICE. For example: "candidate-attribute" field defined for ICE. For 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 also contains fields to indicate which m= The IceCandidate object also contains fields to indicate which m=
line it should be associated with. The m line can be identified in line it should be associated with. The m line can be identified in
one of two ways; either by a m-line index, or a MID. The m-line one of two ways; either by a m-line index, or a MID. The m-line
index is a zero-based index, referring to the Nth m-line in the SDP. index is a zero-based index, with index N referring to the N+1th
m-line in the SDP sent by the entity which sent the IceCandidate.
The MID uses the "media stream identification", as defined in The MID uses the "media stream identification", as defined in
[RFC5888] , to identify the m-line. WebRTC implementations creating [RFC5888], to identify the m-line. WebRTC implementations creating
an ICE Candidate object MUST populate both of these fields. an ICE Candidate object MUST populate both of these fields.
Implementations receiving an ICE Candidate object SHOULD use the MID Implementations receiving an ICE Candidate object SHOULD use the MID
if they implement that functionality, or the m-line index, if not. if they implement that functionality, or the m-line index, if not.
3.4.2. ICE Candidate Pool 3.4.2. ICE Candidate Pool
JSEP applications typically inform the browser to begin ICE gathering JSEP applications typically inform the browser to begin ICE gathering
via the information supplied to setLocalDescription, as this is where via the information supplied to setLocalDescription, as this is where
the app specifies the number of media streams to gather candidates the app specifies the number of media streams for which to gather
for. However, to accelerate cases where the browser knows the number candidates. However, to accelerate cases where the application knows
of media streams to use ahead of time, the application MAY ask the the number of media streams to use ahead of time, it MAY ask the
browser to gather a pool of potential ICE candidates to help ensure browser to gather a pool of potential ICE candidates to help ensure
rapid media setup. When setLocalDescription is eventually called, rapid media setup. When setLocalDescription is eventually called,
and the browser goes to gather the needed ICE candidates, it can and the browser 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 can be handed to the there are candidates in the pool, they SHOULD be handed to the
application immediately via the aforementioned candidate callback. application immediately via the ICE candidate callback. If the pool
If the pool becomes depleted, either because a larger than expected becomes depleted, either because a larger-than-expected number of
number of candidates is needed, or because the pool has not had media streams is used, or because the pool has not had enough time to
enough time to gather candidates, the remaining candidates are gather candidates, the remaining candidates are gathered as usual.
gathered as usual.
3.5. Interactions With Forking 3.5. 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 which 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 to a given remote peer, but keep all the
Start exchanging media to a given remote peer, but keep all the
resources reserved in the offer. resources reserved in the offer.
o Start exchanging media with a given remote peer, and free any
Start exchanging media with a given remote peer, and free any
resources in the offer that are not being used. resources in the offer that are not being used.
3.5.1. Sequential Forking 3.5.1. Sequential Forking
Sequential forking involves a call being dispatched to multiple Sequential forking involves a call being dispatched to multiple
remote callees, where each callee can accept the call, but only one remote callees, where each callee can accept the call, but only one
active session ever exists at a time; no mixing of received media is active session ever exists at a time; no mixing of received media is
performed. performed.
JSEP handles sequential forking well, allowing the application to JSEP handles sequential forking well, allowing the application to
skipping to change at page 13, line 35 skipping to change at page 13, line 26
example, consider having a European ringback tone mixed together with example, consider having a European ringback tone mixed together with
the North American ringback tone - the resulting sound would not be the North American ringback tone - the resulting sound would not be
like either tone, and would confuse the user. If the signaling 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. (It is worth noting strategy that [RFC3960] describes using UPDATE. The UPDATE approach
that use cases where this is the desired behavior are very unusual.) allows the signaling to set up a separate media flow for each peer
The UPDATE approach allows the signaling to set up a separate media that it wishes to exchange media with. In JSEP, this offer used in
flow for each peer that it wishes to exchange media with. In JSEP, the UPDATE would be formed by simply creating a new PeerConnection
this offer used in the UPDATE would be formed by simply creating a and making sure that the same local media streams have been added
new PeerConnection and making sure that the same local media streams into this new PeerConnection. Then the new PeerConnection object
have been added into this new PeerConnection. Then the new would produce a SDP offer that could be used by the signaling to
PeerConnection object would produce a SDP offer that could be used by perform the UPDATE strategy discussed in [RFC3960].
the signaling to perform the UPDATE strategy discussed in [RFC3960].
As a result of sharing the media streams, the application will end up As a result of sharing the media streams, the application will end up
with N parallel PeerConnection sessions, each with a local and remote with N parallel PeerConnection sessions, each with a local and remote
description and their own local and remote addresses. The media flow description and their own local and remote addresses. The media flow
from these sessions can be managed by specifying SDP direction from these sessions can be managed by specifying SDP direction
attributes in the descriptions, or the application can choose to play attributes in the descriptions, or the application can choose to play
out the media from all sessions mixed together. Of course, if the out the media from all sessions mixed together. Of course, if the
application wants to only keep a single session, it can simply application wants to only keep a single session, it can simply
terminate the sessions that it no longer needs. terminate the sessions that it no longer needs.
3.6. Session Rehydration
In the event that the local application state is reinitialized,
either due to a user reload of the page, or a decision within the
application to reload itself (perhaps to update to a new version), it
is possible to keep an existing session alive, via a process called
"rehydration". The explicit goal of rehydration is to carry out this
session resumption with no interaction with the remote side other
than normal call signaling messages.
With rehydration, the current signaling state is persisted somewhere
outside of the page, perhaps on the application server, or in browser
local storage. The page is then reloaded, the saved signaling state
is retrieved, and a new PeerConnection object is created for the
session. The previously obtained MediaStreams are re-acquired, and
are given the same IDs as the original session; this ensures the IDs
in use by the remote side continue to work. Next, a new offer is
generated by the new PeerConnection; this offer will have new ICE and
possibly new DTLS-SRTP certificate fingerprints (since the old ICE
and SRTP state has been lost). Finally, this offer is used to re-
initiate the session with the existing remote endpoint, who simply
sees the new offer as an in-call renegotiation, and replies with an
answer that can be supplied to setRemoteDescription. ICE processing
proceeds as usual, and as soon as connectivity is established, the
session will be back up and running again.
[OPEN ISSUE: EKR proposed an alternative rehydration approach where
the actual internal PeerConnection object in the browser was kept
alive for some time after the web page was killed and provided some
way for a new page to acquire the old PeerConnection object.]
4. Interface 4. Interface
This section details the basic operations that must be present to This section details the basic operations that must be present to
implement JSEP functionality. The actual API exposed in the W3C API implement JSEP functionality. The actual API exposed in the W3C API
may have somewhat different syntax, but should map easily to these may have somewhat different syntax, but should map easily to these
concepts. concepts.
4.1. Methods 4.1. Methods
4.1.1. Constructor 4.1.1. Constructor
The PeerConnection constructor allows the application to specify The PeerConnection constructor allows the application to specify
global parameters for the media session, such as the STUN/TURN global parameters for the media session, such as the STUN/TURN
servers and credentials to use when gathering candidates. The size servers and credentials to use when gathering candidates. The size
of the ICE candidate pool can also be set, if desired; by default the of the ICE candidate pool can also be set, if desired; this indicates
candidate pool size is zero. the number of ICE components to pre-gather candidates for. If the
application does not indicate a candidate pool size, the browser may
select any default candidate pool size.
In addition, the application can specify its preferred policy In addition, the application can specify its preferred policy
regarding use of BUNDLE, the multiplexing mechanism defined in regarding use of BUNDLE, the multiplexing mechanism defined in
[I-D.ietf-mmusic-sdp-bundle-negotiation]. By specifying a policy [I-D.ietf-mmusic-sdp-bundle-negotiation]. By specifying a policy
from the list below, the application can control how aggressively it from the list below, the application can control how aggressively it
will try to BUNDLE media streams together. The set of available will try to BUNDLE media streams together. The set of available
policies is as follows: policies is as follows:
balanced: The application will BUNDLE all media streams of the same
o "default": The application will BUNDLE all media streams of the type together. That is, if there are multiple audio and multiple
same type together. That is, if there are multiple audio and video MediaStreamTracks attached to a PeerConnection, all but the
multiple video MediaStreamTracks attached to a PeerConnection, all first audio and video tracks will be marked as bundle-only, and
but the first audio and video tracks will be marked as bundle- candidates will only be gathered for N media streams, where N is
only, and candidates will only be gathered for N media streams, the number of distinct media types. When talking to a non-BUNDLE-
where N is the number of distinct media types. When talking to a aware endpoint, only the non-bundle-only streams will be
non-BUNDLE-aware endpoint, only the non-bundle-only streams will negotiated. This policy balances desire to multiplex with the
be negotiated. This policy balances desire to multiplex with the
need to ensure basic audio and video still works in legacy cases. need to ensure basic audio and video still works in legacy cases.
Data channels will be in a separate bundle group.
o "max-bundle": The application will BUNDLE all of its media streams max-bundle: The application will BUNDLE all of its media streams,
on a single transport. All streams other than the first will be including data channels, on a single transport. All streams other
marked as bundle-only. This policy aims to minimize candidate than the first will be marked as bundle-only. This policy aims to
gathering and maximize multiplexing, at the cost of less minimize candidate gathering and maximize multiplexing, at the
compatibility with legacy endpoints. cost of less compatibility with legacy endpoints.
o "max-compat": The application will offer BUNDLE, but mark none of max-compat: The application will offer BUNDLE, but mark none of its
its streams as bundle-only. This policy will allow all streams to streams as bundle-only. This policy will allow all streams to be
be received by non-BUNDLE-aware endpoints, but require separate received by non-BUNDLE-aware endpoints, but require separate
candidates to be gathered for each media stream. candidates to be gathered for each media stream.
4.1.2. createOffer 4.1.2. createOffer
The createOffer method generates a blob of SDP that contains a The createOffer method generates a blob of SDP that contains a
[RFC3264] offer with the supported configurations for the session, [RFC3264] offer with the supported configurations for the session,
including descriptions of the local MediaStreams attached to this including descriptions of the local MediaStreams attached to this
PeerConnection, the codec/RTP/RTCP options supported by this PeerConnection, the codec/RTP/RTCP options supported by this
implementation, and any candidates that have been gathered by the ICE implementation, and any candidates that have been gathered by the ICE
Agent. A constraints parameter may be supplied to provide additional Agent. An options parameter may be supplied to provide additional
control over the generated offer. This constraints parameter should control over the generated offer. This options parameter should
allow for the following manipulations to be performed: allow for the following manipulations to be performed:
o To indicate support for a media type even if no MediaStreamTracks o To indicate support for a media type even if no MediaStreamTracks
of that type have been added to the session (e.g., an audio call of that type have been added to the session (e.g., an audio call
that wants to receive video.) that wants to receive video.)
o To trigger an ICE restart, for the purpose of reestablishing o To trigger an ICE restart, for the purpose of reestablishing
connectivity. connectivity.
In the initial offer, the generated SDP will contain all desired In the initial offer, the generated SDP will contain all desired
functionality for the session (certain parts that are supported but functionality for the session (functionality that is supported but
not desired by default may be omitted); for each SDP line, the not desired by default may be omitted); for each SDP line, the
generation of the SDP will follow the process defined for generating generation of the SDP will follow the process defined for generating
an initial offer from the document that specifies the given SDP line. an initial offer from the document that specifies the given SDP line.
The exact handling of initial offer generation is detailed in The exact handling of initial offer generation is detailed in
Section 5.2.1. below. Section 5.2.1 below.
In the event createOffer is called after the session is established, In the event createOffer is called after the session is established,
createOffer will generate an offer to modify the current session createOffer will generate an offer to modify the current session
based on any changes that have been made to the session, e.g. adding based on any changes that have been made to the session, e.g. adding
or removing MediaStreams, or requesting an ICE restart. For each or removing MediaStreams, or requesting an ICE restart. For each
existing stream, the generation of each SDP line must follow the existing stream, the generation of each SDP line must follow the
process defined for generating an updated offer from the document process defined for generating an updated offer from the RFC that
that specifies the given SDP line. For each new stream, the specifies the given SDP line. For each new stream, the generation of
generation of the SDP must follow the process of generating an the SDP must follow the process of generating an initial offer, as
initial offer, as mentioned above. If no changes have been made, or mentioned above. If no changes have been made, or for SDP lines that
for SDP lines that are unaffected by the requested changes, the offer are unaffected by the requested changes, the offer will only contain
will only contain the parameters negotiated by the last offer-answer the parameters negotiated by the last offer-answer exchange. The
exchange. The exact handling of subsequent offer generation is exact handling of subsequent offer generation is detailed in
detailed in Section 5.2.2. below. Section 5.2.2. below.
Session descriptions generated by createOffer must be immediately Session descriptions generated by createOffer must be immediately
usable by setLocalDescription; if a system has limited resources usable by setLocalDescription; if a system has limited resources
(e.g. a finite number of decoders), createOffer should return an (e.g. a finite number of decoders), createOffer should return an
offer that reflects the current state of the system, so that offer that reflects the current state of the system, so that
setLocalDescription will succeed when it attempts to acquire those setLocalDescription will succeed when it attempts to acquire those
resources. Because this method may need to inspect the system state resources. Because this method may need to inspect the system state
to determine the currently available resources, it may be implemented to determine the currently available resources, it may be implemented
as an async operation. as an async operation.
Calling this method may do things such as generate new ICE Calling this method may do things such as generate new ICE
credentials, but does not result in candidate gathering, or cause credentials, but does not result in candidate gathering, or cause
media to start or stop flowing. media to start or stop flowing.
4.1.3. createAnswer 4.1.3. createAnswer
The createAnswer method generates a blob of SDP that contains a The createAnswer method generates a blob of SDP that contains a
[RFC3264] SDP answer with the supported configuration for the session [RFC3264] SDP answer with the supported configuration for the session
that is compatible with the parameters supplied in the offer. Like that is compatible with the parameters supplied in the most recent
createOffer, the returned blob contains descriptions of the local call to setRemoteDescription, which MUST have been called prior to
MediaStreams attached to this PeerConnection, the codec/RTP/RTCP calling createAnswer. Like createOffer, the returned blob contains
options negotiated for this session, and any candidates that have descriptions of the local MediaStreams attached to this
been gathered by the ICE Agent. A constraints parameter may be PeerConnection, the codec/RTP/RTCP options negotiated for this
supplied to provide additional control over the generated answer. session, and any candidates that have been gathered by the ICE Agent.
An options parameter may be supplied to provide additional control
over the generated answer.
As an answer, the generated SDP will contain a specific configuration As an answer, the generated SDP will contain a specific configuration
that specifies how the media plane should be established; for each that specifies how the media plane should be established; for each
SDP line, the generation of the SDP must follow the process defined SDP line, the generation of the SDP must follow the process defined
for generating an answer from the document that specifies the given for generating an answer from the document that specifies the given
SDP line. The exact handling of answer generation is detailed in SDP line. The exact handling of answer generation is detailed in
Section 5.3. below. Section 5.3. below.
Session descriptions generated by createAnswer must be immediately Session descriptions generated by createAnswer must be immediately
usable by setLocalDescription; like createOffer, the returned usable by setLocalDescription; like createOffer, the returned
skipping to change at page 18, line 8 skipping to change at page 17, line 21
were allocated as a result of the offer. As such, the application were allocated as a result of the offer. As such, the application
can use some discretion on whether an answer should be applied as can use some discretion on whether an answer should be applied as
provisional or final, and can change the type of the session provisional or final, and can change the type of the session
description as needed. For example, in a serial forking scenario, an description as needed. For example, in a serial forking scenario, an
application may receive multiple "final" answers, one from each application may receive multiple "final" answers, one from each
remote endpoint. The application could choose to accept the initial remote endpoint. The application could choose to accept the initial
answers as provisional answers, and only apply an answer as final answers as provisional answers, and only apply an answer as final
when it receives one that meets its criteria (e.g. a live user when it receives one that meets its criteria (e.g. a live user
instead of voicemail). instead of voicemail).
"rollback" is a special session description type implying that the
state machine should be rolled back to the previous state, as
described in Section 4.1.4.2. The contents MUST be empty.
4.1.4.1. Use of Provisional Answers 4.1.4.1. Use of Provisional Answers
Most web applications will not need to create answers using the Most web applications will not need to create answers using the
"pranswer" type. While it is good practice to send an immediate "pranswer" type. While it is good practice to send an immediate
response to an "offer", in order to warm up the session transport and response to an "offer", in order to warm up the session transport and
prevent media clipping, the preferred handling for a web application prevent media clipping, the preferred handling for a web application
would be to create and send an "inactive" final answer immediately would be to create and send an "inactive" final answer immediately
after receiving the offer. Later, when the called user actually after receiving the offer. Later, when the called user actually
accepts the call, the application can create a new "sendrecv" offer accepts the call, the application can create a new "sendrecv" offer
to update the previous offer/answer pair and start the media flow. to update the previous offer/answer pair and start the media flow.
While this could also be done with an inactive "pranswer", followed While this could also be done with an inactive "pranswer", followed
by a sendrecv "answer", the initial "pranswer" leaves the offer- by a sendrecv "answer", the initial "pranswer" leaves the offer-
answer exchange open, which means the caller cannot send an updated answer exchange open, which means that neither side can send an
offer during this time. updated offer during this time.
As an example, consider a typical web application that will set up a As an example, consider a typical web application that will set up a
data channel, an audio channel, and a video channel. When an data channel, an audio channel, and a video channel. When an
endpoint receives an offer with these channels, it could send an endpoint receives an offer with these channels, it could send an
answer accepting the data channel for two-way data, and accepting the answer accepting the data channel for two-way data, and accepting the
audio and video tracks as inactive or receive-only. It could then audio and video tracks as inactive or receive-only. It could then
ask the user to accept the call, acquire the local media streams, and ask the user to accept the call, acquire the local media streams, and
send a new offer to the remote side moving the audio and video to be send a new offer to the remote side moving the audio and video to be
two-way media. By the time the human has accepted the call and two-way media. By the time the human has accepted the call and
triggered the new offer, it is likely that the ICE and DTLS triggered the new offer, it is likely that the ICE and DTLS
handshaking for all the channels will already be set up. handshaking for all the channels will already have finished.
Of course, some applications may not be able to perform this double Of course, some applications may not be able to perform this double
offer-answer exchange, particularly ones that are attempting to offer-answer exchange, particularly ones that are attempting to
gateway to legacy signaling protocols. In these cases, "pranswer" gateway to legacy signaling protocols. In these cases, "pranswer"
can still provide the application with a mechanism to warm up the can still provide the application with a mechanism to warm up the
transport. transport.
4.1.4.2. Rollback 4.1.4.2. Rollback
In certain situations it may be desirable to "undo" a change made to In certain situations it may be desirable to "undo" a change made to
skipping to change at page 19, line 13 skipping to change at page 18, line 30
this, we introduce the concept of "rollback". this, we introduce the concept of "rollback".
A rollback discards any proposed changes to the session, returning A rollback discards any proposed changes to the session, returning
the state machine to the stable state, and setting the modified local the state machine to the stable state, and setting the modified local
and/or remote description back to their previous values. Any and/or remote description back to their previous values. Any
resources or candidates that were allocated by the abandoned local resources or candidates that were allocated by the abandoned local
description are discarded; any media that is received will be description are discarded; any media that is received will be
processed according to the previous local and remote descriptions. processed according to the previous local and remote descriptions.
Rollback can only be used to cancel proposed changes; there is no Rollback can only be used to cancel proposed changes; there is no
support for rolling back from a stable state to a previous stable support for rolling back from a stable state to a previous stable
state. state. Note that this implies that once the answerer has performed
setLocalDescription with his answer, this cannot be rolled back.
A rollback is performed by supplying a session description of type A rollback is performed by supplying a session description of type
"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).
4.1.5. setLocalDescription 4.1.5. setLocalDescription
The setLocalDescription method instructs the PeerConnection to apply The setLocalDescription method instructs the PeerConnection to apply
skipping to change at page 20, line 27 skipping to change at page 19, line 42
and the media directions are compatible, and media are available to and the media directions are compatible, and media are available to
send, this will result in the starting of media transmission. send, this will result in the starting of media transmission.
4.1.7. localDescription 4.1.7. localDescription
The localDescription method returns a copy of the current local The localDescription method returns a copy of the current local
configuration, i.e. what was most recently passed to configuration, i.e. what was most recently passed to
setLocalDescription, plus any local candidates that have been setLocalDescription, plus any local candidates that have been
generated by the ICE Agent. generated by the ICE Agent.
TODO: Do we need to expose accessors for both the current and [[OPEN ISSUE: Do we need to expose accessors for both the current
proposed local description? and proposed local description?
https://github.com/rtcweb-wg/jsep/issues/16]]
A null object will be returned if the local description has not yet A null object will be returned if the local description has not yet
been established, or if the PeerConnection has been closed. been established, or if the PeerConnection has been closed.
4.1.8. remoteDescription 4.1.8. remoteDescription
The remoteDescription method returns a copy of the current remote The remoteDescription method returns a copy of the current remote
configuration, i.e. what was most recently passed to configuration, i.e. what was most recently passed to
setRemoteDescription, plus any remote candidates that have been setRemoteDescription, plus any remote candidates that have been
supplied via processIceMessage. supplied via processIceMessage.
TODO: Do we need to expose accessors for both the current and [[OPEN ISSUE: Do we need to expose accessors for both the current
proposed remote description? and proposed remote description?
https://github.com/rtcweb-wg/jsep/issues/16]]
A null object will be returned if the remote description has not yet A null object will be returned if the remote description has not yet
been established, or if the PeerConnection has been closed. been established, or if the PeerConnection has been closed.
4.1.9. updateIce 4.1.9. updateIce
The updateIce method allows the configuration of the ICE Agent to be The updateIce method allows the configuration of the ICE Agent to be
changed during the session, primarily for changing which types of changed during the session, primarily for changing which types of
local candidates are provided to the application and used for local candidates are provided to the application and used for
connectivity checks. A callee may initially configure the ICE Agent connectivity checks. A callee may initially configure the ICE Agent
skipping to change at page 23, line 10 skipping to change at page 22, line 23
media bit-rate boundaries. media bit-rate boundaries.
As required by [RFC4566], Section 5.13, JSEP implementations MUST As required by [RFC4566], Section 5.13, JSEP implementations MUST
ignore unknown attribute (a=) lines. ignore unknown attribute (a=) lines.
5.1.2. Usage Requirements 5.1.2. Usage Requirements
All session descriptions handled by JSEP endpoints, both local and All session descriptions handled by JSEP endpoints, both local and
remote, MUST indicate support for the following specifications. If remote, MUST indicate support for the following specifications. If
any of these are absent, this omission MUST be treated as an error. any of these are absent, this omission MUST be treated as an error.
R-1 Either the UDP/TLS/RTP/SAVP or the UDP/TLS/RTP/SAVPF RTP R-1 Either the UDP/TLS/RTP/SAVP or the UDP/TLS/RTP/SAVPF RTP
profile, as specified in [RFC5764], MUST be used. profile, as specified in [RFC5764], MUST be used.
R-2 ICE, as specified in [RFC5245], MUST be used. Note that the R-2 ICE, as specified in [RFC5245], MUST be used. Note that the
remote endpoint MAY use a Lite implementation. remote endpoint may use a Lite implementation; implementations
MUST properly handle remote endpoints which do ICE-Lite.
R-3 DTLS-SRTP, as specified in [RFC5763], MUST be used. R-3 DTLS-SRTP, as specified in [RFC5763], MUST be used.
5.2. Constructing an Offer 5.2. Constructing an Offer
When createOffer is called, a new SDP description must be created When createOffer is called, a new SDP description must be created
that includes the functionality specified in that includes the functionality specified in
[I-D.ietf-rtcweb-rtp-usage]. The exact details of this process are [I-D.ietf-rtcweb-rtp-usage]. The exact details of this process are
explained below. explained below.
5.2.1. Initial Offers 5.2.1. Initial Offers
skipping to change at page 23, line 49 skipping to change at page 23, line 9
[RFC4566], Section 5.2. The value of the <username> field SHOULD [RFC4566], Section 5.2. The value of the <username> field SHOULD
be "-". The value of the <sess-id> field SHOULD be a be "-". The value of the <sess-id> field SHOULD be a
cryptographically random number. To ensure uniqueness, this cryptographically random number. To ensure uniqueness, this
number SHOULD be at least 64 bits long. The value of the <sess- number SHOULD be at least 64 bits long. The value of the <sess-
version> field SHOULD be zero. The value of the <nettype> version> field SHOULD be zero. The value of the <nettype>
<addrtype> <unicast-address> tuple SHOULD be set to a non- <addrtype> <unicast-address> tuple SHOULD be set to a non-
meaningful address, such as IN IP4 0.0.0.0, to prevent leaking the meaningful address, such as IN IP4 0.0.0.0, to prevent leaking the
local address in this field. As mentioned in [RFC4566], the local address in this field. As mentioned in [RFC4566], the
entire o= line needs to be unique, but selecting a random number entire o= line needs to be unique, but selecting a random number
for <sess-id> is sufficient to accomplish this. 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=-". as the session name, e.g. "s=-". Note that this differs from the
advice in [RFC4566] which proposes a single space, but as both
"o=" and "s=" are meaningless, having the same meaningless value
seems clearer.
o Session Information ("i="), URI ("u="), Email Address ("e="), o Session Information ("i="), URI ("u="), Email Address ("e="),
Phone Number ("p="), Bandwidth ("b="), Repeat Times ("r="), and Phone Number ("p="), Bandwidth ("b="), Repeat Times ("r="), and
Time Zones ("z=") lines are not useful in this context and SHOULD Time Zones ("z=") lines are not useful in this context and SHOULD
NOT be included. 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=msid-semantic:WMS" line MUST be added, as specified in o An "a=msid-semantic:WMS" line MUST be added, as specified in
[I-D.ietf-mmusic-msid], Section 4. [I-D.ietf-mmusic-msid], Section 4.
The next step is to generate m= sections for each MediaStreamTrack The next step is to generate m= sections, as specified in [RFC4566]
that has been added to the PeerConnection via the addStream method. Section 5.14, for each MediaStreamTrack that has been added to the
(Note that this method takes a MediaStream, which can contain PeerConnection via the addStream method. (Note that this method
multiple MediaStreamTracks, and therefore multiple m= sections can be takes a MediaStream, which can contain multiple MediaStreamTracks,
generated even if addStream is only called once.) When generating m= and therefore multiple m= sections can be generated even if addStream
sections, the ordering is based on (1) the order in which the is only called once.) m=sections MUST be sorted first by the order in
MediaStreams were added to the PeerConnection, and (2) the which the MediaStreams were added to the PeerConnection, and then by
alphabetical ordering of the media type for the MediaStreamTrack. the alphabetical ordering of the media type for the MediaStreamTrack.
For example, if a MediaStream containing both an audio and a video For example, if a MediaStream containing both an audio and a video
MediaStreamTrack is added to a PeerConnection, the resultant m=audio MediaStreamTrack is added to a PeerConnection, the resultant m=audio
section will precede the m=video section. section will precede the m=video section. If a second MediaStream
containing an audio MediaStreamTrack was added, it would follow the
m=video section.
Each m= section, provided it is not being bundled into another m= Each m= section, provided it is not being bundled into another m=
section, MUST generate a unique set of ICE credentials and gather its section, MUST generate a unique set of ICE credentials and gather its
own unique set of ICE candidates. Otherwise, it MUST use the same own unique set of ICE candidates. Otherwise, it MUST use the same
ICE credentials and candidates that were used in the m= section that ICE credentials and candidates as the m= section into which it is
it is being bundled into. being bundled. Note that this means that for offers, any m= sections
which are not bundle-only MUST have unique ICE credentials and
candidates, since it is possible that the answerer will accept them
without bundling them.
For DTLS, all m= sections MUST use the certificate for the identity For DTLS, all m= sections MUST use the certificate for the identity
that has been specified for the PeerConnection; as a result, they that has been specified for the PeerConnection; as a result, they
MUST all have the same [RFC4572]fingerprint value. MUST all have the same [RFC4572] fingerprint value, or this value
MUST be a session-level attribute.
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 The port value is set to the port of the default ICE candidate for
this m= section; if this m= section is not being bundled into this m= section; if this m= section is not being bundled into
another m= section, the port value MUST be unique. If no another m= section, the port value MUST be unique. If no
candidates have yet been gathered, and a 'null' port value is candidates have yet been gathered, and a 'null' port value is
being used, as indicated in [I-D.ietf-mmusic-trickle-ice], being used, as indicated in [I-D.ietf-mmusic-trickle-ice], Section
Section 5.1., this port MUST still be unique. 5.1., this port MUST still be unique.
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.
Each m= section MUST include the following attribute lines: Each m= section MUST include the following attribute lines:
o An "a=mid" line, as specified in [RFC5888], Section 4. o An "a=mid" line, as specified in [RFC5888], Section 4.
o An "a=msid" line, as specified in [I-D.ietf-mmusic-msid], Section
o An "a=msid" line, as specified in [I-D.ietf-mmusic-msid], 2.
Section 2. o [OPEN ISSUE: Use of AppID]
o [OPEN ISSUE: Use of AppID]
o An "a=sendrecv" line, as specified in [RFC3264], Section 5.1. o An "a=sendrecv" line, as specified in [RFC3264], Section 5.1.
o For each supported codec, "a=rtpmap" and "a=fmtp" lines, as o For each supported codec, "a=rtpmap" and "a=fmtp" lines, as
specified in [RFC4566], Section 6. For audio, the codecs specified in [RFC4566], Section 6. For audio, the codecs
specified in [I-D.ietf-rtcweb-audio], Section 3, MUST be be specified in [I-D.ietf-rtcweb-audio], Section 3, MUST be be
supported. supported.
o For each primary codec where RTP retransmission should be used, a o For each primary codec where RTP retransmission should be used, a
corresponding "a=rtpmap" line indicating "rtx" with the clock rate corresponding "a=rtpmap" line indicating "rtx" with the clock rate
of the primary codec and an "a=fmtp" line that references the of the primary codec and an "a=fmtp" line that references the
payload type fo the primary codec, as specified in [RFC4588], payload type of the primary codec, as specified in [RFC4588],
Section 8.1. Section 8.1.
o For each supported FEC mechanism, a corresponding "a=rtpmap" line o For each supported FEC mechanism, a corresponding "a=rtpmap" line
indicating the desired FEC codec. indicating the desired FEC codec.
o "a=ice-ufrag" and "a=ice-passwd" lines, as specified in [RFC5245], o "a=ice-ufrag" and "a=ice-passwd" lines, as specified in [RFC5245],
Section 15.4. Section 15.4.
o An "a=ice-options" line, with the "trickle" option, as specified o An "a=ice-options" line, with the "trickle" option, as specified
in [I-D.ietf-mmusic-trickle-ice], Section 4. in [I-D.ietf-mmusic-trickle-ice], Section 4.
o For each candidate that has been gathered during the most recent o For each candidate that has been gathered during the most recent
gathering phase, an "a=candidate" line, as specified in [RFC5245], gathering phase, an "a=candidate" line, as specified in [RFC5245],
Section 4.3., paragraph 3. Section 4.3., paragraph 3.
o For the current default candidate, a "c=" line, as specified in
o For the current default candidate, a "c=" line, as specific in
[RFC5245], Section 4.3., paragraph 6. If no candidates have been [RFC5245], Section 4.3., paragraph 6. If no candidates have been
gathered yet, the default candidate should be set to the 'null' gathered yet, the default candidate should be set to the 'null'
value defined in [I-D.ietf-mmusic-trickle-ice], Section 5.1. value defined in [I-D.ietf-mmusic-trickle-ice], Section 5.1.
o An "a=fingerprint" line, as specified in [RFC4572], Section 5; the o An "a=fingerprint" line, as specified in [RFC4572], Section 5; the
algorithm used for the fingerprint MUST match that used in the algorithm used for the fingerprint MUST match that used in the
certificate signature. certificate 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=rtcp-mux" line, as specified in [RFC5761], Section 5.1.1. o An "a=rtcp-mux" line, as specified in [RFC5761], Section 5.1.1.
o An "a=rtcp-rsize" line, as specified in [RFC5506], Section 5. o An "a=rtcp-rsize" line, as specified in [RFC5506], Section 5.
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"
mechanism, as specified in [RFC4585], Section 4.2. The list of mechanism, as specified in [RFC4585], Section 4.2. The list of
RTCP feedback mechanisms that SHOULD/MUST be supported is RTCP feedback mechanisms that SHOULD/MUST be supported is
specified in [I-D.ietf-rtcweb-rtp-usage], Section 5.1. specified in [I-D.ietf-rtcweb-rtp-usage], Section 5.1.
o An "a=ssrc" line, as specified in [RFC5576], Section 4.1, o An "a=ssrc" line, as specified in [RFC5576], Section 4.1,
indicating the SSRC to be used for sending media, along with the indicating the SSRC to be used for sending media, along with the
mandatory "cname" source attribute, as specified in Section 6.1, mandatory "cname" source attribute, as specified in Section 6.1,
indicating the CNAME for the source. The CNAME must be generated indicating the CNAME for the source. The CNAME must be generated
in accordance with draft-rescorla-random-cname-00. [OPEN ISSUE: in accordance with [RFC7022]. [OPEN ISSUE: How are CNAMEs
How are CNAMEs specified for MSTs? Are they randomly generated specified for MSTs? Are they randomly generated for each
for each MediaStream? If so, can two MediaStreams be synced?] MediaStream? If so, can two MediaStreams be synced? See:
https://github.com/rtcweb-wg/jsep/issues/4]
o If RTX is supported for this media type, another "a=ssrc" line o If RTX is supported for this media type, another "a=ssrc" line
with the RTX SSRC, and an "a=ssrc-group" line, as specified in with the RTX SSRC, and an "a=ssrc-group" line, as specified in
[RFC5576], section 4.2, with semantics set to "FID" and including [RFC5576], section 4.2, with semantics set to "FID" and including
the primary and RTX SSRCs. the primary and RTX SSRCs.
o If FEC is supported for this media type, another "a=ssrc" line o If FEC is supported for this media type, another "a=ssrc" line
with the FEC SSRC, and an "a=ssrc-group" line, as specified in with the FEC SSRC, and an "a=ssrc-group" line, as specified in
[RFC5576], section 4.2, with semantics set to "FEC" and including [RFC5576], section 4.2, with semantics set to "FEC" and including
the primary and FEC SSRCs. the primary and FEC SSRCs.
o [OPEN ISSUE: Handling of a=imageattr]
o [OPEN ISSUE: Handling of a=imageattr]
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 "default", and this is not the first m= section policy is set to "default", 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.
Lastly, if a data channel has been created, a m= section MUST be Lastly, if a data channel has been created, a m= section MUST be
generated for data. The <media> field MUST be set to "application" generated for data. The <media> field MUST be set to "application"
and the <proto> field MUST be set to "DTLS/SCTP", as specified in and the <proto> field MUST be set to "DTLS/SCTP", as specified in
[I-D.ietf-mmusic-sctp-sdp], Section 3; the "fmt" value MUST be set to [I-D.ietf-mmusic-sctp-sdp], Section 3; the "fmt" value MUST be set to
the SCTP port number, as specified in Section 4.1. the SCTP port number, as specified in Section 4.1.
Within the data m= section, the "a=mid", "a=ice-ufrag", "a=ice- Within the data m= section, the "a=mid", "a=ice-ufrag", "a=ice-
passwd", "a=ice-options", "a=candidate", "a=fingerprint", and passwd", "a=ice-options", "a=candidate", "a=fingerprint", and
"a=setup" lines MUST be included as mentioned above, along with an "a=setup" lines MUST be included as mentioned above, along with an
"a=sctpmap" line referencing the SCTP port number and specifying the "a=sctpmap" line referencing the SCTP port number and specifying the
application protocol indicated in [I-D.ietf-rtcweb-data-protocol]. application protocol indicated in [I-D.ietf-rtcweb-data-protocol].
[OPEN ISSUE: the -01 of this document is missing this information.]
[OPEN ISSUE: the -01 of this document is missing this information.]
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. each m= section. The effect of this is that the browser offers all
m= sections as one BUNDLE group. However, whether the m= sections
are bundle-only or not depends on the BUNDLE policy.
Attributes that are common between all m= sections MAY be moved to Attributes which SDP permits to either be at the session level or the
session-level, if explicitly defined to be valid at session-level. media level SHOULD generally be at the media level even if they are
identical. This promotes readability, especially if one of a set of
initially identical attributes is subsequently changed.
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"
o "a=ice-lite" o "a=ice-lite"
Note that when BUNDLE is used, any additional attributes that are Note that when BUNDLE is used, any additional attributes that are
added MUST follow the advice in added MUST follow the advice in [I-D.ietf-mmusic-sdp-mux-attributes]
[I-D.nandakumar-mmusic-sdp-mux-attributes] on how those attributes on how those attributes interact with BUNDLE.
interact with BUNDLE.
Note that these requirements are in some cases stricter than those of
SDP. Implementations MUST be prepared to accept compliant SDP even
if it would not conform to the requirements for generating SDP in
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 initial 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
skipping to change at page 28, line 12 skipping to change at page 27, line 7
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, including the addition of ICE description changes in any way, including the addition of ICE
candidates. candidates.
If the initial offer was applied using setLocalDescription, but an If the initial offer was applied using setLocalDescription, but an
answer from the remote side has not yet been applied, meaning the answer from the remote side has not yet been applied, meaning the
PeerConnection is still in the "local-offer" state, an offer is PeerConnection is still in the "local-offer" state, an offer is
generated by following the steps in the "stable" state above, along generated by following the steps in the "stable" state above, along
with these exceptions: with these exceptions:
o The "s=" and "t=" lines MUST stay the same. o The "s=" and "t=" lines MUST stay the same.
o Each "a=mid" line MUST stay the same. o Each "a=mid" line MUST stay the same.
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. the "IceRestart" option (Section 5.2.3 was specified. Note that
it's not clear why you would actually want to do this, since at
this point ICE has not yet started and is thus unlikely to need a
restart.
o For MediaStreamTracks that are still present, the "a=msid", o For MediaStreamTracks that are still present, the "a=msid",
"a=ssrc", and "a=ssrc-group" lines MUST stay the same. "a=ssrc", and "a=ssrc-group" lines MUST stay the same.
o If any MediaStreamTracks have been removed, either through the o If any MediaStreamTracks have been removed, either through the
removeStream method or by removing them from an added MediaStream, removeStream method or by removing them from an added MediaStream,
their m= sections MUST be marked as recvonly by changing the value their m= sections MUST be marked as recvonly by changing the value
of the [RFC3264] directional attribute to "a=recvonly". The of the [RFC3264] directional attribute to "a=recvonly". The
"a=msid", "a=ssrc", and "a=ssrc-group" lines MUST be removed from "a=msid", "a=ssrc", and "a=ssrc-group" lines MUST be removed from
the associated m= sections. the associated m= sections.
o If any MediaStreamTracks have been added, and there exist m= o If any MediaStreamTracks have been added, and there exist m=
sections of the appropriate media type with no associated sections of the appropriate media type with no associated
MediaStreamTracks (i.e. as described in the preceding paragraph), MediaStreamTracks (i.e. as described in the preceding paragraph),
those m= sections MUST be recycled by adding the new those m= sections MUST be recycled by adding the new
MediaStreamTrack to the m= section. This is done by adding the MediaStreamTrack to the m= section. This is done by adding the
necessary "a=msid", "a=ssrc", and "a=ssrc-group" lines to the necessary "a=msid", "a=ssrc", and "a=ssrc-group" lines to the
recycled m= section, and removing the "a=recvonly" attribute. recycled m= section, and removing the "a=recvonly" attribute.
If the initial offer was applied using setLocalDescription, and an If the initial offer was applied using setLocalDescription, and an
answer from the remote side has been applied using answer from the remote side has been applied using
skipping to change at page 28, line 42 skipping to change at page 27, line 35
those m= sections MUST be recycled by adding the new those m= sections MUST be recycled by adding the new
MediaStreamTrack to the m= section. This is done by adding the MediaStreamTrack to the m= section. This is done by adding the
necessary "a=msid", "a=ssrc", and "a=ssrc-group" lines to the necessary "a=msid", "a=ssrc", and "a=ssrc-group" lines to the
recycled m= section, and removing the "a=recvonly" attribute. recycled m= section, and removing the "a=recvonly" attribute.
If the initial offer was applied using setLocalDescription, and an If the initial offer was applied using setLocalDescription, and an
answer from the remote side has been applied using answer from the remote side has been applied using
setRemoteDescription, meaning the PeerConnection is in the "remote- setRemoteDescription, meaning the PeerConnection is in the "remote-
pranswer" or "stable" states, an offer is generated based on the pranswer" or "stable" states, an offer is generated based on the
negotiated session descriptions by following the steps mentioned for negotiated session descriptions by following the steps mentioned for
the "local-offer" state above, along with these exceptions: [OPEN the "local-offer" state above, along with these exceptions: [OPEN
ISSUE: should this be permitted in the remote-pranswer state?] ISSUE: should this be permitted in the remote-pranswer state?]
o If a m= section exists in the current local description, but does o If a m= section exists in the current local description, but does
not have an associated local MediaStreamTrack (possibly because not have an associated local MediaStreamTrack (possibly because
said MediaStreamTrack was removed since the last exchange), a m= said MediaStreamTrack was removed since the last exchange), a m=
section MUST still be generated in the new offer, as indicated in section MUST still be generated in the new offer, as indicated in
[RFC3264], Section 8. The disposition of this section will depend [RFC3264], Section 8. The disposition of this section will depend
on the state of the remote MediaStreamTrack associated with this on the state of the remote MediaStreamTrack associated with this
m= section. If one exists, and it is still in the "live" state, m= section. If one exists, and it is still in the "live" state,
the new m= section MUST be marked as "a=recvonly", with no the new m= section MUST be marked as "a=recvonly", with no
"a=msid" or related attributes present. If no remote "a=msid" or related attributes present. If no remote
MediaStreamTrack exists, or it is in the "ended" state, the m= MediaStreamTrack exists, or it is in the "ended" state, the m=
skipping to change at page 29, line 9 skipping to change at page 27, line 49
said MediaStreamTrack was removed since the last exchange), a m= said MediaStreamTrack was removed since the last exchange), a m=
section MUST still be generated in the new offer, as indicated in section MUST still be generated in the new offer, as indicated in
[RFC3264], Section 8. The disposition of this section will depend [RFC3264], Section 8. The disposition of this section will depend
on the state of the remote MediaStreamTrack associated with this on the state of the remote MediaStreamTrack associated with this
m= section. If one exists, and it is still in the "live" state, m= section. If one exists, and it is still in the "live" state,
the new m= section MUST be marked as "a=recvonly", with no the new m= section MUST be marked as "a=recvonly", with no
"a=msid" or related attributes present. If no remote "a=msid" or related attributes present. If no remote
MediaStreamTrack exists, or it is in the "ended" state, the m= MediaStreamTrack exists, or it is in the "ended" state, the m=
section MUST be marked as rejected, by setting the port to zero, section MUST be marked as rejected, by setting the port to zero,
as indicated in [RFC3264], Section 8.2. as indicated in [RFC3264], Section 8.2.
o If any MediaStreamTracks have been added, and there exist recvonly o If any MediaStreamTracks have been added, and there exist recvonly
m= sections of the appropriate media type with no associated m= sections of the appropriate media type with no associated
MediaStreamTracks, or rejected m= sections of any media type, MediaStreamTracks, or rejected m= sections of any media type,
those m= sections MUST be recycled, and the local those m= sections MUST be recycled, and a local MediaStreamTrack
MediaStreamTracks associated with these recycled m= sections. associated with these recycled m= sections until all such existing
This includes any recvonly or rejected m= sections created by the m= sections have been used. This includes any recvonly or
preceding paragraph. rejected m= sections created by the preceding paragraph.
In addition, for each non-recycled, non-rejected m= section in the In addition, for each non-recycled, non-rejected m= section in the
new offer, the following adjustments are made based on the contents new offer, the following adjustments are made based on the contents
of the corresponding m= section in the current remote description: of the corresponding m= section in the current remote description:
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 remote description. only include codecs present in the remote description.
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 remote description. in the remote description.
o The RTCP feedback extensions MUST only include those that are o The RTCP feedback extensions MUST only include those that are
present in the remote description. present in the remote description.
o The "a=rtcp-mux" line MUST only be added if present in the remote o The "a=rtcp-mux" line MUST only be added if present in the remote
description. description.
o The "a=rtcp-rsize" line MUST only be added if present in the o The "a=rtcp-rsize" line MUST only be added if present in the
remote description. remote description.
The "a=group:BUNDLE" attribute MUST include the mid identifiers The "a=group:BUNDLE" attribute MUST include the mid identifiers
specified in the BUNDLE group in the most recent answer, minus any m= specified in the BUNDLE group in the most recent answer, minus any m=
sections that have been marked as rejected, plus any newly added or sections that have been marked as rejected, plus any newly added or
re-enabled m= sections. In other words, the BUNDLE attribute must re-enabled m= sections. In other words, the BUNDLE attribute must
contain all m= sections that were previously bundled, as long as they contain all m= sections that were previously bundled, as long as they
are still alive, as well as any new m= sections. are still alive, as well as any new m= sections.
skipping to change at page 29, line 44 skipping to change at page 28, line 30
o The "a=rtcp-rsize" line MUST only be added if present in the o The "a=rtcp-rsize" line MUST only be added if present in the
remote description. remote description.
The "a=group:BUNDLE" attribute MUST include the mid identifiers The "a=group:BUNDLE" attribute MUST include the mid identifiers
specified in the BUNDLE group in the most recent answer, minus any m= specified in the BUNDLE group in the most recent answer, minus any m=
sections that have been marked as rejected, plus any newly added or sections that have been marked as rejected, plus any newly added or
re-enabled m= sections. In other words, the BUNDLE attribute must re-enabled m= sections. In other words, the BUNDLE attribute must
contain all m= sections that were previously bundled, as long as they contain all m= sections that were previously bundled, as long as they
are still alive, as well as any new m= sections. are still alive, as well as any new m= sections.
5.2.3. Constraints Handling 5.2.3. Options Handling
The createOffer method takes as a parameter a MediaConstraints The createOffer method takes as a parameter an RTCOfferOptions
object. Special processing is performed when generating a SDP object. Special processing is performed when generating a SDP
description if the following constraints are present. description if the following constraints are present.
5.2.3.1. OfferToReceiveAudio 5.2.3.1. OfferToReceiveAudio
If the "OfferToReceiveAudio" constraint is specified, with a value of If the "OfferToReceiveAudio" option is specified, with an integer
"N", the offer MUST include N non-rejected m= sections with media value of N, the offer MUST include N non-rejected m= sections with
type "audio", even if fewer than N audio MediaStreamTracks have been media type "audio", even if fewer than N audio MediaStreamTracks have
added to the PeerConnection. This allows the offerer to receive been added to the PeerConnection. This allows the offerer to receive
audio, including multiple independent streams, even when not sending audio, including multiple independent streams, even when not sending
it; accordingly, the directional attribute on the audio m= sections it; accordingly, the directional attribute on the audio m= sections
without associated MediaStreamTracks MUST be set to recvonly. If without associated MediaStreamTracks MUST be set to recvonly. If
this constraint is specified in the case where at least N audio this option is specified in the case where at least N audio
MediaStreamTracks have already been added to the PeerConnection, or N MediaStreamTracks have already been added to the PeerConnection, or N
non-rejected m= sections with media type "audio" would otherwise be non-rejected m= sections with media type "audio" would otherwise be
generated, it has no effect. For backwards compatibility, a value of generated, it has no effect. For backwards compatibility, a value of
"true" is interpreted as equivalent to N=1. "true" is interpreted as equivalent to N=1.
5.2.3.2. OfferToReceiveVideo 5.2.3.2. OfferToReceiveVideo
If the "OfferToReceiveVideo" constraint is specified, with a value of If the "OfferToReceiveVideo" option is specified, with an integer
"N", the offer MUST include N non-rejected m= sections with media value of N, the offer MUST include N non-rejected m= sections with
type "video", even if fewer than N video MediaStreamTracks have been media type "video", even if fewer than N video MediaStreamTracks have
added to the PeerConnection. This allows the offerer to receive been added to the PeerConnection. This allows the offerer to receive
video, including multiple independent streams, even when not sending video, including multiple independent streams, even when not sending
it; accordingly, the directional attribute on the video m= sections it; accordingly, the directional attribute on the video m= sections
without associated MediaStreamTracks MUST be set to recvonly. If without associated MediaStreamTracks MUST be set to recvonly. If
this constraint is specified in the case where at least N video this option is specified in the case where at least N video
MediaStreamTracks have already been added to the PeerConnection, or N MediaStreamTracks have already been added to the PeerConnection, or N
non-rejected m= sections with media type "video" would otherwise be non-rejected m= sections with media type "video" would otherwise be
generated, it has no effect. For backwards compatibility, a value of generated, it has no effect. For backwards compatibility, a value of
"true" is interpreted as equivalent to N=1. "true" is interpreted as equivalent to N=1.
5.2.3.3. VoiceActivityDetection 5.2.3.3. VoiceActivityDetection
If the "VoiceActivityDetection" constraint is specified, with a value If the "VoiceActivityDetection" option is specified, with a value of
of "true", the offer MUST indicate support for silence suppression by "true", the offer MUST indicate support for silence suppression in
including comfort noise ("CN") codecs for each supported clock rate, the audio it receives by including comfort noise ("CN") codecs for
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
support. For codecs that have their own internal silence suppression
support, the appropriate fmtp parameters for that codec MUST be
specified to indicate that silence suppression for received audio is
desired. For example, when using the Opus codec, the "usedtx=1"
parameter would be specified in the offer. This option allows the
endpoint to significantly reduce the amount of audio bandwidth it
receives, at the cost of some fidelity, depending on the quality of
the remote VAD algorithm.
5.2.3.4. IceRestart 5.2.3.4. IceRestart
If the "IceRestart" constraint is specified, with a value of "true", If the "IceRestart" option is specified, with a value of "true", the
the offer MUST indicate an ICE restart by generating new ICE ufrag offer MUST indicate an ICE restart by generating new ICE ufrag and
and pwd attributes, as specified in RFC5245, Section 9.1.1.1. If pwd attributes, as specified in RFC5245, Section 9.1.1.1. If this
this constraint is specified on an initial offer, it has no effect option is specified on an initial offer, it has no effect (since a
(since a new ICE ufrag and pwd are already generated). new ICE ufrag and pwd are already generated). This option is useful
for reestablishing connectivity in cases where failures are detected.
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
details of this process are explained below. details of this process are explained below.
5.3.1. Initial Answers 5.3.1. Initial Answers
skipping to change at page 31, line 37 skipping to change at page 30, line 30
The first step in generating an initial answer is to generate The first step in generating an initial answer is to generate
session-level attributes. The process here is identical to that session-level attributes. The process here is identical to that
indicated in the Initial Offers section above. indicated in the Initial Offers section above.
The next step is to generate m= sections for each m= section that is The next step is to generate m= sections for each m= section that is
present in the remote offer, as specified in [RFC3264], Section 6. present in the remote offer, as specified in [RFC3264], Section 6.
For the purposes of this discussion, any session-level attributes in For the purposes of this discussion, any session-level attributes in
the offer that are also valid as media-level attributes SHALL be the offer that are also valid as media-level attributes SHALL be
considered to be present in each m= section. considered to be present in each m= section.
For each offered m= section of a given media type, if there is a The next step is to go through each offered m= section. If there is
local MediaStreamTrack of the specified type which has been added to a local MediaStreamTrack of the same type which has been added to the
the PeerConnection via addStream and not yet associated with a m= PeerConnection via addStream and not yet associated with a m=
section, and the specific m= section is either sendrecv or recvonly, section, and the specific m= section is either sendrecv or recvonly,
the MediaStreamTrack is associated with the m= section at this time. the MediaStreamTrack will be associated with the m= section at this
If there are more m= sections of a certain type than time. MediaStreamTracks are assigned to m= sections using the
MediaStreamTracks, some m= sections will not have an associated canonical order described in Section 5.2.1. If there are more m=
MediaStreamTrack. If there are more MediaStreamTracks of a certain sections of a certain type than MediaStreamTracks, some m= sections
type than compatible m= sections, only the first N MediaStreamTracks will not have an associated MediaStreamTrack. If there are more
will be able to be associated in the constructed answer. The MediaStreamTracks of a certain type than compatible m= sections, only
remainder will need to be associated in a subsequent offer. the first N MediaStreamTracks will be able to be associated in the
constructed answer. The remainder will need to be associated in a
subsequent offer.
For each offered m= section, if the associated remote For each offered m= section, if the associated remote
MediaStreamTrack has been stopped, and is therefore in state "ended", MediaStreamTrack has been stopped, and is therefore in state "ended",
and no local MediaStreamTrack has been associated, the corresponding and no local MediaStreamTrack has been associated, the corresponding
m= section in the answer MUST be marked as rejected by setting the m= section in the answer MUST be marked as rejected by setting the
port in the m= line to zero, as indicated in [RFC3264], Section 6., port in the m= line to zero, as indicated in [RFC3264], Section 6.,
and further processing for this m= section can be skipped. and further processing for this m= section can be skipped.
Provided that is not the case, each m= section in the answer should Provided that is not the case, each m= section in the answer should
then generated as specified in [RFC3264], Section 6.1. Because use then be generated as specified in [RFC3264], Section 6.1. Because
of DTLS is mandatory, the <proto> field MUST be set to "UDP/TLS/RTP/ use of DTLS is mandatory, the <proto> field MUST be set to "UDP/TLS/
SAVPF". If the offer supports BUNDLE, all m= sections to be BUNDLEd RTP/SAVPF". If the offer supports BUNDLE, all m= sections to be
must use the same ICE credentials and candidates; all m= sections not BUNDLEd must use the same ICE credentials and candidates; all m=
being BUNDLEd must use unique ICE credentials and candidates. Each sections not being BUNDLEd must use unique ICE credentials and
m= section MUST include the following: candidates. Each m= section MUST include the following:
o If present in the offer, an "a=mid" line, as specified in o If present in the offer, an "a=mid" line, as specified in
[RFC5888], Section 9.1. The "mid" value MUST match that specified [RFC5888], Section 9.1. The "mid" value MUST match that specified
in the offer. in the offer.
o If a local MediaStreamTrack has been associated, an "a=msid" line, o If a local MediaStreamTrack has been associated, an "a=msid" line,
as specified in [I-D.ietf-mmusic-msid], Section 2. as specified in [I-D.ietf-mmusic-msid], Section 2.
o [OPEN ISSUE: Use of AppID]
o [OPEN ISSUE: Use of AppID]
o Depending on the directionality of the offer, the disposition of o Depending on the directionality of the offer, the disposition of
any associated remote MediaStreamTrack, and the presence of an any associated remote MediaStreamTrack, and the presence of an
associated local MediaStreamTrack, the appropriate directionality associated local MediaStreamTrack, the appropriate directionality
attribute, as specified in [RFC3264], Section 6.1. If the offer attribute, as specified in [RFC3264], Section 6.1. If the offer
was sendrecv, and the remote MediaStreamTrack is still "live", and was sendrecv, and the remote MediaStreamTrack is still "live", and
there a local MediaStreamTrack has been associated, the there is a local MediaStreamTrack that has been associated, the
directionality MUST be set as sendrecv. If the offer was directionality MUST be set as sendrecv. If the offer was
sendonly, and the remote MediaStreamTrack is still "live", the sendonly, and the remote MediaStreamTrack is still "live", the
directionality MUST be set as recvonly. If the offer was directionality MUST be set as recvonly. If the offer was
recvonly, and a local MediaStreamTrack has been associated, the recvonly, and a local MediaStreamTrack has been associated, the
directionality MUST be set as sendonly. If the offer was directionality MUST be set as sendonly. If the offer was
inactive, the directionality MUST be set as inactive. inactive, the directionality MUST be set as inactive.
o For each supported codec that is present in the offer, "a=rtpmap" o For each supported codec that is present in the offer, "a=rtpmap"
and "a=fmtp" lines, as specified in [RFC4566], Section 6, and and "a=fmtp" lines, as specified in [RFC4566], Section 6, and
[RFC3264], Section 6.1. For audio, the codecs specified in [RFC3264], Section 6.1. For audio, the codecs specified in
[I-D.ietf-rtcweb-audio], Section 3, MUST be be supported. Note [I-D.ietf-rtcweb-audio], Section 3, MUST be supported. Note that
that for simplicity, the answerer MAY use different payload types for simplicity, the answerer MAY use different payload types for
for codecs than the offerer, as it is not prohibited by codecs than the offerer, as it is not prohibited by Section 6.1.
Section 6.1.
o If "rtx" is present in the offer, for each primary codec where RTP o If "rtx" is present in the offer, for each primary codec where RTP
retransmission should be used, a corresponding "a=rtpmap" line retransmission should be used, a corresponding "a=rtpmap" line
indicating "rtx" with the clock rate of the primary codec and an indicating "rtx" with the clock rate of the primary codec and an
"a=fmtp" line that references the payload type fo the primary "a=fmtp" line that references the payload type of the primary
codec, as specified in [RFC4588], Section 8.1. codec, as specified in [RFC4588], Section 8.1.
o For each supported FEC mechanism that is present in the offer, a o For each supported FEC mechanism that is present in the offer, a
corresponding "a=rtpmap" line indicating the desired FEC codec. corresponding "a=rtpmap" line indicating the desired FEC codec.
o "a=ice-ufrag" and "a=ice-passwd" lines, as specified in [RFC5245], o "a=ice-ufrag" and "a=ice-passwd" lines, as specified in [RFC5245],
Section 15.4. Section 15.4.
o If the "trickle" ICE option is present in the offer, an "a=ice- o If the "trickle" ICE option is present in the offer, an "a=ice-
options" line, with the "trickle" option, as specified in options" line, with the "trickle" option, as specified in
[I-D.ietf-mmusic-trickle-ice], Section 4. [I-D.ietf-mmusic-trickle-ice], Section 4.
o For each candidate that has been gathered during the most recent o For each candidate that has been gathered during the most recent
gathering phase, an "a=candidate" line, as specified in [RFC5245], gathering phase, an "a=candidate" line, as specified in [RFC5245],
Section 4.3., paragraph 3. Section 4.3., paragraph 3.
o For the current default candidate, a "c=" line, as specified in
o For the current default candidate, a "c=" line, as specific in
[RFC5245], Section 4.3., paragraph 6. If no candidates have been [RFC5245], Section 4.3., paragraph 6. If no candidates have been
gathered yet, the default candidate should be set to the 'null' gathered yet, the default candidate should be set to the 'null'
value defined in [I-D.ietf-mmusic-trickle-ice], Section 5.1. value defined in [I-D.ietf-mmusic-trickle-ice], Section 5.1.
o An "a=fingerprint" line, as specified in [RFC4572], Section 5; the o An "a=fingerprint" line, as specified in [RFC4572], Section 5; the
algorithm used for the fingerprint MUST match that used in the algorithm used for the fingerprint MUST match that used in the
certificate signature. certificate 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"; the
"active" role is RECOMMENDED. "active" role is RECOMMENDED.
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 5.1.1. [RFC5761], Section 5.1.1.
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.
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" mechanism, 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 a local MediaStreamTrack has been associated, an "a=ssrc" line, o If a local MediaStreamTrack has been associated, an "a=ssrc" line,
as specified in [RFC5576], Section 4.1, indicating the SSRC to be as specified in [RFC5576], Section 4.1, indicating the SSRC to be
used for sending media. used for sending media.
o If a local MediaStreamTrack has been associated, and RTX has been o If a local MediaStreamTrack has been associated, and RTX has been
negotiated for this m= section, another "a=ssrc" line with the RTX negotiated for this m= section, another "a=ssrc" line with the RTX
SSRC, and an "a=ssrc-group" line, as specified in [RFC5576], SSRC, and an "a=ssrc-group" line, as specified in [RFC5576],
section 4.2, with semantics set to "FID" and including the primary section 4.2, with semantics set to "FID" and including the primary
and RTX SSRCs. and RTX SSRCs.
o If a local MediaStreamTrack has been associated, and FEC has been o If a local MediaStreamTrack has been associated, and FEC has been
negotiated for this m= section, another "a=ssrc" line with the FEC negotiated for this m= section, another "a=ssrc" line with the FEC
SSRC, and an "a=ssrc-group" line, as specified in [RFC5576], SSRC, and an "a=ssrc-group" line, as specified in [RFC5576],
section 4.2, with semantics set to "FEC" and including the primary section 4.2, with semantics set to "FEC" and including the primary
and FEC SSRCs. and FEC SSRCs.
o [OPEN ISSUE: Handling of a=imageattr]
o [OPEN ISSUE: Handling of a=imageattr]
If a data channel m= section has been offered, a m= section MUST also If a data channel m= section has been offered, a m= section MUST also
be generated for data. The <media> field MUST be set to be generated for data. The <media> field MUST be set to
"application" and the <proto> field MUST be set to "DTLS/SCTP", as "application" and the <proto> field MUST be set to "DTLS/SCTP", as
specified in [I-D.ietf-mmusic-sctp-sdp], Section 3; the "fmt" value specified in [I-D.ietf-mmusic-sctp-sdp], Section 3; the "fmt" value
MUST be set to the SCTP port number, as specified in Section 4.1. MUST be set to the SCTP port number, as specified in Section 4.1.
Within the data m= section, the "a=mid", "a=ice-ufrag", "a=ice- Within the data m= section, the "a=mid", "a=ice-ufrag", "a=ice-
passwd", "a=ice-options", "a=candidate", "a=fingerprint", and passwd", "a=ice-options", "a=candidate", "a=fingerprint", and
"a=setup" lines MUST be included as mentioned above, along with an "a=setup" lines MUST be included as mentioned above, along with an
"a=sctpmap" line referencing the SCTP port number and specifying the "a=sctpmap" line referencing the SCTP port number and specifying the
application protocol indicated in [I-D.ietf-rtcweb-data-protocol]. application protocol indicated in [I-D.ietf-rtcweb-data-protocol].
[OPEN ISSUE: the -01 of this document is missing this information.]
[OPEN ISSUE: the -01 of this document is missing this information.]
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
BUNDLE groups that have not been rejected. Note that regardless of BUNDLE groups that have not been rejected. Note that regardless of
the presence of "a=bundle-only" in the offer, no m= sections in the the presence of "a=bundle-only" in the offer, no m= sections in the
answer should have an "a=bundle-only" line. answer should have an "a=bundle-only" line.
Attributes that are common between all m= sections MAY be moved to Attributes that are common between all m= sections MAY be moved to
session-level, if explicitly defined to be valid at session-level. session-level, if explicitly defined to be valid at session-level.
The attributes prohibited in the creation of offers are also The attributes prohibited in the creation of offers are also
prohibited in the creation of answers. prohibited in the creation of answers.
5.3.2. Subsequent Answers 5.3.2. Subsequent Answers
5.3.3. Constraints Handling 5.3.3. Options Handling
5.4. Parsing an Offer 5.4. Parsing an Offer
5.5. Parsing an Answer 5.5. Parsing an Answer
5.6. Applying a Local Description 5.6. Applying a Local Description
5.7. Applying a Remote Description 5.7. Applying a Remote Description
6. Configurable SDP Parameters 6. Configurable SDP Parameters
Note: This section is still very early and is likely to significantly It is possible to change elements in the SDP returned from
change as we get a better understanding of a) the use cases for this createOffer before passing it to setLocalDescription. When an
b) the implications at the protocol level c) feedback from implementation receives modified SDP it MUST either:
implementors on what they can do.
o Accept the changes and adjust its behavior to match the SDP.
o Reject the changes and return an error via the error callback.
Changes MUST NOT be silently ignored.
The following elements of the SDP media description MUST NOT be The following elements of the SDP media description MUST NOT be
changed between the createOffer and the setLocalDescription, since changed between the createOffer and the setLocalDescription, since
they reflect transport attributes that are solely under browser they reflect transport attributes that are solely under browser
control, and the browser MUST NOT honor an attempt to change them: control, and the browser MUST NOT honor an attempt to change them:
o The number, type and port number of m-lines. o The number, type and port number of m-lines.
o The generated ICE credentials (a=ice-ufrag and a=ice-pwd). o The generated ICE credentials (a=ice-ufrag and a=ice-pwd).
o The set of ICE candidates and their parameters (a=candidate). o The set of ICE candidates and their parameters (a=candidate).
The following modifications, if done by the browser to a description The following modifications, if done by the browser to a description
between createOffer/createAnswer and the setLocalDescription, MUST be between createOffer/createAnswer and the setLocalDescription, MUST be
honored by the browser: honored by the browser:
o Remove or reorder codecs (m=) o Remove or reorder codecs (m=)
The following parameters may be controlled by constraints passed into The following parameters may be controlled by constraints passed into
createOffer/createAnswer. As an open issue, these changes may also createOffer/createAnswer. As an open issue, these changes may also
skipping to change at page 35, line 48 skipping to change at page 34, line 21
o Remove or reorder codecs (m=) o Remove or reorder codecs (m=)
The following parameters may be controlled by constraints passed into The following parameters may be controlled by constraints passed into
createOffer/createAnswer. As an open issue, these changes may also createOffer/createAnswer. As an open issue, these changes may also
be be performed by manipulating the SDP returned from createOffer/ be be performed by manipulating the SDP returned from createOffer/
createAnswer, as indicated above, as long as the capabilities of the createAnswer, as indicated above, as long as the capabilities of the
endpoint are not exceeded (e.g. asking for a resolution greater than endpoint are not exceeded (e.g. asking for a resolution greater than
what the endpoint can encode): what the endpoint can encode):
o disable BUNDLE (a=group) o [[OPEN ISSUE: This is a placeholder for other modifications,
which we may continue adding as use cases appear.]]
o disable RTCP mux (a=rtcp-mux)
o change send resolution or frame rate
o change desired recv resolution or frame rate
o change maximum total bandwidth (b=) [OPEN ISSUE: need to clarify
if this is CT or AS - see section 5.8 of [RFC4566]]
o remove desired AVPF mechanisms (a=rtcp-fb)
o remove RTP header extensions (a=extmap)
o change media send/recv state (a=sendonly/recvonly/inactive)
For example, an application could implement call hold by adding an Implementations MAY choose to either honor or reject any elements not
a=inactive attribute to its local description, and then applying and listed in the above two categories, but must do so explicitly as
signaling that description. described at the beginning of this section. Note that future
standards may add new SDP elements to the list of elements which must
be accepted or rejected, but due to version skew, applications must
be prepared for implementations to accept changes which must be
rejected and vice versa.
The application can also modify the SDP to reduce the capabilities in The application can also modify the SDP to reduce the capabilities in
the offer it sends to the far side in any way the application sees the offer it sends to the far side or the offer that it installs from
fit, as long as it is a valid SDP offer and specifies a subset of the far side in any way the application sees fit, as long as it is a
what the browser is expecting to do. valid SDP offer and specifies a subset of what was in the original
offer. This is safe because the answer is not permitted to expand
capabilities and therefore will just respond to what is actually in
the offer.
As always, the application is solely responsible for what it sends to As always, the application is solely responsible for what it sends to
the other party, and all incoming SDP will be processed by the the other party, and all incoming SDP will be processed by the
browser to the extent of its capabilities. It is an error to assume browser to the extent of its capabilities. It is an error to assume
that all SDP is well-formed; however, one should be able to assume that all SDP is well-formed; however, one should be able to assume
that any implementation of this specification will be able to 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.
7. Security Considerations 7. Security Considerations
The intent of the WebRTC protocol suite is to provide an environment The IETF has published separate documents
that is securable by default: all media is encrypted, keys are [I-D.ietf-rtcweb-security-arch] [I-D.ietf-rtcweb-security] describing
exchanged in a secure fashion, and the Javascript API includes the security architecture for WebRTC as a whole. The remainder of
functions that can be used to verify the identity of communication this section describes security considerations for this document.
partners.
While formally the JSEP interface is an API, it is better to think of
it is an Internet protocol, with the JS being untrustworthy from the
perspective of the browser. Thus, the threat model of [RFC3552]
applies. In particular, JS can call the API in any order and with
any inputs, including malicious ones. This is particularly relevant
when we consider the SDP which is passed to setLocalDescription().
While correct API usage requires that the application pass in SDP
which was derived from createOffer() or createAnswer() (perhaps
suitably modified as described in Section 6, there is no guarantee
that applications do so. The browser MUST be prepared for the JS to
pass in bogus data instead.
Conversely, the application programmer MUST recognize that the JS
does not have complete control of browser behavior. One case that
bears particular mention is that editing ICE candidates out of the
SDP or suppressing trickled candidates does not have the expected
behavior: implementations will still perform checks from those
candidates even if they are not sent to the other side. Thus, for
instance, it is not possible to prevent the remote peer from learning
your public IP address by removing server reflexive candidates.
Applications which wish to conceal their public IP address should
instead configure the ICE agent to use only relay candidates.
8. IANA Considerations 8. IANA Considerations
This document requires no actions from IANA. This document requires no actions from IANA.
9. Acknowledgements 9. Acknowledgements
Significant text incorporated in the draft as well and review was Significant text incorporated in the draft as well and review was
provided by Harald Alvestrand and Suhas Nandakumar. Dan Burnett, provided by Harald Alvestrand and Suhas Nandakumar. Dan Burnett,
Neil Stratford, Eric Rescorla, Anant Narayanan, Andrew Hutton, Neil Stratford, Eric Rescorla, Anant Narayanan, Andrew Hutton,
Richard Ejzak, and Adam Bergkvist all provided valuable feedback on Richard Ejzak, Adam Bergkvist and Matthew Kaufman all provided
this proposal. Matthew Kaufman provided the observation that keeping valuable feedback on this proposal.
state out of the browser allows a call to continue even if the page
is reloaded.
10. References 10. References
10.1. Normative References 10.1. Normative References
[I-D.ietf-mmusic-msid] [I-D.ietf-mmusic-msid]
Alvestrand, H., "Cross Session Stream Identification in Alvestrand, H., "Cross Session Stream Identification in
the Session Description Protocol", draft-ietf-mmusic- the Session Description Protocol",
msid-01 (work in progress), August 2013. draft-ietf-mmusic-msid-01 (work in progress), August 2013.
[I-D.ietf-mmusic-sctp-sdp] [I-D.ietf-mmusic-sctp-sdp]
Loreto, S. and G. Camarillo, "Stream Control Transmission Loreto, S. and G. Camarillo, "Stream Control Transmission
Protocol (SCTP)-Based Media Transport in the Session Protocol (SCTP)-Based Media Transport in the Session
Description Protocol (SDP)", draft-ietf-mmusic-sctp-sdp-04 Description Protocol (SDP)", draft-ietf-mmusic-sctp-sdp-04
(work in progress), June 2013. (work in progress), June 2013.
[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 "Multiplexing Negotiation Using Session Description
Protocol (SDP) Port Numbers", draft-ietf-mmusic-sdp- Protocol (SDP) Port Numbers",
bundle-negotiation-04 (work in progress), June 2013. draft-ietf-mmusic-sdp-bundle-negotiation-04 (work in
progress), June 2013.
[I-D.ietf-mmusic-sdp-mux-attributes]
Nandakumar, S., "A Framework for SDP Attributes when
Multiplexing", draft-ietf-mmusic-sdp-mux-attributes-01
(work in progress), February 2014.
[I-D.ietf-rtcweb-audio] [I-D.ietf-rtcweb-audio]
Valin, J. and C. Bran, "WebRTC Audio Codec and Processing Valin, J. and C. Bran, "WebRTC Audio Codec and Processing
Requirements", draft-ietf-rtcweb-audio-02 (work in Requirements", draft-ietf-rtcweb-audio-02 (work in
progress), August 2013. progress), August 2013.
[I-D.ietf-rtcweb-data-protocol]
Jesup, R., Loreto, S., and M. Tuexen, "WebRTC Data Channel
Protocol", draft-ietf-rtcweb-data-protocol-04 (work in
progress), February 2013.
[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-09 (work in progress),
September 2013. September 2013.
[I-D.nandakumar-mmusic-sdp-mux-attributes] [I-D.ietf-rtcweb-security]
Nandakumar, S., "A Framework for SDP Attributes when Rescorla, E., "Security Considerations for WebRTC",
Multiplexing", draft-nandakumar-mmusic-sdp-mux- draft-ietf-rtcweb-security-06 (work in progress),
attributes-03 (work in progress), July 2013. January 2014.
[I-D.ietf-rtcweb-security-arch]
Rescorla, E., "WebRTC Security Architecture",
draft-ietf-rtcweb-security-arch-09 (work in progress),
February 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, March 1997.
[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. June 2002.
[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. June 2002.
[RFC3552] Rescorla, E. and B. Korver, "Guidelines for Writing RFC
Text on Security Considerations", BCP 72, RFC 3552,
July 2003.
[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. September 2005.
[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, July 2006.
[RFC4572] Lennox, J., "Connection-Oriented Media Transport over the [RFC4572] Lennox, J., "Connection-Oriented Media Transport over the
Transport Layer Security (TLS) Protocol in the Session Transport Layer Security (TLS) Protocol in the Session
Description Protocol (SDP)", RFC 4572, July 2006. 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. July 2006.
[RFC5124] Ott, J. and E. Carrara, "Extended Secure RTP Profile for [RFC5124] Ott, J. and E. Carrara, "Extended Secure RTP Profile for
Real-time Transport Control Protocol (RTCP)-Based Feedback Real-time Transport Control Protocol (RTCP)-Based Feedback
(RTP/SAVPF)", RFC 5124, February 2008. (RTP/SAVPF)", RFC 5124, February 2008.
[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. April 2010.
[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, July 2008.
[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, April 2010.
[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, June 2010.
[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. April 2013.
[RFC7022] Begen, A., Perkins, C., Wing, D., and E. Rescorla,
"Guidelines for Choosing RTP Control Protocol (RTCP)
Canonical Names (CNAMEs)", RFC 7022, September 2013.
10.2. Informative References 10.2. Informative References
[I-D.ietf-mmusic-trickle-ice] [I-D.ietf-mmusic-trickle-ice]
Ivov, E., Rescorla, E., and J. Uberti, "Trickle ICE: Ivov, E., Rescorla, E., and J. Uberti, "Trickle ICE:
Incremental Provisioning of Candidates for the Interactive Incremental Provisioning of Candidates for the Interactive
Connectivity Establishment (ICE) Protocol", draft-ietf- Connectivity Establishment (ICE) Protocol",
mmusic-trickle-ice-00 (work in progress), March 2013. draft-ietf-mmusic-trickle-ice-00 (work in progress),
March 2013.
[I-D.ietf-rtcweb-data-protocol]
Jesup, R., Loreto, S., and M. Tuexen, "WebRTC Data Channel
Protocol", draft-ietf-rtcweb-data-protocol-04 (work in
progress), February 2013.
[I-D.jennings-rtcweb-signaling]
Jennings, C., Rosenberg, J., and R. Jesup, "RTCWeb Offer/
Answer Protocol (ROAP)", draft-jennings-rtcweb-
signaling-01 (work in progress), October 2011.
[I-D.nandakumar-rtcweb-sdp] [I-D.nandakumar-rtcweb-sdp]
Nandakumar, S. and C. Jennings, "SDP for the WebRTC", Nandakumar, S. and C. Jennings, "SDP for the WebRTC",
draft-nandakumar-rtcweb-sdp-02 (work in progress), July draft-nandakumar-rtcweb-sdp-02 (work in progress),
2013. July 2013.
[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, September 2002.
[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, July 2003.
[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, December 2004.
[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, July 2006.
[RFC4588] Rey, J., Leon, D., Miyazaki, A., Varsa, V., and R. [RFC4588] Rey, J., Leon, D., Miyazaki, A., Varsa, V., and R.
skipping to change at page 40, line 14 skipping to change at page 39, line 14
[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, May 2010.
[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, May 2010.
[W3C.WD-webrtc-20111027] [W3C.WD-webrtc-20140617]
Bergkvist, A., Burnett, D., Narayanan, A., and C. Bergkvist, A., Burnett, D., Narayanan, A., and C.
Jennings, "WebRTC 1.0: Real-time Communication Between Jennings, "WebRTC 1.0: Real-time Communication Between
Browsers", World Wide Web Consortium WD WD- Browsers", World Wide Web Consortium WD WD-webrtc-
webrtc-20111027, October 2011, 20140617, June 2014,
<http://www.w3.org/TR/2011/WD-webrtc-20111027>. <http://www.w3.org/TR/2011/WD-webrtc-20140617>.
Appendix A. JSEP Implementation Examples Appendix A. JSEP Implementation Examples
A.1. Example API Flows A.1. Example API Flows
Below are several sample flows for the new PeerConnection and library Below are several sample flows for the new PeerConnection and library
APIs, demonstrating when the various APIs are called in different APIs, demonstrating when the various APIs are called in different
situations and with various transport protocols. For clarity and situations and with various transport protocols. For clarity and
simplicity, the createOffer/createAnswer calls are assumed to be simplicity, the createOffer/createAnswer calls are assumed to be
synchronous in these examples, whereas the actual APIs are async. synchronous in these examples, whereas the actual APIs are async.
skipping to change at page 50, line 8 skipping to change at page 49, line 8
a=ice-pwd:cOTZKZNVlO9RSGsEGM63JXT2 a=ice-pwd:cOTZKZNVlO9RSGsEGM63JXT2
a=fingerprint:sha-256 6B:8B:F0:65:5F:78:E2:51:3B:AC:6F:F3:3F:46:1B:35 a=fingerprint:sha-256 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=mid:data a=mid:data
a=sctpmap:5000 webrtc-datachannel 16 a=sctpmap:5000 webrtc-datachannel 16
A.2.3. Call Flows A.2.3. Call Flows
Example SDP for WebRTC call flows can be found in Example SDP for WebRTC call flows can be found in
[I-D.nandakumar-rtcweb-sdp]. [TODO: should these call flows be [I-D.nandakumar-rtcweb-sdp]. [TODO: should these call flows be
merged into this section?] merged into this section?]
Appendix B. Change log Appendix B. Change log
Changes in draft-06: Changes in draft-06:
o Reworked handling of m= line recycling. o Reworked handling of m= line recycling.
o Added handling of BUNDLE and bundle-only. o Added handling of BUNDLE and bundle-only.
o Clarified handling of rollback. o Clarified handling of rollback.
o Added text describing the ICE Candidate Pool and its behavior. o Added text describing the ICE Candidate Pool and its behavior.
o Allowed OfferToReceiveX to create multiple recvonly m= sections. o Allowed OfferToReceiveX to create multiple recvonly m= sections.
Changes in draft-05: Changes in draft-05:
o Fixed several issues identified in the createOffer/Answer sections o Fixed several issues identified in the createOffer/Answer sections
during document review. during document review.
o Updated references. o Updated references.
Changes in draft-04: Changes in draft-04:
o Filled in sections on createOffer and createAnswer. o Filled in sections on createOffer and createAnswer.
o Added SDP examples. o Added SDP examples.
o Fixed references. o Fixed references.
Changes in draft-03: Changes in draft-03:
o Added text describing relationship to W3C specification o Added text describing relationship to W3C specification
Changes in draft-02: Changes in draft-02:
o Converted from nroff o Converted from nroff
o Removed comparisons to old approaches abandoned by the working o Removed comparisons to old approaches abandoned by the working
group group
o Removed stuff that has moved to W3C specification o Removed stuff that has moved to W3C specification
o Align SDP handling with W3C draft o Align SDP handling with W3C draft
o Clarified section on forking. o Clarified section on forking.
Changes in draft-01: Changes in draft-01:
o Added diagrams for architecture and state machine. o Added diagrams for architecture and state machine.
o Added sections on forking and rehydration. o Added sections on forking and rehydration.
o Clarified meaning of "pranswer" and "answer". o Clarified meaning of "pranswer" and "answer".
o Reworked how ICE restarts and media directions are controlled. o Reworked how ICE restarts and media directions are controlled.
o Added list of parameters that can be changed in a description. o Added list of parameters that can be changed in a description.
o Updated suggested API and examples to match latest thinking. o Updated suggested API and examples to match latest thinking.
o Suggested API and examples have been moved to an appendix. o Suggested API and examples have been moved to an appendix.
Changes in draft -00: Changes in draft -00:
o Migrated from draft-uberti-rtcweb-jsep-02. o Migrated from draft-uberti-rtcweb-jsep-02.
Authors' Addresses Authors' Addresses
Justin Uberti Justin Uberti
Google Google
skipping to change at page 51, line 36 skipping to change at page 50, line 15
o Migrated from draft-uberti-rtcweb-jsep-02. o Migrated from draft-uberti-rtcweb-jsep-02.
Authors' Addresses Authors' Addresses
Justin Uberti Justin Uberti
Google Google
747 6th Ave S 747 6th Ave S
Kirkland, WA 98033 Kirkland, WA 98033
USA USA
Email: justin@uberti.name Email: justin@uberti.name
Cullen Jennings Cullen Jennings
Cisco Cisco
170 West Tasman Drive 170 West Tasman Drive
San Jose, CA 95134 San Jose, CA 95134
USA USA
Email: fluffy@iii.ca Email: fluffy@iii.ca
Eric Rescorla (editor)
Mozilla
331 Evelyn Ave
Mountain View, CA 94041
USA
Email: ekr@rtfm.com
 End of changes. 183 change blocks. 
493 lines changed or deleted 434 lines changed or added

This html diff was produced by rfcdiff 1.41. The latest version is available from http://tools.ietf.org/tools/rfcdiff/