draft-ietf-rtcweb-jsep-17.txt   draft-ietf-rtcweb-jsep-18.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: April 24, 2017 Cisco Expires: July 20, 2017 Cisco
E. Rescorla, Ed. E. Rescorla, Ed.
Mozilla Mozilla
October 21, 2016 January 16, 2017
Javascript Session Establishment Protocol Javascript Session Establishment Protocol
draft-ietf-rtcweb-jsep-17 draft-ietf-rtcweb-jsep-18
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
skipping to change at page 1, line 36 skipping to change at page 1, line 36
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 April 24, 2017. This Internet-Draft will expire on July 20, 2017.
Copyright Notice Copyright Notice
Copyright (c) 2016 IETF Trust and the persons identified as the Copyright (c) 2017 IETF Trust and the persons identified as the
document authors. All rights reserved. document authors. All rights reserved.
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Provisions Relating to IETF Documents Provisions Relating to IETF Documents
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the Trust Legal Provisions and are provided without warranty as the Trust Legal Provisions and are provided without warranty as
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3.4. Session Description Control . . . . . . . . . . . . . . . 10 3.4. Session Description Control . . . . . . . . . . . . . . . 10
3.4.1. RtpTransceivers . . . . . . . . . . . . . . . . . . . 10 3.4.1. RtpTransceivers . . . . . . . . . . . . . . . . . . . 10
3.4.2. RtpSenders . . . . . . . . . . . . . . . . . . . . . 11 3.4.2. RtpSenders . . . . . . . . . . . . . . . . . . . . . 11
3.4.3. RtpReceivers . . . . . . . . . . . . . . . . . . . . 11 3.4.3. RtpReceivers . . . . . . . . . . . . . . . . . . . . 11
3.5. ICE . . . . . . . . . . . . . . . . . . . . . . . . . . . 11 3.5. ICE . . . . . . . . . . . . . . . . . . . . . . . . . . . 11
3.5.1. ICE Gathering Overview . . . . . . . . . . . . . . . 11 3.5.1. ICE Gathering Overview . . . . . . . . . . . . . . . 11
3.5.2. ICE Candidate Trickling . . . . . . . . . . . . . . . 12 3.5.2. ICE Candidate Trickling . . . . . . . . . . . . . . . 12
3.5.2.1. ICE Candidate Format . . . . . . . . . . . . . . 12 3.5.2.1. ICE Candidate Format . . . . . . . . . . . . . . 12
3.5.3. ICE Candidate Policy . . . . . . . . . . . . . . . . 13 3.5.3. ICE Candidate Policy . . . . . . . . . . . . . . . . 13
3.5.4. ICE Candidate Pool . . . . . . . . . . . . . . . . . 14 3.5.4. ICE Candidate Pool . . . . . . . . . . . . . . . . . 14
3.6. Video Size Negotiation . . . . . . . . . . . . . . . . . 14 3.6. Video Size Negotiation . . . . . . . . . . . . . . . . . 15
3.6.1. Creating an imageattr Attribute . . . . . . . . . . . 15 3.6.1. Creating an imageattr Attribute . . . . . . . . . . . 15
3.6.2. Interpreting an imageattr Attribute . . . . . . . . . 16 3.6.2. Interpreting an imageattr Attribute . . . . . . . . . 16
3.7. Simulcast . . . . . . . . . . . . . . . . . . . . . . . . 17 3.7. Simulcast . . . . . . . . . . . . . . . . . . . . . . . . 17
3.8. Interactions With Forking . . . . . . . . . . . . . . . . 18 3.8. Interactions With Forking . . . . . . . . . . . . . . . . 18
3.8.1. Sequential Forking . . . . . . . . . . . . . . . . . 18 3.8.1. Sequential Forking . . . . . . . . . . . . . . . . . 19
3.8.2. Parallel Forking . . . . . . . . . . . . . . . . . . 19 3.8.2. Parallel Forking . . . . . . . . . . . . . . . . . . 19
4. Interface . . . . . . . . . . . . . . . . . . . . . . . . . . 20 4. Interface . . . . . . . . . . . . . . . . . . . . . . . . . . 20
4.1. PeerConnection . . . . . . . . . . . . . . . . . . . . . 20 4.1. PeerConnection . . . . . . . . . . . . . . . . . . . . . 20
4.1.1. Constructor . . . . . . . . . . . . . . . . . . . . . 20 4.1.1. Constructor . . . . . . . . . . . . . . . . . . . . . 20
4.1.2. addTrack . . . . . . . . . . . . . . . . . . . . . . 22 4.1.2. addTrack . . . . . . . . . . . . . . . . . . . . . . 22
4.1.3. addTransceiver . . . . . . . . . . . . . . . . . . . 22 4.1.3. removeTrack . . . . . . . . . . . . . . . . . . . . . 23
4.1.4. createDataChannel . . . . . . . . . . . . . . . . . . 23 4.1.4. addTransceiver . . . . . . . . . . . . . . . . . . . 23
4.1.5. createOffer . . . . . . . . . . . . . . . . . . . . . 23 4.1.5. createDataChannel . . . . . . . . . . . . . . . . . . 23
4.1.6. createAnswer . . . . . . . . . . . . . . . . . . . . 24 4.1.6. createOffer . . . . . . . . . . . . . . . . . . . . . 24
4.1.7. SessionDescriptionType . . . . . . . . . . . . . . . 25 4.1.7. createAnswer . . . . . . . . . . . . . . . . . . . . 25
4.1.7.1. Use of Provisional Answers . . . . . . . . . . . 25 4.1.8. SessionDescriptionType . . . . . . . . . . . . . . . 25
4.1.7.2. Rollback . . . . . . . . . . . . . . . . . . . . 26 4.1.8.1. Use of Provisional Answers . . . . . . . . . . . 26
4.1.8. setLocalDescription . . . . . . . . . . . . . . . . . 27 4.1.8.2. Rollback . . . . . . . . . . . . . . . . . . . . 27
4.1.9. setRemoteDescription . . . . . . . . . . . . . . . . 28 4.1.9. setLocalDescription . . . . . . . . . . . . . . . . . 28
4.1.10. currentLocalDescription . . . . . . . . . . . . . . . 28 4.1.10. setRemoteDescription . . . . . . . . . . . . . . . . 28
4.1.11. pendingLocalDescription . . . . . . . . . . . . . . . 28 4.1.11. currentLocalDescription . . . . . . . . . . . . . . . 29
4.1.12. currentRemoteDescription . . . . . . . . . . . . . . 28 4.1.12. pendingLocalDescription . . . . . . . . . . . . . . . 29
4.1.13. pendingRemoteDescription . . . . . . . . . . . . . . 29 4.1.13. currentRemoteDescription . . . . . . . . . . . . . . 29
4.1.14. canTrickleIceCandidates . . . . . . . . . . . . . . . 29 4.1.14. pendingRemoteDescription . . . . . . . . . . . . . . 29
4.1.15. setConfiguration . . . . . . . . . . . . . . . . . . 30 4.1.15. canTrickleIceCandidates . . . . . . . . . . . . . . . 30
4.1.16. addIceCandidate . . . . . . . . . . . . . . . . . . . 30 4.1.16. setConfiguration . . . . . . . . . . . . . . . . . . 30
4.2. RtpTransceiver . . . . . . . . . . . . . . . . . . . . . 31 4.1.17. addIceCandidate . . . . . . . . . . . . . . . . . . . 31
4.2.1. stop . . . . . . . . . . . . . . . . . . . . . . . . 31 4.2. RtpTransceiver . . . . . . . . . . . . . . . . . . . . . 32
4.2.2. stopped . . . . . . . . . . . . . . . . . . . . . . . 31 4.2.1. stop . . . . . . . . . . . . . . . . . . . . . . . . 32
4.2.3. setDirection . . . . . . . . . . . . . . . . . . . . 31 4.2.2. stopped . . . . . . . . . . . . . . . . . . . . . . . 32
4.2.4. setCodecPreferences . . . . . . . . . . . . . . . . . 32 4.2.3. setDirection . . . . . . . . . . . . . . . . . . . . 32
5. SDP Interaction Procedures . . . . . . . . . . . . . . . . . 32 4.2.4. direction . . . . . . . . . . . . . . . . . . . . . . 32
5.1. Requirements Overview . . . . . . . . . . . . . . . . . . 32 4.2.5. currentDirection . . . . . . . . . . . . . . . . . . 33
5.1.1. Implementation Requirements . . . . . . . . . . . . . 33 4.2.6. setCodecPreferences . . . . . . . . . . . . . . . . . 33
5.1.2. Usage Requirements . . . . . . . . . . . . . . . . . 34 5. SDP Interaction Procedures . . . . . . . . . . . . . . . . . 33
5.1.3. Profile Names and Interoperability . . . . . . . . . 34 5.1. Requirements Overview . . . . . . . . . . . . . . . . . . 34
5.2. Constructing an Offer . . . . . . . . . . . . . . . . . . 35 5.1.1. Implementation Requirements . . . . . . . . . . . . . 34
5.2.1. Initial Offers . . . . . . . . . . . . . . . . . . . 35 5.1.2. Usage Requirements . . . . . . . . . . . . . . . . . 35
5.2.2. Subsequent Offers . . . . . . . . . . . . . . . . . . 41 5.1.3. Profile Names and Interoperability . . . . . . . . . 36
5.2.3. Options Handling . . . . . . . . . . . . . . . . . . 44 5.2. Constructing an Offer . . . . . . . . . . . . . . . . . . 37
5.2.3.1. IceRestart . . . . . . . . . . . . . . . . . . . 44 5.2.1. Initial Offers . . . . . . . . . . . . . . . . . . . 37
5.2.3.2. VoiceActivityDetection . . . . . . . . . . . . . 45 5.2.2. Subsequent Offers . . . . . . . . . . . . . . . . . . 42
5.3. Generating an Answer . . . . . . . . . . . . . . . . . . 45 5.2.3. Options Handling . . . . . . . . . . . . . . . . . . 46
5.3.1. Initial Answers . . . . . . . . . . . . . . . . . . . 45 5.2.3.1. IceRestart . . . . . . . . . . . . . . . . . . . 46
5.3.2. Subsequent Answers . . . . . . . . . . . . . . . . . 50 5.2.3.2. VoiceActivityDetection . . . . . . . . . . . . . 46
5.3.3. Options Handling . . . . . . . . . . . . . . . . . . 51 5.3. Generating an Answer . . . . . . . . . . . . . . . . . . 47
5.3.3.1. VoiceActivityDetection . . . . . . . . . . . . . 51 5.3.1. Initial Answers . . . . . . . . . . . . . . . . . . . 47
5.4. Modifying an Offer or Answer . . . . . . . . . . . . . . 51 5.3.2. Subsequent Answers . . . . . . . . . . . . . . . . . 51
5.5. Processing a Local Description . . . . . . . . . . . . . 52 5.3.3. Options Handling . . . . . . . . . . . . . . . . . . 53
5.6. Processing a Remote Description . . . . . . . . . . . . . 53 5.3.3.1. VoiceActivityDetection . . . . . . . . . . . . . 53
5.7. Parsing a Session Description . . . . . . . . . . . . . . 53 5.4. Modifying an Offer or Answer . . . . . . . . . . . . . . 53
5.7.1. Session-Level Parsing . . . . . . . . . . . . . . . . 54 5.5. Processing a Local Description . . . . . . . . . . . . . 54
5.7.2. Media Section Parsing . . . . . . . . . . . . . . . . 55 5.6. Processing a Remote Description . . . . . . . . . . . . . 54
5.7.3. Semantics Verification . . . . . . . . . . . . . . . 58 5.7. Parsing a Session Description . . . . . . . . . . . . . . 55
5.8. Applying a Local Description . . . . . . . . . . . . . . 59 5.7.1. Session-Level Parsing . . . . . . . . . . . . . . . . 55
5.9. Applying a Remote Description . . . . . . . . . . . . . . 60 5.7.2. Media Section Parsing . . . . . . . . . . . . . . . . 57
5.10. Applying an Answer . . . . . . . . . . . . . . . . . . . 64 5.7.3. Semantics Verification . . . . . . . . . . . . . . . 59
6. Processing RTP/RTCP packets . . . . . . . . . . . . . . . . . 66 5.8. Applying a Local Description . . . . . . . . . . . . . . 60
5.9. Applying a Remote Description . . . . . . . . . . . . . . 62
5.10. Applying an Answer . . . . . . . . . . . . . . . . . . . 65
6. Processing RTP/RTCP . . . . . . . . . . . . . . . . . . . . . 68
7. Examples . . . . . . . . . . . . . . . . . . . . . . . . . . 68 7. Examples . . . . . . . . . . . . . . . . . . . . . . . . . . 68
7.1. Simple Example . . . . . . . . . . . . . . . . . . . . . 68 7.1. Simple Example . . . . . . . . . . . . . . . . . . . . . 68
7.2. Normal Examples . . . . . . . . . . . . . . . . . . . . . 72 7.2. Normal Examples . . . . . . . . . . . . . . . . . . . . . 72
8. Security Considerations . . . . . . . . . . . . . . . . . . . 81 8. Security Considerations . . . . . . . . . . . . . . . . . . . 81
9. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 81 9. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 81
10. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 81 10. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 81
11. References . . . . . . . . . . . . . . . . . . . . . . . . . 82 11. References . . . . . . . . . . . . . . . . . . . . . . . . . 82
11.1. Normative References . . . . . . . . . . . . . . . . . . 82 11.1. Normative References . . . . . . . . . . . . . . . . . . 82
11.2. Informative References . . . . . . . . . . . . . . . . . 85 11.2. Informative References . . . . . . . . . . . . . . . . . 85
Appendix A. Appendix A . . . . . . . . . . . . . . . . . . . . . 86 Appendix A. Appendix A . . . . . . . . . . . . . . . . . . . . . 87
Appendix B. Change log . . . . . . . . . . . . . . . . . . . . . 87 Appendix B. Appendix B . . . . . . . . . . . . . . . . . . . . . 88
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 94 Appendix C. Change log . . . . . . . . . . . . . . . . . . . . . 91
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 99
1. Introduction 1. Introduction
This document describes how the W3C WEBRTC RTCPeerConnection This document describes how the W3C WEBRTC RTCPeerConnection
interface [W3C.WD-webrtc-20140617] is used to control the setup, interface [W3C.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
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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
application optionally modifies that offer, and then uses it to set application then uses that offer to set up its local config via the
up its local config via the setLocalDescription() API. The offer is setLocalDescription() API. The offer is finally sent off to the
then sent off to the remote side over its preferred signaling remote side over its preferred signaling mechanism (e.g.,
mechanism (e.g., WebSockets); upon receipt of that offer, the remote WebSockets); upon receipt of that offer, the remote party installs it
party installs it using the setRemoteDescription() API. using the setRemoteDescription() API.
To complete the offer/answer exchange, the remote party uses the To complete the offer/answer exchange, the remote party uses the
createAnswer() API to generate an appropriate answer, applies it createAnswer() API to generate an appropriate answer, applies it
using the setLocalDescription() API, and sends the answer back to the using the setLocalDescription() API, and sends the answer back to the
initiator over the signaling channel. When the initiator gets that initiator over the signaling channel. When the initiator gets that
answer, it installs it using the setRemoteDescription() API, and answer, it installs it using the setRemoteDescription() API, and
initial setup is complete. This process can be repeated for initial setup is complete. This process can be repeated for
additional offer/answer exchanges. additional offer/answer exchanges.
Regarding ICE [RFC5245], JSEP decouples the ICE state machine from Regarding ICE [RFC5245], JSEP decouples the ICE state machine from
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JSEP also allows for an answer to be treated as provisional by the JSEP also allows for an answer to be treated as provisional by the
application. Provisional answers provide a way for an answerer to application. Provisional answers provide a way for an answerer to
communicate initial session parameters back to the offerer, in order communicate initial session parameters back to the offerer, in order
to allow the session to begin, while allowing a final answer to be to allow the session to begin, while allowing a final answer to be
specified later. This concept of a final answer is important to the specified later. This concept of a final answer is important to the
offer/answer model; when such an answer is received, any extra offer/answer model; when such an answer is received, any extra
resources allocated by the caller can be released, now that the exact resources allocated by the caller can be released, now that the exact
session configuration is known. These "resources" can include things session configuration is known. These "resources" can include things
like extra ICE components, TURN candidates, or video decoders. like extra ICE components, 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; as a
results; as a result, multiple dissimilar provisional answers can be result, multiple dissimilar provisional answers, with their own codec
received and applied during call setup. choices, transport parameters, etc., can be received and applied
during call setup. Note that the final answer itself may be
different than any received provisional answers.
In [RFC3264], the constraint at the signaling level is that only one In [RFC3264], the constraint at the signaling level is that only one
offer can be outstanding for a given session, but at the media stack offer can be outstanding for a given session, but at the media stack
level, a new offer can be generated at any point. For example, when level, a new offer can be generated at any point. For example, when
using SIP for signaling, if one offer is sent, then cancelled using a using SIP for signaling, if one offer is sent, then cancelled using a
SIP CANCEL, another offer can be generated even though no answer was SIP CANCEL, another offer can be generated even though no answer was
received for the first offer. To support this, the JSEP media layer received for the first offer. To support this, the JSEP media layer
can provide an offer via the createOffer() method whenever the can provide an offer via the createOffer() method whenever the
Javascript application needs one for the signaling. The answerer can Javascript application needs one for the signaling. The answerer can
send back zero or more provisional answers, and finally end the send back zero or more provisional answers, and finally end the
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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 JSEP's session descriptions use SDP syntax for their internal
SDP messages. While this format is not optimal for manipulation from representation. While this format is not optimal for manipulation
Javascript, it is widely accepted, and frequently updated with new from Javascript, it is widely accepted, and frequently updated with
features. Any alternate encoding of session descriptions would have new features; any alternate encoding of session descriptions would
to keep pace with the changes to SDP, at least until the time that have to keep pace with the changes to SDP, at least until the time
this new encoding eclipsed SDP in popularity. As a result, JSEP that this new encoding eclipsed SDP in popularity.
currently uses SDP as the internal representation for its session
descriptions.
However, to simplify Javascript processing, and provide for future However, to simplify Javascript processing, and provide for future
flexibility, the SDP syntax is encapsulated within a flexibility, the SDP syntax is encapsulated within a
SessionDescription object, which can be constructed from SDP, and be SessionDescription object, which can be constructed from SDP, and be
serialized out to SDP. If future specifications agree on a JSON serialized out to SDP. If future specifications agree on a JSON
format for session descriptions, we could easily enable this object format for session descriptions, we could easily enable this object
to generate and consume that JSON. to generate and consume that JSON.
Other methods may be added to SessionDescription in the future to 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
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RtpTransceivers allow the application to control the RTP media RtpTransceivers allow the application to control the RTP media
associated with one m= section. Each RtpTransceiver has an RtpSender associated with one m= section. Each RtpTransceiver has an RtpSender
and an RtpReceiver, which an application can use to control the and an RtpReceiver, which an application can use to control the
sending and receiving of RTP media. The application may also modify sending and receiving of RTP media. The application may also modify
the RtpTransceiver directly, for instance, by stopping it. the RtpTransceiver directly, for instance, by stopping it.
RtpTransceivers generally have a 1:1 mapping with m= sections, RtpTransceivers generally have a 1:1 mapping with m= sections,
although there may be more RtpTransceivers than m= sections when although there may be more RtpTransceivers than m= sections when
RtpTransceivers are created but not yet associated with a m= section, RtpTransceivers are created but not yet associated with a m= section,
or if RtpTransceivers have been stopped and disassociated from m= or if RtpTransceivers have been stopped and disassociated from m=
sections. An RtpTransceiver is never associated with more than one sections. An RtpTransceiver is said to be associated with an m=
m= section, and once a session description is applied, a m= section section if its mid property is non-null; otherwise it is said to be
is always associated with exactly one RtpTransceiver. disassociated. The associated m= section is determined using a
mapping between transceivers and m= section indices, formed when
creating an offer or applying a remote offer. An RtpTransceiver is
never associated with more than one m= section, and once a session
description is applied, a m= section is always associated with
exactly one RtpTransceiver.
RtpTransceivers can be created explicitly by the application or RtpTransceivers can be created explicitly by the application or
implicitly by calling setRemoteDescription with an offer that adds implicitly by calling setRemoteDescription with an offer that adds
new m= sections. new m= sections.
3.4.2. RtpSenders 3.4.2. RtpSenders
RtpSenders allow the application to control how RTP media is sent. RtpSenders allow the application to control how RTP media is sent.
An RtpSender is conceptually responsible for the outgoing RTP
stream(s) described by an m= section. This includes encoding the
attached MediaStreamTrack, sending RTP media packets, and generating/
processing RTCP for the outgoing RTP streams(s).
3.4.3. RtpReceivers 3.4.3. RtpReceivers
RtpReceivers allows the application to control how RTP media is RtpReceivers allow the application to inspect how RTP media is
received. received. An RtpReceiver is conceptually responsible for the
incoming RTP stream(s) described by an m= section. This includes
processing received RTP media packets, decoding the incoming
stream(s) to produce a remote MediaStreamTrack, and generating/
processing RTCP for the incoming RTP stream(s).
3.5. ICE 3.5. ICE
3.5.1. ICE Gathering Overview 3.5.1. ICE Gathering Overview
JSEP gathers ICE candidates as needed by the application. Collection JSEP gathers ICE candidates as needed by the application. Collection
of ICE candidates is referred to as a gathering phase, and this is of ICE candidates is referred to as a gathering phase, and this is
triggered either by the addition of a new or recycled m= line to the triggered either by the addition of a new or recycled m= section to
local session description, or new ICE credentials in the description, the local session description, or new ICE credentials in the
indicating an ICE restart. Use of new ICE credentials can be description, indicating an ICE restart. Use of new ICE credentials
triggered explicitly by the application, or implicitly by the browser can be triggered explicitly by the application, or implicitly by the
in response to changes in the ICE configuration. browser in response to changes in the ICE configuration.
When the ICE configuration changes in a way that requires a new When the ICE configuration changes in a way that requires a new
gathering phase, a 'needs-ice-restart' bit is set. When this bit is gathering phase, a 'needs-ice-restart' bit is set. When this bit is
set, calls to the createOffer API will generate new ICE credentials. set, calls to the createOffer API will generate new ICE credentials.
This bit is cleared by a call to the setLocalDescription API with new This bit is cleared by a call to the setLocalDescription API with new
ICE credentials from either an offer or an answer, i.e., from either ICE credentials from either an offer or an answer, i.e., from either
a local- or remote-initiated ICE restart. a local- or remote-initiated ICE restart.
When a new gathering phase starts, the ICE Agent will notify the When a new gathering phase starts, the ICE Agent will notify the
application that gathering is occurring through an event. Then, when application that gathering is occurring through an event. Then, when
each new ICE candidate becomes available, the ICE Agent will supply each new ICE candidate becomes available, the ICE Agent will supply
it to the application via an additional event; these candidates will it to the application via an additional event; these candidates will
also automatically be added to the current and/or pending local also automatically be added to the current and/or pending local
session description. Finally, when all candidates have been session description. Finally, when all candidates have been
gathered, an event will be dispatched to signal that the gathering gathered, an event will be dispatched to signal that the gathering
process is complete. process is complete.
Note that gathering phases only gather the candidates needed by Note that gathering phases only gather the candidates needed by
new/recycled/restarting m= lines; other m= lines continue to use new/recycled/restarting m= sections; other m= sections continue to
their existing candidates. Also, when bundling is active, candidates use their existing candidates. Also, when bundling is active,
are only gathered (and exchanged) for the m= lines referenced in candidates are only gathered (and exchanged) for the m= sections
BUNDLE-tags, as described in referenced in BUNDLE-tags, as described in
[I-D.ietf-mmusic-sdp-bundle-negotiation]. [I-D.ietf-mmusic-sdp-bundle-negotiation].
3.5.2. ICE Candidate Trickling 3.5.2. ICE Candidate Trickling
Candidate trickling is a technique through which a caller may Candidate trickling is a technique through which a caller may
incrementally provide candidates to the callee after the initial incrementally provide candidates to the callee after the initial
offer has been dispatched; the semantics of "Trickle ICE" are defined offer has been dispatched; the semantics of "Trickle ICE" are defined
in [I-D.ietf-ice-trickle]. This process allows the callee to begin in [I-D.ietf-ice-trickle]. This process allows the callee to begin
acting upon the call and setting up the ICE (and perhaps DTLS) acting upon the call and setting up the ICE (and perhaps DTLS)
connections immediately, without having to wait for the caller to connections immediately, without having to wait for the caller to
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applications that do not support this feature can simply wait for the applications that do not support this feature can simply wait for the
indication that gathering is complete, and then create and send their indication that gathering is complete, and then create and send their
offer, with all the candidates, at this time. offer, with all the candidates, at this time.
Upon receipt of trickled candidates, the receiving application will Upon receipt of trickled candidates, the receiving application will
supply them to its ICE Agent. This triggers the ICE Agent to start supply them to its ICE Agent. This triggers the ICE Agent to start
using the new remote candidates for connectivity checks. using the new remote candidates for connectivity checks.
3.5.2.1. ICE Candidate Format 3.5.2.1. ICE Candidate Format
As with session descriptions, the syntax of the IceCandidate object In JSEP, ICE candidates are abstracted by an IceCandidate object, and
provides some abstraction, but can be easily converted to and from as with session descriptions, SDP syntax is used for the internal
the SDP candidate lines. representation.
The candidate lines are the only SDP information that is contained The candidate details are specified in an IceCandidate field, using
within IceCandidate, as they represent the only information needed the same SDP syntax as the "candidate-attribute" field defined in
that is not present in the initial offer (i.e., for trickle [RFC5245], Section 15.1. For example:
candidates). This information is carried with the same syntax as the
"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 contains a field to indicate which ICE ufrag
it is associated with, as defined in [RFC5245], Section 15.4. This
value is used to determine which session description (and thereby
which gathering phase) this IceCandidate belongs to, which helps
resolve ambiguities during ICE restarts. If this field is absent in
a received IceCandidate (perhaps when communicating with a non-JSEP
endpoint), the most recently received session description is assumed.
The IceCandidate object also contains fields to indicate which m= The IceCandidate object also contains fields to indicate which m=
line it should be associated with. The m= line can be identified in section it is associated with, which can be identified in one of two
one of two ways; either by a m= line index, or a MID. The m= line ways, either by a m= section index, or a MID. The m= section index
index is a zero-based index, with index N referring to the N+1th m= is a zero-based index, with index N referring to the N+1th m= section
line in the SDP sent by the entity which sent the IceCandidate. The in the session description referenced by this IceCandidate. The MID
MID uses the "media stream identification" attribute, as defined in is a "media stream identification" value, as defined in [RFC5888],
[RFC5888], Section 4, to identify the m= line. JSEP implementations Section 4, which provides a more robust way to identify the m=
creating an ICE Candidate object MUST populate both of these fields, section in the session description, using the MID of the associated
using the MID of the associated RtpTransceiver object (which may be RtpTransceiver object (which may have been locally generated by the
locally generated by the answerer when interacting with a non-JSEP answerer when interacting with a non-JSEP endpoint that does not
remote endpoint that does not support the MID attribute, as discussed support the MID attribute, as discussed in Section 5.9 below). If
in Section 5.9 below). Implementations receiving an ICE Candidate the MID field is present in a received IceCandidate, it MUST be used
object MUST use the MID if present, or the m= line index, if not (the for identification; otherwise, the m= section index is used instead.
non-JSEP remote endpoint case).
When creating an IceCandidate object, JSEP implementations MUST
populate all of these fields.
3.5.3. ICE Candidate Policy 3.5.3. ICE Candidate Policy
Typically, when gathering ICE candidates, the browser will gather all Typically, when gathering ICE candidates, the browser will gather all
possible forms of initial candidates - host, server reflexive, and possible forms of initial candidates - host, server reflexive, and
relay. However, in certain cases, applications may want to have more relay. However, in certain cases, applications may want to have more
specific control over the gathering process, due to privacy or specific control over the gathering process, due to privacy or
related concerns. For example, one may want to suppress the use of related concerns. For example, one may want to only use relay
host candidates, to avoid exposing information about the local candidates, to leak as little location information as possible
network, or go as far as only using relay candidates, to leak as (keeping in mind that this choice comes with corresponding
little location information as possible (note that these choices come operational costs). To accomplish this, JSEP allows the application
with corresponding operational costs). To accomplish this, the to restrict which ICE candidates are used in a session. Note that
browser MUST allow the application to restrict which ICE candidates this filtering is applied on top of any restrictions the browser
are used in a session. Note that this filtering is applied on top of chooses to enforce regarding which IP addresses are permitted for the
any restrictions the browser chooses to enforce regarding which IP application, as discussed in [I-D.ietf-rtcweb-ip-handling].
addresses are permitted for the application, as discussed in
[I-D.ietf-rtcweb-ip-handling].
There may also be cases where the application wants to change which There may also be cases where the application wants to change which
types of candidates are used while the session is active. A prime types of candidates are used while the session is active. A prime
example is where a callee may initially want to use only relay example is where a callee may initially want to use only relay
candidates, to avoid leaking location information to an arbitrary candidates, to avoid leaking location information to an arbitrary
caller, but then change to use all candidates (for lower operational caller, but then change to use all candidates (for lower operational
cost) once the user has indicated they want to take the call. For cost) once the user has indicated they want to take the call. For
this scenario, the browser MUST allow the candidate policy to be this scenario, the browser MUST allow the candidate policy to be
changed in mid-session, subject to the aforementioned interactions changed in mid-session, subject to the aforementioned interactions
with local policy. with local policy.
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ahead of time, it may ask the browser to gather a pool of potential ahead of time, it may ask the browser to gather a pool of potential
ICE candidates to help ensure rapid media setup. ICE candidates to help ensure rapid media setup.
When setLocalDescription is eventually called, and the browser goes When setLocalDescription is eventually called, and the browser goes
to gather the needed ICE candidates, it SHOULD start by checking if to gather the needed ICE candidates, it SHOULD start by checking if
any candidates are available in the pool. If there are candidates in any candidates are available in the pool. If there are candidates in
the pool, they SHOULD be handed to the application immediately via the pool, they SHOULD be handed to the application immediately via
the ICE candidate event. If the pool becomes depleted, either the ICE candidate event. If the pool becomes depleted, either
because a larger-than-expected number of ICE components is used, or because a larger-than-expected number of ICE components is used, or
because the pool has not had enough time to gather candidates, the because the pool has not had enough time to gather candidates, the
remaining candidates are gathered as usual. remaining candidates are gathered as usual. This only occurs for the
first offer/answer exchange, after which the candidate pool is
emptied and no longer used.
One example of where this concept is useful is an application that One example of where this concept is useful is an application that
expects an incoming call at some point in the future, and wants to expects an incoming call at some point in the future, and wants to
minimize the time it takes to establish connectivity, to avoid minimize the time it takes to establish connectivity, to avoid
clipping of initial media. By pre-gathering candidates into the clipping of initial media. By pre-gathering candidates into the
pool, it can exchange and start sending connectivity checks from pool, it can exchange and start sending connectivity checks from
these candidates almost immediately upon receipt of a call. Note these candidates almost immediately upon receipt of a call. Note
though that by holding on to these pre-gathered candidates, which though that by holding on to these pre-gathered candidates, which
will be kept alive as long as they may be needed, the application will be kept alive as long as they may be needed, the application
will consume resources on the STUN/TURN servers it is using. will consume resources on the STUN/TURN servers it is using.
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frame sizes it is capable of receiving. A receiver may have hard frame sizes it is capable of receiving. A receiver may have hard
limits on what its video decoder can process, or it may wish to limits on what its video decoder can process, or it may wish to
constrain what it receives due to application preferences, e.g. a constrain what it receives due to application preferences, e.g. a
specific size for the window in which the video will be displayed. specific size for the window in which the video will be displayed.
Note that certain codecs support transmission of samples with aspect Note that certain codecs support transmission of samples with aspect
ratios other than 1.0 (i.e., non-square pixels). JSEP ratios other than 1.0 (i.e., non-square pixels). JSEP
implementations will not transmit non-square pixels, but SHOULD implementations will not transmit non-square pixels, but SHOULD
receive and render such video with the correct aspect ratio. receive and render such video with the correct aspect ratio.
However, sample aspect ratio has no impact on the size negotiation However, sample aspect ratio has no impact on the size negotiation
described below; all dimensions assume square pixels. described below; all dimensions are measured in pixels, whether
square or not.
3.6.1. Creating an imageattr Attribute 3.6.1. Creating an imageattr Attribute
In order to determine the limits on what video resolution a receiver In order to determine the limits on what video resolution a receiver
wants to receive, it will intersect its decoder hard limits with any wants to receive, it will intersect its decoder hard limits with any
mandatory constraints that have been applied to the associated mandatory constraints that have been applied to the associated
MediaStreamTrack. If the decoder limits are unknown, e.g. when using MediaStreamTrack. If the decoder limits are unknown, e.g. when using
a software decoder, the mandatory constraints are used directly. For a software decoder, the mandatory constraints are used directly. For
the answerer, these mandatory constraints can be applied to the the answerer, these mandatory constraints can be applied to the
remote MediaStreamTracks that are created by a setRemoteDescription remote MediaStreamTracks that are created by a setRemoteDescription
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will be a new offer-answer to communicate them. will be a new offer-answer to communicate them.
If there are no known decoder limits or mandatory constraints, the If there are no known decoder limits or mandatory constraints, the
"a=imageattr" attribute SHOULD be omitted. "a=imageattr" attribute SHOULD be omitted.
Otherwise, an "a=imageattr" attribute is created with "recv" Otherwise, an "a=imageattr" attribute is created with "recv"
direction, and the resulting resolution space formed by intersecting direction, and the resulting resolution space formed by intersecting
the decoder limits and constraints is used to specify its minimum and the decoder limits and constraints is used to specify its minimum and
maximum x= and y= values. If the intersection is the null set, i.e., maximum x= and y= values. If the intersection is the null set, i.e.,
there are no resolutions that are permitted by both the decoder and there are no resolutions that are permitted by both the decoder and
the mandatory constraints, this SHOULD be represented by x=0 and y=0 the mandatory constraints, this MUST be represented by x=0 and y=0
values. values.
The rules here express a single set of preferences, and therefore, The rules here express a single set of preferences, and therefore,
the "a=imageattr" q= value is not important. It SHOULD be set to the "a=imageattr" q= value is not important. It SHOULD be set to
1.0. 1.0.
The "a=imageattr" field is payload type specific. When all video The "a=imageattr" field is payload type specific. When all video
codecs supported have the same capabilities, use of a single codecs supported have the same capabilities, use of a single
attribute, with the wildcard payload type (*), is RECOMMENDED. attribute, with the wildcard payload type (*), is RECOMMENDED.
However, when the supported video codecs have differing capabilities, However, when the supported video codecs have differing capabilities,
specific "a=imageattr" attributes MUST be inserted for each payload specific "a=imageattr" attributes MUST be inserted for each payload
type. type.
As an example, consider a system with a HD-capable, multiformat video As an example, consider a system with a multiformat video decoder,
decoder, where the application has constrained the received track to which is capable of decoding any resolution from 48x48 to 720p, and
at most 360p. In this case, the implementation would generate this where the application has constrained the received track to at most
360p. In this case, the implementation would generate this
attribute: attribute:
a=imageattr:* recv [x=[16:640],y=[16:360],q=1.0] a=imageattr:* recv [x=[48:640],y=[48:360],q=1.0]
This declaration indicates that the receiver is capable of decoding This declaration indicates that the receiver is capable of decoding
any image resolution from 16x16 up to 640x360 pixels. any image resolution from 48x48 up to 640x360 pixels.
3.6.2. Interpreting an imageattr Attribute 3.6.2. Interpreting an imageattr Attribute
[RFC6236] defines "a=imageattr" to be an advisory field. This means [RFC6236] defines "a=imageattr" to be an advisory field. This means
that it does not absolutely constrain the video formats that the that it does not absolutely constrain the video formats that the
sender can use, but gives an indication of the preferred values. sender can use, but gives an indication of the preferred values.
This specification prescribes more specific behavior. When a sender This specification prescribes more specific behavior. When a sender
of a given MediaStreamTrack, which is producing video of a certain of a given MediaStreamTrack, which is producing video of a certain
resolution, receives an "a=imageattr recv" attribute, it MUST check resolution, receives an "a=imageattr recv" attribute, it MUST check
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producing rotated video, the unrotated resolution MUST be used. This producing rotated video, the unrotated resolution MUST be used. This
is required regardless of whether the receiver supports performing is required regardless of whether the receiver supports performing
receive-side rotation (e.g., through CVO), as it significantly receive-side rotation (e.g., through CVO), as it significantly
simplifies the matching logic. simplifies the matching logic.
For the purposes of resolution negotiation, only size limits are For the purposes of resolution negotiation, only size limits are
considered. Any other values, e.g. picture or sample aspect ratio, considered. Any other values, e.g. picture or sample aspect ratio,
MUST be ignored. MUST be ignored.
When communicating with a non-JSEP endpoint, multiple relevant When communicating with a non-JSEP endpoint, multiple relevant
"a=imageattr recv" attributes may be received. If this occurs, "a=imageattr recv" attributes may be present in a received m=
attributes other than the one with the highest "q=" value MUST be section. If this occurs, attributes other than the one with the
ignored. highest "q=" value MUST be ignored. If multiple attributes have the
same "q=" value, those that appear after the first such attribute in
the m= section MUST be ignored.
If an "a=imageattr recv" attribute references a different video codec If an "a=imageattr recv" attribute references a different video
than what has been selected for the MediaStreamTrack, it MUST be payload type than what has been selected for sending the
ignored. MediaStreamTrack, it MUST be ignored.
If the original resolution matches the size limits in the attribute, If the original resolution matches the size limits in the attribute,
the track MUST be transmitted untouched. the track MUST be transmitted untouched.
If the original resolution exceeds the size limits in the attribute, If the original resolution exceeds the size limits in the attribute,
the sender SHOULD apply downscaling to the output of the the sender SHOULD apply downscaling to the output of the
MediaStreamTrack in order to satisfy the limits. Downscaling MUST MediaStreamTrack in order to satisfy the limits. Downscaling MUST
NOT change the track aspect ratio. NOT change the track aspect ratio.
If the original resolution is less than the size limits in the If the original resolution is less than the size limits in the
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If the attribute includes a "sar=" (sample aspect ratio) value set to If the attribute includes a "sar=" (sample aspect ratio) value set to
something other than "1.0", indicating the receiver wants to receive something other than "1.0", indicating the receiver wants to receive
non-square pixels, this cannot be satisfied and the sender MUST NOT non-square pixels, this cannot be satisfied and the sender MUST NOT
transmit the track. transmit the track.
In the special case of receiving a maximum resolution of [0, 0], as In the special case of receiving a maximum resolution of [0, 0], as
described above, the sender MUST NOT transmit the track. described above, the sender MUST NOT transmit the track.
3.7. Simulcast 3.7. Simulcast
JSEP supports simulcast of a MediaStreamTrack, where multiple JSEP supports simulcast transmission of a MediaStreamTrack, where
encodings of the source media can be transmitted within the context multiple encodings of the source media can be transmitted within the
of a single m= section. The current JSEP API is designed to allow context of a single m= section. The current JSEP API is designed to
applications to send simulcasted media but only to receive a single allow applications to send simulcasted media but only to receive a
encoding. This allows for multi-user scenarios where each sending single encoding. This allows for multi-user scenarios where each
client sends multiple encodings to a server, which then, for each sending client sends multiple encodings to a server, which then, for
receiving client, chooses the appropriate encoding to forward. each receiving client, chooses the appropriate encoding to forward.
Applications request support for simulcast by configuring multiple Applications request support for simulcast by configuring multiple
encodings on an RTPSender, which, upon generation of an offer or encodings on an RtpSender, which, upon generation of an offer or
answer, are indicated in SDP markings on the corresponding m= answer, are indicated in SDP markings on the corresponding m=
section, as described below. Receivers that understand simulcast and section, as described below. Receivers that understand simulcast and
are willing to receive it will also include SDP markings to indicate are willing to receive it will also include SDP markings to indicate
their support, and JSEP endpoints will use these markings to their support, and JSEP endpoints will use these markings to
determine whether simulcast is permitted for a given RTPSender. If determine whether simulcast is permitted for a given RtpSender. If
simulcast support is not negotiated, the RTPSender will only use the simulcast support is not negotiated, the RtpSender will only use the
first configured encoding. first configured encoding.
Note that the exact simulcast parameters are up to the sending Note that the exact simulcast parameters are up to the sending
application. While the aforementioned SDP markings are provided to application. While the aforementioned SDP markings are provided to
ensure the remote side can receive and demux multiple simulcast ensure the remote side can receive and demux multiple simulcast
encodings, the specific resolutions and bitrates to be used for each encodings, the specific resolutions and bitrates to be used for each
encoding are purely a send-side decision in JSEP. encoding are purely a send-side decision in JSEP.
JSEP currently does not provide an API to configure receipt of JSEP currently does not provide a mechanism to configure receipt of
simulcast. This means that if simulcast is offered by the remote simulcast. This means that if simulcast is offered by the remote
endpoint, the answer generated by a JSEP endpoint will not indicate endpoint, the answer generated by a JSEP endpoint will not indicate
support for receipt of simulcast, and as such the remote endpoint support for receipt of simulcast, and as such the remote endpoint
will only send a single encoding per m= section. In addition, when will only send a single encoding per m= section.
the JSEP endpoint is the answerer, the permitted encodings for the
RTPSender must be consistent with the offer, but this information is
currently not surfaced through any API. This means that established
simulcast streams will continue to work through a received re-offer,
but setting up initial simulcast by way of a received offer requires
out-of-band signaling or SDP inspection. Future versions of this
specification may add additional APIs to provide this control.
When using JSEP to transmit multiple encodings from a RTPSender, the In addition, JSEP does not provide a mechanism to handle an incoming
offer requesting simulcast from the JSEP endpoint. This means that
established simulcast streams will continue to work through a
received re-offer, but setting up initial simulcast by way of a
received offer requires out-of-band signaling or SDP inspection.
Future versions of this specification may add additional APIs to
provide direct control.
When using JSEP to transmit multiple encodings from a RtpSender, the
techniques from [I-D.ietf-mmusic-sdp-simulcast] and techniques from [I-D.ietf-mmusic-sdp-simulcast] and
[I-D.ietf-mmusic-rid] are used. Specifically, when multiple [I-D.ietf-mmusic-rid] are used. Specifically, when multiple
encodings have been configured for a RTPSender, the m= section for encodings have been configured for a RtpSender, the m= section for
the RTPSender will include an "a=simulcast" attribute, as defined in the RtpSender will include an "a=simulcast" attribute, as defined in
[I-D.ietf-mmusic-sdp-simulcast], Section 6.2, with a "send" simulcast [I-D.ietf-mmusic-sdp-simulcast], Section 6.2, with a "send" simulcast
stream description that lists each desired encoding, and no "recv" stream description that lists each desired encoding, and no "recv"
simulcast stream description. The m= section will also include an simulcast stream description. The m= section will also include an
"a=rid" attribute for each encoding, as specfied in "a=rid" attribute for each encoding, as specified in
[I-D.ietf-mmusic-rid], Section 4; the use of RID identifiers allows [I-D.ietf-mmusic-rid], Section 4; the use of RID identifiers allows
the individual encodings to be disambiguated even though they are all the individual encodings to be disambiguated even though they are all
part of the same m= section. part of the same m= section.
3.8. Interactions With Forking 3.8. Interactions With Forking
Some call signaling systems allow various types of forking where an Some call signaling systems allow various types of forking where an
SDP Offer may be provided to more than one device. For example, SIP SDP Offer may be provided to more than one device. For example, SIP
[RFC3261] defines both a "Parallel Search" and "Sequential Search". [RFC3261] defines both a "Parallel Search" and "Sequential Search".
Although these are primarily signaling level issues that are outside Although these are primarily signaling level issues that are outside
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that it wishes to exchange media with. In JSEP, this offer used in that it wishes to exchange media with. In JSEP, this offer used in
the UPDATE would be formed by simply creating a new PeerConnection the UPDATE would be formed by simply creating a new PeerConnection
and making sure that the same local media streams have been added and making sure that the same local media streams have been added
into this new PeerConnection. Then the new PeerConnection object into this new PeerConnection. Then the new PeerConnection object
would produce a SDP offer that could be used by the signaling to would produce a SDP offer that could be used by the signaling to
perform the UPDATE strategy discussed in [RFC3960]. perform the UPDATE strategy discussed in [RFC3960].
As a result of sharing the media streams, the application will end up As a result of sharing the media streams, the application will end up
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 using setDirection (see
attributes in the descriptions, or the application can choose to play Section 4.2.3), or the application can choose to play out the media
out the media from all sessions mixed together. Of course, if the from all sessions mixed together. Of course, if the application
application wants to only keep a single session, it can simply wants to only keep a single session, it can simply terminate the
terminate the sessions that it no longer needs. sessions that it no longer needs.
4. Interface 4. Interface
This section details the basic operations that must be present to This section details the basic operations that must be present to
implement JSEP functionality. The actual API exposed in the W3C API implement JSEP functionality. The actual API exposed in the W3C API
may have somewhat different syntax, but should map easily to these may have somewhat different syntax, but should map easily to these
concepts. concepts.
4.1. PeerConnection 4.1. PeerConnection
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candidates (including any internal browser filtering) to the candidates (including any internal browser filtering) to the
application, and only use those candidates for connectivity checks. application, and only use those candidates for connectivity checks.
The set of available policies is as follows: The set of available policies is as follows:
all: All candidates permitted by browser policy will be gathered and all: All candidates permitted by browser policy will be gathered and
used. used.
relay: All candidates except relay candidates will be filtered out. relay: All candidates except relay candidates will be filtered out.
This obfuscates the location information that might be ascertained This obfuscates the location information that might be ascertained
by the remote peer from the received candidates. Depending on how by the remote peer from the received candidates. Depending on how
the application deploys its relay servers, this could obfuscate the application deploys and chooses relay servers, this could
location to a metro or possibly even global level. obfuscate location to a metro or possibly even global level.
The default ICE candidate policy MUST be set to "all" as this is The default ICE candidate policy MUST be set to "all" as this is
generally the desired policy, and also typically reduces use of generally the desired policy, and also typically reduces use of
application TURN server resources significantly. application TURN server resources significantly.
If a size is specified for the ICE candidate pool, this indicates the If a size is specified for the ICE candidate pool, this indicates the
number of ICE components to pre-gather candidates for. Because pre- number of ICE components to pre-gather candidates for. Because pre-
gathering results in utilizing STUN/TURN server resources for gathering results in utilizing STUN/TURN server resources for
potentially long periods of time, this must only occur upon potentially long periods of time, this must only occur upon
application request, and therefore the default candidate pool size application request, and therefore the default candidate pool size
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maximize multiplexing, at the cost of less compatibility with maximize multiplexing, at the cost of less compatibility with
legacy endpoints. When acting as answerer, the implementation legacy endpoints. When acting as answerer, the implementation
will reject any m= sections other than the first m= section, will reject any m= sections other than the first m= section,
unless they are in the same bundle group as that m= section. unless they are in the same bundle group as that m= section.
As it provides the best tradeoff between performance and As it provides the best tradeoff between performance and
compatibility with legacy endpoints, the default bundle policy MUST compatibility with legacy endpoints, the default bundle policy MUST
be set to "balanced". be set to "balanced".
The application can specify its preferred policy regarding use of The application can specify its preferred policy regarding use of
RTP/RTCP multiplexing [RFC5761] using one of the following policies: RTP/RTCP multiplexing [RFC5761] using one of the following policies:
negotiate: The browser will gather both RTP and RTCP candidates but negotiate: The browser will gather both RTP and RTCP candidates but
also will offer "a=rtcp-mux", thus allowing for compatibility with also will offer "a=rtcp-mux", thus allowing for compatibility with
either multiplexing or non-multiplexing endpoints. either multiplexing or non-multiplexing endpoints.
require: The browser will only gather RTP candidates. This halves require: The browser will only gather RTP candidates. This halves
the number of candidates that the offerer needs to gather. When the number of candidates that the offerer needs to gather.
acting as answerer, the implementation will reject any m= section Applying a description with an m= section that does not contain an
that does not contain an "a=rtcp-mux" attribute. "a=rtcp-mux" attribute will cause an error to be returned.
The default multiplexing policy MUST be set to "require". The default multiplexing policy MUST be set to "require".
Implementations MAY choose to reject attempts by the application to Implementations MAY choose to reject attempts by the application to
set the multiplexing policy to "negotiate". set the multiplexing policy to "negotiate".
4.1.2. addTrack 4.1.2. addTrack
The addTrack method adds a MediaStreamTrack to the PeerConnection, The addTrack method adds a MediaStreamTrack to the PeerConnection,
using the MediaStream argument to associate the track with other using the MediaStream argument to associate the track with other
tracks in the same MediaStream, so that they can be added to the same tracks in the same MediaStream, so that they can be added to the same
"LS" group when creating an offer or answer. addTrack attempts to "LS" group when creating an offer or answer. addTrack attempts to
minimize the number of transceivers as follows: If the PeerConnection minimize the number of transceivers as follows: If the PeerConnection
is in the "have-remote-offer" state, the track will be attached to is in the "have-remote-offer" state, the track will be attached to
the first compatible transceiver that was created by the most recent the first compatible transceiver that was created by the most recent
call to setRemoteDescription() and does not have a local track. call to setRemoteDescription() and does not have a local track.
Otherwise, a new transceiver will be created, as described in Otherwise, a new transceiver will be created, as described in
Section 4.1.3. Section 4.1.4.
4.1.3. addTransceiver 4.1.3. removeTrack
The addTransceiver method adds a new RTPTransceiver to the The removeTrack method removes a MediaStreamTrack from the
PeerConnection, using the RtpSender argument to indicate which sender
should have its track removed. The sender's track is cleared, and
the sender stops sending. Future calls to createOffer will mark the
media description associated with the sender as recvonly (if
transceiver.currentDirection is sendrecv) or as inactive (if
transceiver.currentDirection is sendonly).
4.1.4. addTransceiver
The addTransceiver method adds a new RtpTransceiver to the
PeerConnection. If a MediaStreamTrack argument is provided, then the PeerConnection. If a MediaStreamTrack argument is provided, then the
transceiver will be configured with that media type and the track transceiver will be configured with that media type and the track
will be attached to the transceiver. Otherwise, the application MUST will be attached to the transceiver. Otherwise, the application MUST
explicitly specify the type; this mode is useful for creating explicitly specify the type; this mode is useful for creating
recvonly transceivers as well as for creating transceivers to which a recvonly transceivers as well as for creating transceivers to which a
track can be attached at some later point. track can be attached at some later point.
At the time of creation, the application can also specify a At the time of creation, the application can also specify a
transceiver direction attribute, a set of MediaStreams which the transceiver direction attribute, a set of MediaStreams which the
transceiver is associated with (allowing LS group assignments), and a transceiver is associated with (allowing LS group assignments), and a
set of encodings for the media (used for simulcast as described in set of encodings for the media (used for simulcast as described in
Section 3.7). Section 3.7).
4.1.4. createDataChannel 4.1.5. createDataChannel
The createDataChannel method creates a new data channel and attaches The createDataChannel method creates a new data channel and attaches
it to the PeerConnection. If no data channel currently exists for it to the PeerConnection. If no data channel currently exists for
this PeerConnection, then a new offer/answer exchange is required. this PeerConnection, then a new offer/answer exchange is required.
All data channels on a given PeerConnection share the same SCTP/DTLS All data channels on a given PeerConnection share the same SCTP/DTLS
association and therefore the same m= section, so subsequent creation association and therefore the same m= section, so subsequent creation
of data channels does not have any impact on the JSEP state. of data channels does not have any impact on the JSEP state.
The createDataChannel method also includes a number of arguments The createDataChannel method also includes a number of arguments
which are used by the PeerConnection (e.g., maxPacketLifetime) but which are used by the PeerConnection (e.g., maxPacketLifetime) but
are not reflected in the SDP and do not affect the JSEP state. are not reflected in the SDP and do not affect the JSEP state.
4.1.5. createOffer 4.1.6. 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 media added to this PeerConnection, the including descriptions of the media added to this PeerConnection, the
codec/RTP/RTCP options supported by this implementation, and any codec/RTP/RTCP options supported by this implementation, and any
candidates that have been gathered by the ICE Agent. An options candidates that have been gathered by the ICE Agent. An options
parameter may be supplied to provide additional control over the parameter may be supplied to provide additional control over the
generated offer. This options parameter allows an application to generated offer. This options parameter allows an application to
trigger an ICE restart, for the purpose of reestablishing trigger an ICE restart, for the purpose of reestablishing
connectivity. connectivity.
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are unaffected by the requested changes, the offer will only contain are unaffected by the requested changes, the offer will only contain
the parameters negotiated by the last offer-answer exchange. The the parameters negotiated by the last offer-answer exchange. The
exact handling of subsequent offer generation is detailed in exact handling of subsequent offer generation is detailed in
Section 5.2.2. below. Section 5.2.2. below.
Session descriptions generated by createOffer must be immediately Session descriptions generated by createOffer must be immediately
usable by setLocalDescription; if a system has limited resources usable by setLocalDescription; if a system has limited resources
(e.g. a finite number of decoders), createOffer should return an (e.g. a finite number of decoders), createOffer should return an
offer that reflects the current state of the system, so that offer that reflects the current state of the system, so that
setLocalDescription will succeed when it attempts to acquire those setLocalDescription will succeed when it attempts to acquire those
resources. Because this method may need to inspect the system state resources.
to determine the currently available resources, it may be implemented
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.
4.1.6. createAnswer 4.1.7. 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 most recent that is compatible with the parameters supplied in the most recent
call to setRemoteDescription, which MUST have been called prior to call to setRemoteDescription, which MUST have been called prior to
calling createAnswer. Like createOffer, the returned blob contains calling createAnswer. Like createOffer, the returned blob contains
descriptions of the media added to this PeerConnection, the descriptions of the media added to this PeerConnection, the
codec/RTP/RTCP options negotiated for this session, and any codec/RTP/RTCP options negotiated for this session, and any
candidates that have been gathered by the ICE Agent. An options candidates that have been gathered by the ICE Agent. An options
parameter may be supplied to provide additional control over the parameter may be supplied to provide additional control over the
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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
description should reflect the current state of the system. Because description should reflect the current state of the system.
this method may need to inspect the system state to determine the
currently available resources, it may need to be implemented 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 trigger candidate gathering or change media credentials, but does not trigger candidate gathering or change media
state. state.
4.1.7. SessionDescriptionType 4.1.8. SessionDescriptionType
Session description objects (RTCSessionDescription) may be of type Session description objects (RTCSessionDescription) may be of type
"offer", "pranswer", "answer" or "rollback". These types provide "offer", "pranswer", "answer" or "rollback". These types provide
information as to how the description parameter should be parsed, and information as to how the description parameter should be parsed, and
how the media state should be changed. how the media state should be changed.
"offer" indicates that a description should be parsed as an offer; "offer" indicates that a description should be parsed as an offer;
said description may include many possible media configurations. A said description may include many possible media configurations. A
description used as an "offer" may be applied anytime the description used as an "offer" may be applied anytime the
PeerConnection is in a stable state, or as an update to a previously PeerConnection is in a stable state, or as an update to a previously
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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 "rollback" is a special session description type implying that the
state machine should be rolled back to the previous stable state, as state machine should be rolled back to the previous stable state, as
described in Section 4.1.7.2. The contents MUST be empty. described in Section 4.1.8.2. The contents MUST be empty.
4.1.7.1. Use of Provisional Answers 4.1.8.1. 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
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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 have finished. 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.
4.1.7.2. Rollback 4.1.8.2. Rollback
In certain situations it may be desirable to "undo" a change made to In certain situations it may be desirable to "undo" a change made to
setLocalDescription or setRemoteDescription. Consider a case where a setLocalDescription or setRemoteDescription. Consider a case where a
call is ongoing, and one side wants to change some of the session call is ongoing, and one side wants to change some of the session
parameters; that side generates an updated offer and then calls parameters; that side generates an updated offer and then calls
setLocalDescription. However, the remote side, either before or setLocalDescription. However, the remote side, either before or
after setRemoteDescription, decides it does not want to accept the after setRemoteDescription, decides it does not want to accept the
new parameters, and sends a reject message back to the offerer. Now, new parameters, and sends a reject message back to the offerer. Now,
the offerer, and possibly the answerer as well, need to return to a the offerer, and possibly the answerer as well, need to return to a
stable state and the previous local/remote description. To support stable state and the previous local/remote description. To support
this, we introduce the concept of "rollback". 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 pending local the state machine to the stable state, and setting the pending local
and/or remote description back to null. Any resources or candidates and/or remote description (see Section 4.1.12 and Section 4.1.14) to
that were allocated by the abandoned local description are discarded; null. Any resources or candidates that were allocated by the
any media that is received will be processed according to the abandoned local description are discarded; any media that is received
previous local and remote descriptions. Rollback can only be used to will be processed according to the previous local and remote
cancel proposed changes; there is no support for rolling back from a descriptions. Rollback can only be used to cancel proposed changes;
stable state to a previous stable state. Note that this implies that there is no support for rolling back from a stable state to a
once the answerer has performed setLocalDescription with his answer, previous stable state. Note that this implies that once the answerer
this cannot be rolled back. has performed setLocalDescription with his answer, this cannot be
rolled back.
A rollback will disassociate any RtpTransceivers that were associated A rollback will disassociate any RtpTransceivers that were associated
with m= sections by the application of the rolled-back session with m= sections by the application of the rolled-back session
description (see Section 5.9 and Section 5.8). This means that some description (see Section 5.9 and Section 5.8). This means that some
RtpTransceivers that were previously associated will no longer be RtpTransceivers that were previously associated will no longer be
associated with any m= section; in such cases, the value of the associated with any m= section; in such cases, the value of the
RtpTransceiver's mid attribute MUST be set to null. RtpTransceivers RtpTransceiver's mid property MUST be set to null, and the mapping
that were created by applying a remote offer that was subsequently between the transceiver and its m= section index MUST be discarded.
rolled back MUST be removed. However, a RtpTransceiver MUST NOT be RtpTransceivers that were created by applying a remote offer that was
removed if the RtpTransceiver's RtpSender was activated by the subsequently rolled back MUST be stopped and removed from the
addTrack method. This is so that an application may call addTrack, PeerConnection. However, a RtpTransceiver MUST NOT be removed if a
then call setRemoteDescription with an offer, then roll back that track was attached to the RtpTransceiver via the addTrack method.
offer, then call createOffer and have a m= section for the added This is so that an application may call addTrack, then call
track appear in the generated offer. setRemoteDescription with an offer, then roll back that offer, then
call createOffer and have a m= section for the added track appear in
the generated offer.
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).
4.1.8. setLocalDescription 4.1.9. setLocalDescription
The setLocalDescription method instructs the PeerConnection to apply The setLocalDescription method instructs the PeerConnection to apply
the supplied session description as its local configuration. The the supplied session description as its local configuration. The
type field indicates whether the description should be processed as type field indicates whether the description should be processed as
an offer, provisional answer, or final answer; offers and answers are an offer, provisional answer, or final answer; offers and answers are
checked differently, using the various rules that exist for each SDP checked differently, using the various rules that exist for each SDP
line. line.
This API changes the local media state; among other things, it sets This API changes the local media state; among other things, it sets
up local resources for receiving and decoding media. In order to up local resources for receiving and decoding media. In order to
successfully handle scenarios where the application wants to offer to successfully handle scenarios where the application wants to offer to
change from one media format to a different, incompatible format, the change from one media format to a different, incompatible format, the
PeerConnection must be able to simultaneously support use of both the PeerConnection must be able to simultaneously support use of both the
current and pending local descriptions (e.g. support codecs that current and pending local descriptions (e.g., support the codecs that
exist in both descriptions) until a final answer is received, at exist in either description). This dual processing begins when the
which point the PeerConnection can fully adopt the pending local PeerConnection enters the have-local-offer state, and continues until
description, or roll back to the current description if the remote setRemoteDescription is called with either a final answer, at which
side denied the change. point the PeerConnection can fully adopt the pending local
description, or a rollback, which results in a revert to the current
local description.
This API indirectly controls the candidate gathering process. When a This API indirectly controls the candidate gathering process. When a
local description is supplied, and the number of transports currently local description is supplied, and the number of transports currently
in use does not match the number of transports needed by the local in use does not match the number of transports needed by the local
description, the PeerConnection will create transports as needed and description, the PeerConnection will create transports as needed and
begin gathering candidates for them. begin gathering candidates for each transport, using ones from the
candidate pool if available.
If setRemoteDescription was previously called with an offer, and If setRemoteDescription was previously called with an offer, and
setLocalDescription is called with an answer (provisional or final), setLocalDescription is called with an answer (provisional or final),
and the media directions are compatible, and media are available to and the media directions are compatible, and media is available to
send, this will result in the starting of media transmission. send, this will result in the starting of media transmission.
4.1.9. setRemoteDescription 4.1.10. setRemoteDescription
The setRemoteDescription method instructs the PeerConnection to apply The setRemoteDescription method instructs the PeerConnection to apply
the supplied session description as the desired remote configuration. the supplied session description as the desired remote configuration.
As in setLocalDescription, the type field of the description As in setLocalDescription, the type field of the description
indicates how it should be processed. indicates how it should be processed.
This API changes the local media state; among other things, it sets This API changes the local media state; among other things, it sets
up local resources for sending and encoding media. up local resources for sending and encoding media.
If setLocalDescription was previously called with an offer, and If setLocalDescription was previously called with an offer, and
setRemoteDescription is called with an answer (provisional or final), setRemoteDescription is called with an answer (provisional or final),
and the media directions are compatible, and media are available to and the media directions are compatible, and media is available to
send, this will result in the starting of media transmission. send, this will result in the starting of media transmission.
4.1.10. currentLocalDescription 4.1.11. currentLocalDescription
The currentLocalDescription method returns a copy of the current The currentLocalDescription method returns the current negotiated
negotiated local description - i.e., the local description from the local description - i.e., the local description from the last
last successful offer/answer exchange - in addition to any local successful offer/answer exchange - in addition to any local
candidates that have been generated by the ICE Agent since the local candidates that have been generated by the ICE Agent since the local
description was set. description was set.
A null object will be returned if an offer/answer exchange has not A null object will be returned if an offer/answer exchange has not
yet been completed. yet been completed.
4.1.11. pendingLocalDescription 4.1.12. pendingLocalDescription
The pendingLocalDescription method returns a copy of the local The pendingLocalDescription method returns a copy of the local
description currently in negotiation - i.e., a local offer set description currently in negotiation - i.e., a local offer set
without any corresponding remote answer - in addition to any local without any corresponding remote answer - in addition to any local
candidates that have been generated by the ICE Agent since the local candidates that have been generated by the ICE Agent since the local
description was set. description was set.
A null object will be returned if the state of the PeerConnection is A null object will be returned if the state of the PeerConnection is
"stable" or "have-remote-offer". "stable" or "have-remote-offer".
4.1.12. currentRemoteDescription 4.1.13. currentRemoteDescription
The currentRemoteDescription method returns a copy of the current The currentRemoteDescription method returns a copy of the current
negotiated remote description - i.e., the remote description from the negotiated remote description - i.e., the remote description from the
last successful offer/answer exchange - in addition to any remote last successful offer/answer exchange - in addition to any remote
candidates that have been supplied via processIceMessage since the candidates that have been supplied via processIceMessage since the
remote description was set. remote description was set.
A null object will be returned if an offer/answer exchange has not A null object will be returned if an offer/answer exchange has not
yet been completed. yet been completed.
4.1.13. pendingRemoteDescription 4.1.14. pendingRemoteDescription
The pendingRemoteDescription method returns a copy of the remote The pendingRemoteDescription method returns a copy of the remote
description currently in negotiation - i.e., a remote offer set description currently in negotiation - i.e., a remote offer set
without any corresponding local answer - in addition to any remote without any corresponding local answer - in addition to any remote
candidates that have been supplied via processIceMessage since the candidates that have been supplied via processIceMessage since the
remote description was set. remote description was set.
A null object will be returned if the state of the PeerConnection is A null object will be returned if the state of the PeerConnection is
"stable" or "have-local-offer". "stable" or "have-local-offer".
4.1.14. canTrickleIceCandidates 4.1.15. canTrickleIceCandidates
The canTrickleIceCandidates property indicates whether the remote The canTrickleIceCandidates property indicates whether the remote
side supports receiving trickled candidates. There are three side supports receiving trickled candidates. There are three
potential values: potential values:
null: No SDP has been received from the other side, so it is not null: No SDP has been received from the other side, so it is not
known if it can handle trickle. This is the initial value before known if it can handle trickle. This is the initial value before
setRemoteDescription() is called. setRemoteDescription() is called.
true: SDP has been received from the other side indicating that it true: SDP has been received from the other side indicating that it
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needed for Trickle ICE. However, applications can use the needed for Trickle ICE. However, applications can use the
canTrickleIceCandidates property to determine whether their peer can canTrickleIceCandidates property to determine whether their peer can
actually do Trickle ICE, i.e., whether it is safe to send an initial actually do Trickle ICE, i.e., whether it is safe to send an initial
offer or answer followed later by candidates as they are gathered. offer or answer followed later by candidates as they are gathered.
As "true" is the only value that definitively indicates remote As "true" is the only value that definitively indicates remote
Trickle ICE support, an application which compares Trickle ICE support, an application which compares
canTrickleIceCandidates against "true" will by default attempt Half canTrickleIceCandidates against "true" will by default attempt Half
Trickle on initial offers and Full Trickle on subsequent interactions Trickle on initial offers and Full Trickle on subsequent interactions
with a Trickle ICE-compatible agent. with a Trickle ICE-compatible agent.
4.1.15. setConfiguration 4.1.16. setConfiguration
The setConfiguration method allows the global configuration of the The setConfiguration method allows the global configuration of the
PeerConnection, which was initially set by constructor parameters, to PeerConnection, which was initially set by constructor parameters, to
be changed during the session. The effects of this method call be changed during the session. The effects of this method call
depend on when it is invoked, and differ depending on which specific depend on when it is invoked, and differ depending on which specific
parameters are changed: parameters are changed:
o Any changes to the STUN/TURN servers to use affect the next o Any changes to the STUN/TURN servers to use affect the next
gathering phase. If an ICE gathering phase has already started or gathering phase. If an ICE gathering phase has already started or
completed, the 'needs-ice-restart' bit mentioned in Section 3.5.1 completed, the 'needs-ice-restart' bit mentioned in Section 3.5.1
will be set. This will cause the next call to createOffer to will be set. This will cause the next call to createOffer to
generate new ICE credentials, for the purpose of forcing an ICE generate new ICE credentials, for the purpose of forcing an ICE
restart and kicking off a new gathering phase, in which the new restart and kicking off a new gathering phase, in which the new
servers will be used. If the ICE candidate pool has a nonzero servers will be used. If the ICE candidate pool has a nonzero
size, any existing candidates will be discarded, and new size, and a local description has not yet been applied, any
candidates will be gathered from the new servers. existing candidates will be discarded, and new candidates will be
gathered from the new servers.
o Any change to the ICE candidate policy affects the next gathering o Any change to the ICE candidate policy affects the next gathering
phase. If an ICE gathering phase has already started or phase. If an ICE gathering phase has already started or
completed, the 'needs-ice-restart' bit will be set. Either way, completed, the 'needs-ice-restart' bit will be set. Either way,
changes to the policy have no effect on the candidate pool, changes to the policy have no effect on the candidate pool,
because pooled candidates are not surfaced to the application because pooled candidates are not surfaced to the application
until a gathering phase occurs, and so any necessary filtering can until a gathering phase occurs, and so any necessary filtering can
still be done on any pooled candidates. still be done on any pooled candidates.
o Any changes to the ICE candidate pool size take effect o The ICE candidate pool size MUST NOT be changed after applying a
local description. If a local description has not yet been
applied, any changes to the ICE candidate pool size take effect
immediately; if increased, additional candidates are pre-gathered; immediately; if increased, additional candidates are pre-gathered;
if decreased, the now-superfluous candidates are discarded. if decreased, the now-superfluous candidates are discarded.
o The bundle and RTCP-multiplexing policies MUST NOT be changed o The bundle and RTCP-multiplexing policies MUST NOT be changed
after the construction of the PeerConnection. after the construction of the PeerConnection.
This call may result in a change to the state of the ICE Agent, and This call may result in a change to the state of the ICE Agent.
may result in a change to media state if it results in connectivity
being established.
4.1.16. addIceCandidate 4.1.17. addIceCandidate
The addIceCandidate method provides a remote candidate to the ICE The addIceCandidate method provides a remote candidate to the ICE
Agent, which, if parsed successfully, will be added to the current Agent, which, if parsed successfully, will be added to the current
and/or pending remote description according to the rules defined for and/or pending remote description according to the rules defined for
Trickle ICE. The pair of MID and ufrag is used to determine the m= Trickle ICE. The pair of MID and ufrag is used to determine the m=
section and ICE candidate generation to which the candidate belongs. section and ICE candidate generation to which the candidate belongs.
If the MID is not present, the m= line index is used to look up the If the MID is not present, the m= section index is used to look up
locally generated MID (see Section 5.9), which is used in place of a the locally generated MID (see Section 5.9), which is used in place
supplied MID. If these values or the candidate string are invalid, of a supplied MID. If these values or the candidate string are
an error is generated. invalid, an error is generated.
The purpose of the ufrag is to resolve ambiguities when trickle ICE The purpose of the ufrag is to resolve ambiguities when trickle ICE
is in progress during an ICE restart. If the ufrag is absent, the is in progress during an ICE restart. If the ufrag is absent, the
candidate MUST be assumed to belong to the most recently applied candidate MUST be assumed to belong to the most recently applied
remote description. Connectivity checks will be sent to the new remote description. Connectivity checks will be sent to the new
candidate. candidate.
This method can also be used to provide an end-of-candidates This method can also be used to provide an end-of-candidates
indication to the ICE Agent, as defined in [I-D.ietf-ice-trickle]). indication to the ICE Agent, as defined in [I-D.ietf-ice-trickle]).
The MID and ufrag are used as described above to determine the m= The MID and ufrag are used as described above to determine the m=
skipping to change at page 31, line 35 skipping to change at page 32, line 21
4.2. RtpTransceiver 4.2. RtpTransceiver
4.2.1. stop 4.2.1. stop
The stop method stops an RtpTransceiver. This will cause future The stop method stops an RtpTransceiver. This will cause future
calls to createOffer to generate a zero port for the associated m= calls to createOffer to generate a zero port for the associated m=
section. See below for more details. section. See below for more details.
4.2.2. stopped 4.2.2. stopped
The stopped method returns "true" if the transceiver has been The stopped property indicates whether the transceiver has been
stopped, either by a call to stopTransceiver or by applying an answer stopped, either by a call to stopTransceiver or by applying an answer
that rejects the associated m= section, and "false" otherwise. that rejects the associated m= section. In either of these cases, it
is set to "true", and otherwise will be set to "false".
A stopped RtpTransceiver does not send any outgoing RTP or RTCP or A stopped RtpTransceiver does not send any outgoing RTP or RTCP or
process any incoming RTP or RTCP. It cannot be restarted. process any incoming RTP or RTCP. It cannot be restarted.
4.2.3. setDirection 4.2.3. setDirection
The setDirection method sets the direction of a transceiver, which The setDirection method sets the direction of a transceiver, which
affects the direction attribute of the associated m= section on affects the direction property of the associated m= section on future
future calls to createOffer and createAnswer. calls to createOffer and createAnswer.
When creating offers, the transceiver direction is directly reflected When creating offers, the transceiver direction is directly reflected
in the output, even for reoffers. When creating answers, the in the output, even for reoffers. When creating answers, the
transceiver direction is intersected with the offered direction, as transceiver direction is intersected with the offered direction, as
explained in the Section 5.3 section below. explained in the Section 5.3 section below.
4.2.4. setCodecPreferences Note that while setDirection sets the direction property of the
transceiver immediately (Section 4.2.4), this property does not
immediately affect whether the transceiver's RtpSender will send or
its RtpReceiver will receive. The direction in effect is represented
by the currentDirection property, which is only updated when an
answer is applied.
4.2.4. direction
The direction property indicates the last value passed into
setDirection. If setDirection has never been called, it is set to
the direction the transceiver was initialized with.
4.2.5. currentDirection
The currentDirection property indicates the last negotiated direction
for the transceiver's associated m= section. More specifically, it
indicates the [RFC3264] directional attribute of the associated m=
section in the last applied answer, with "send" and "recv" directions
reversed if it was a remote answer. For example, if the directional
attribute for the associated m= section in a remote answer is
"recvonly", currentDirection is set to "sendonly".
If an answer that references this transceiver has not yet been
applied, or if the transceiver is stopped, currentDirection is set to
null.
4.2.6. setCodecPreferences
The setCodecPreferences method sets the codec preferences of a The setCodecPreferences method sets the codec preferences of a
transceiver, which in turn affect the presence and order of codecs of transceiver, which in turn affect the presence and order of codecs of
the associated m= section on future calls to createOffer and the associated m= section on future calls to createOffer and
createAnswer. Note that setCodecPreferences does not directly affect createAnswer. Note that setCodecPreferences does not directly affect
which codec the implemtation decides to send. It only affects which which codec the implementation decides to send. It only affects
codecs the implementation indicates that it prefers to receive, via which codecs the implementation indicates that it prefers to receive,
the offer or answer. Even when a codec is excluded by via the offer or answer. Even when a codec is excluded by
setCodecPreferences, it still may be used to send until the next setCodecPreferences, it still may be used to send until the next
offer/answer exchange discards it. offer/answer exchange discards it.
The codec preferences of an RtpTransceiver can cause codecs to be The codec preferences of an RtpTransceiver can cause codecs to be
excluded by subsequent calls to createOffer and createAnswer, in excluded by subsequent calls to createOffer and createAnswer, in
which case the corresponding media formats in the associated m= which case the corresponding media formats in the associated m=
section will be excluded. The codec preferences cannot add media section will be excluded. The codec preferences cannot add media
formats that would otherwise not be present. This includes codecs formats that would otherwise not be present. This includes codecs
that were not negotiated in a previous offer/answer exchange that that were not negotiated in a previous offer/answer exchange that
included the transceiver. included the transceiver.
skipping to change at page 33, line 7 skipping to change at page 34, line 21
All implementations must support these behaviors, but may not use all All implementations must support these behaviors, but may not use all
of them if the remote side, which may not be a JSEP endpoint, does of them if the remote side, which may not be a JSEP endpoint, does
not support them. not support them.
The second set of specifications is the "mandatory-to-use" set. The The second set of specifications is the "mandatory-to-use" set. The
local JSEP endpoint and any remote endpoint must indicate support for local JSEP endpoint and any remote endpoint must indicate support for
these specifications in their session descriptions. these specifications in their session descriptions.
5.1.1. Implementation Requirements 5.1.1. Implementation Requirements
Implementations of JSEP MUST conform to [I-D.ietf-rtcweb-rtp-usage].
This list of mandatory-to-implement specifications is derived from This list of mandatory-to-implement specifications is derived from
the requirements outlined in [I-D.ietf-rtcweb-rtp-usage]. the requirements outlined in that document and from
[I-D.ietf-rtcweb-security-arch].
R-1 [RFC4566] is the base SDP specification and MUST be R-1 [RFC4566] is the base SDP specification and MUST be
implemented. implemented.
R-2 [RFC5764] MUST be supported for signaling the UDP/TLS/RTP/SAVPF R-2 [RFC5764] MUST be supported for signaling the UDP/TLS/RTP/SAVPF
[RFC5764], TCP/DTLS/RTP/SAVPF [RFC5764], TCP/DTLS/RTP/SAVPF [RFC7850], "UDP/DTLS/SCTP"
[I-D.nandakumar-mmusic-proto-iana-registration], "UDP/DTLS/ [I-D.ietf-mmusic-sctp-sdp], and "TCP/DTLS/SCTP"
SCTP" [I-D.ietf-mmusic-sctp-sdp], and "TCP/DTLS/SCTP"
[I-D.ietf-mmusic-sctp-sdp] RTP profiles. [I-D.ietf-mmusic-sctp-sdp] RTP profiles.
R-3 [RFC5245] MUST be implemented for signaling the ICE credentials R-3 [RFC5245] MUST be implemented for signaling the ICE credentials
and candidate lines corresponding to each media stream. The and candidate lines corresponding to each media stream. The
ICE implementation MUST be a Full implementation, not a Lite ICE implementation MUST be a Full implementation, not a Lite
implementation. implementation.
R-4 [RFC5763] MUST be implemented to signal DTLS certificate R-4 [RFC5763] MUST be implemented to signal DTLS certificate
fingerprints. fingerprints.
R-5 [RFC4568] MUST NOT be implemented to signal SDES SRTP keying R-5 [RFC5888] MUST be implemented for signaling grouping
information. information, and MUST be used to identify m= lines via the
a=mid attribute.
R-6 The [RFC5888] grouping framework MUST be implemented for
signaling grouping information, and MUST be used to identify m=
lines via the a=mid attribute.
R-7 [I-D.ietf-mmusic-msid] MUST be supported, in order to signal R-6 [I-D.ietf-mmusic-msid] MUST be supported, in order to signal
associations between RTP objects and W3C MediaStreams and associations between RTP objects and W3C MediaStreams and
MediaStreamTracks in a standard way. MediaStreamTracks in a standard way.
R-8 The bundle mechanism in R-7 The bundle mechanism in
[I-D.ietf-mmusic-sdp-bundle-negotiation] MUST be supported to [I-D.ietf-mmusic-sdp-bundle-negotiation] MUST be supported to
signal the ability to multiplex RTP streams on a single UDP signal the ability to multiplex RTP streams on a single UDP
port, in order to avoid excessive use of port number resources. port, in order to avoid excessive use of port number resources.
R-9 The SDP attributes of "sendonly", "recvonly", "inactive", and R-8 The SDP attributes of "sendonly", "recvonly", "inactive", and
"sendrecv" from [RFC4566] MUST be implemented to signal "sendrecv" from [RFC4566] MUST be implemented to signal
information about media direction. information about media direction.
R-10 [RFC5576] MUST be implemented to signal RTP SSRC values and R-9 [RFC5576] MUST be implemented to signal RTP SSRC values and
grouping semantics. grouping semantics.
R-11 [RFC4585] MUST be implemented to signal RTCP based feedback. R-10 [RFC4585] MUST be implemented to signal RTCP based feedback.
R-12 [RFC5761] MUST be implemented to signal multiplexing of RTP and R-11 [RFC5761] MUST be implemented to signal multiplexing of RTP and
RTCP. RTCP.
R-13 [RFC5506] MUST be implemented to signal reduced-size RTCP R-12 [RFC5506] MUST be implemented to signal reduced-size RTCP
messages. messages.
R-14 [RFC4588] MUST be implemented to signal RTX payload type R-13 [RFC4588] MUST be implemented to signal RTX payload type
associations. associations.
R-15 [RFC3556] with bandwidth modifiers MAY be supported for R-14 [RFC3556] MUST be supported for control of RTCP bandwidth
specifying RTCP bandwidth as a fraction of the media bandwidth, limits.
RTCP fraction allocated to the senders and setting maximum
media bit-rate boundaries.
R-16 TODO: any others? The SDES SRTP keying mechanism from [RFC4568] MUST NOT be
implemented, as discussed in [I-D.ietf-rtcweb-security-arch].
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 ICE, as specified in [RFC5245], MUST be used. Note that the U-1 ICE, as specified in [RFC5245], MUST be used. Note that the
remote endpoint may use a Lite implementation; implementations remote endpoint may use a Lite implementation; implementations
MUST properly handle remote endpoints which do ICE-Lite. MUST properly handle remote endpoints which do ICE-Lite.
R-2 DTLS [RFC6347] or DTLS-SRTP [RFC5763], MUST be used, as U-2 DTLS [RFC6347] or DTLS-SRTP [RFC5763], MUST be used, as
appropriate for the media type, as specified in appropriate for the media type, as specified in
[I-D.ietf-rtcweb-security-arch] [I-D.ietf-rtcweb-security-arch]
5.1.3. Profile Names and Interoperability 5.1.3. Profile Names and Interoperability
For media m= sections, JSEP endpoints MUST support both the "UDP/TLS/ For media m= sections, JSEP endpoints MUST support both the "UDP/TLS/
RTP/SAVPF" and "TCP/DTLS/RTP/SAVPF" profiles and MUST indicate one of RTP/SAVPF" and "TCP/DTLS/RTP/SAVPF" profiles and MUST indicate one of
these two profiles for each media m= line they produce in an offer. these two profiles for each media m= line they produce in an offer.
For data m= sections, JSEP endpoints must support both the "UDP/DTLS/ For data m= sections, JSEP endpoints must support both the "UDP/DTLS/
SCTP" and "TCP/DTLS/SCTP" profiles and MUST indicate one of these two SCTP" and "TCP/DTLS/SCTP" profiles and MUST indicate one of these two
skipping to change at page 35, line 25 skipping to change at page 36, line 41
o Because DTLS-SRTP is REQUIRED, the choice of SAVP or AVP has no o Because DTLS-SRTP is REQUIRED, the choice of SAVP or AVP has no
effect; support for DTLS-SRTP is determined by the presence of one effect; support for DTLS-SRTP is determined by the presence of one
or more "a=fingerprint" attribute. Note that lack of an or more "a=fingerprint" attribute. Note that lack of an
"a=fingerprint" attribute will lead to negotiation failure. "a=fingerprint" attribute will lead to negotiation failure.
o The use of AVPF or AVP simply controls the timing rules used for o The use of AVPF or AVP simply controls the timing rules used for
RTCP feedback. If AVPF is provided, or an "a=rtcp-fb" attribute RTCP feedback. If AVPF is provided, or an "a=rtcp-fb" attribute
is present, assume AVPF timing, i.e., a default value of "trr- is present, assume AVPF timing, i.e., a default value of "trr-
int=0". Otherwise, assume that AVPF is being used in an AVP int=0". Otherwise, assume that AVPF is being used in an AVP
compatible mode and use AVP timing, i.e., "trr-int=4". compatible mode and use a value of "trr-int=4000".
o For data m= sections, JSEP endpoints MUST support receiving the o For data m= sections, JSEP endpoints MUST support receiving the
"UDP/ DTLS/SCTP", "TCP/DTLS/SCTP", or "DTLS/SCTP" (for backwards "UDP/DTLS/SCTP", "TCP/DTLS/SCTP", or "DTLS/SCTP" (for backwards
compatibility) profiles. compatibility) profiles.
Note that re-offers by JSEP endpoints MUST use the correct profile Note that re-offers by JSEP endpoints MUST use the correct profile
strings even if the initial offer/answer exchange used an (incorrect) strings even if the initial offer/answer exchange used an (incorrect)
older profile string. older profile string.
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
skipping to change at page 36, line 28 skipping to change at page 37, line 44
a random number for <sess-id> is sufficient to accomplish this. a random number for <sess-id> is sufficient to accomplish this.
o The third SDP line MUST be a "s=" line, as specified in [RFC4566], o The third SDP line MUST be a "s=" line, as specified in [RFC4566],
Section 5.3; to match the "o=" line, a single dash SHOULD be used Section 5.3; to match the "o=" line, a single dash SHOULD be used
as the session name, e.g. "s=-". Note that this differs from the as the session name, e.g. "s=-". Note that this differs from the
advice in [RFC4566] which proposes a single space, but as both advice in [RFC4566] which proposes a single space, but as both
"o=" and "s=" are meaningless, having the same meaningless value "o=" and "s=" are meaningless, having the same meaningless value
seems clearer. 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="), Repeat Times ("r="), and Time Zones ("z=")
Time Zones ("z=") lines are not useful in this context and SHOULD 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=ice-options" line with the "trickle" option MUST be added, o An "a=ice-options" line with the "trickle" option MUST be added,
as specified in [I-D.ietf-ice-trickle], Section 4. as specified in [I-D.ietf-ice-trickle], Section 4.
The next step is to generate m= sections, as specified in [RFC4566] The next step is to generate m= sections, as specified in [RFC4566]
Section 5.14. An m= section is generated for each RtpTransceiver Section 5.14. An m= section is generated for each RtpTransceiver
that has been added to the PeerConnection. This is done in the order that has been added to the PeerConnection, excluding any stopped
that their associated RtpTransceivers were added to the RtpTransceivers. This is done in the order the RtpTransceivers were
PeerConnection and excludes RtpTransceivers that are stopped and not added to the PeerConnection.
associated with an m= section (either due to an m= section being
recycled or an RtpTransceiver having been stopped before being For each m= section generated for an RtpTransceiver, establish a
associated with an m= section) . mapping between the transceiver and the index of the generated m=
section.
Each m= section, provided it is not marked as bundle-only, MUST Each m= section, provided it is not marked as bundle-only, MUST
generate a unique set of ICE credentials and gather its own unique generate a unique set of ICE credentials and gather its own unique
set of ICE candidates. Bundle-only m= sections MUST NOT contain any set of ICE candidates. Bundle-only m= sections MUST NOT contain any
ICE credentials and MUST NOT gather any candidates. ICE credentials and MUST NOT gather any candidates.
For DTLS, all m= sections MUST use all the certificate(s) that have For DTLS, all m= sections MUST use all the certificate(s) that have
been specified for the PeerConnection; as a result, they MUST all been specified for the PeerConnection; as a result, they MUST all
have the same [I-D.ietf-mmusic-4572-update] fingerprint value(s), or have the same [I-D.ietf-mmusic-4572-update] fingerprint value(s), or
these value(s) MUST be session-level attributes. these value(s) MUST be session-level attributes.
Each m= section should be generated as specified in [RFC4566], Each m= section should be generated as specified in [RFC4566],
Section 5.14. For the m= line itself, the following rules MUST be Section 5.14. For the m= line itself, the following rules MUST be
followed: followed:
o The port value is set to the port of the default ICE candidate for o The port value is set to the port of the default ICE candidate for
this m= section, but given that no candidates have yet been this m= section, but given that no candidates are available yet,
gathered, the "dummy" port value of 9 (Discard) MUST be used, as the "dummy" port value of 9 (Discard) MUST be used, as indicated
indicated in [I-D.ietf-ice-trickle], Section 5.1. in [I-D.ietf-ice-trickle], Section 5.1.
o To properly indicate use of DTLS, the <proto> field MUST be set to o To properly indicate use of DTLS, the <proto> field MUST be set to
"UDP/TLS/RTP/SAVPF", as specified in [RFC5764], Section 8, if the "UDP/TLS/RTP/SAVPF", as specified in [RFC5764], Section 8.
default candidate uses UDP transport, or "TCP/DTLS/RTP/SAVPF", as
specified in [I-D.nandakumar-mmusic-proto-iana-registration] if
the default candidate uses TCP transport.
o If codec preferences have been set for the associated transceiver, o If codec preferences have been set for the associated transceiver,
media formats MUST be generated in the corresponding order, and media formats MUST be generated in the corresponding order, and
MUST exclude any codecs not present in the codec preferences. MUST exclude any codecs not present in the codec preferences.
o The media formats in the answer MAY include codecs present in the
offer that were discarded in a previous offer/answer exchange.
This is necessary for compatibility with third- party call control
and SIP use cases.
o Unless excluded by the above restrictions, the media formats MUST o Unless excluded by the above restrictions, the media formats MUST
include the mandatory audio/video codecs as specified in include the mandatory audio/video codecs as specified in
[I-D.ietf-rtcweb-audio](see Section 3) and [I-D.ietf-rtcweb-audio](see Section 3) and
[I-D.ietf-rtcweb-video](see Section 5). [I-D.ietf-rtcweb-video](see Section 5).
The m= line MUST be followed immediately by a "c=" line, as specified The m= line MUST be followed immediately by a "c=" line, as specified
in [RFC4566], Section 5.7. Again, as no candidates have yet been in [RFC4566], Section 5.7. Again, as no candidates are available
gathered, the "c=" line must contain the "dummy" value "IN IP4 yet, the "c=" line must contain the "dummy" value "IN IP4 0.0.0.0",
0.0.0.0", as defined in [I-D.ietf-ice-trickle], Section 5.1. as defined in [I-D.ietf-ice-trickle], Section 5.1.
[I-D.ietf-mmusic-sdp-mux-attributes] groups SDP attributes into [I-D.ietf-mmusic-sdp-mux-attributes] groups SDP attributes into
different categories. To avoid unnecessary duplication when different categories. To avoid unnecessary duplication when
bundling, Section 8.1 of [I-D.ietf-mmusic-sdp-bundle-negotiation] bundling, Section 8.1 of [I-D.ietf-mmusic-sdp-bundle-negotiation]
specifies that attributes of category IDENTICAL or TRANSPORT should specifies that attributes of category IDENTICAL or TRANSPORT should
not be repeated in bundled m= sections. not be repeated in bundled m= sections.
The following attributes, which are of a category other than The following attributes, which are of a category other than
IDENTICAL or TRANSPORT, MUST be included in each m= section: IDENTICAL or TRANSPORT, MUST be included in each m= section:
o An "a=mid" line, as specified in [RFC5888], Section 4. When o An "a=mid" line, as specified in [RFC5888], Section 4. All MID
generating mid values, it is RECOMMENDED that the values be 3 values MUST be generated in a fashion that does not leak user
bytes or less, to allow them to efficiently fit into the RTP information, e.g., randomly or using a per-PeerConnection counter,
header extension defined in and SHOULD be 3 bytes or less, to allow them to efficiently fit
[I-D.ietf-mmusic-sdp-bundle-negotiation], Section 11. into the RTP header extension defined in
[I-D.ietf-mmusic-sdp-bundle-negotiation], Section 14. Note that
this does not set the RtpTransceiver mid property, as that only
occurs when the description is applied. The generated MID value
can be considered a "proposed" MID at this point.
o A direction attribute which is the same as that of the associated o A direction attribute which is the same as that of the associated
transceiver. transceiver.
o For each media format on the m= line, "a=rtpmap" and "a=fmtp" o For each media format on the m= line, "a=rtpmap" and "a=fmtp"
lines, as specified in [RFC4566], Section 6, and [RFC3264], lines, as specified in [RFC4566], Section 6, and [RFC3264],
Section 5.1. Section 5.1.
o If this m= section is for media with configurable frame sizes, o If this m= section is for media with configurable durations of
e.g. audio, an "a=maxptime" line, indicating the smallest of the media per packet, e.g., audio, an "a=maxptime" line, indicating
maximum supported frame sizes out of all codecs included above, as the maximum amount of media, specified in milliseconds, that can
specified in [RFC4566], Section 6. be encapsulated in each packet, as specified in [RFC4566],
Section 6. This value is set to the smallest of the maximum
duration values across all the codecs included in the m= section.
o If this m= section is for video media, and there are known o If this m= section is for video media, and there are known
limitations on the size of images which can be decoded, an limitations on the size of images which can be decoded, an
"a=imageattr" line, as specified in Section 3.6. "a=imageattr" line, as specified in Section 3.6.
o For each 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 of the primary codec, as specified in [RFC4588], payload type of the primary codec, as specified in [RFC4588],
Section 8.1. Section 8.1.
skipping to change at page 39, line 17 skipping to change at page 40, line 40
* An "a=msid" line, as specified in [I-D.ietf-mmusic-msid], * An "a=msid" line, as specified in [I-D.ietf-mmusic-msid],
Section 2. Section 2.
o If the RtpTransceiver has a sendrecv or sendonly direction, and o If the RtpTransceiver has a sendrecv or sendonly direction, and
the application has specified RID values or has specified more the application has specified RID values or has specified more
than one encoding in the RtpSenders's parameters, an "a=rid" line than one encoding in the RtpSenders's parameters, an "a=rid" line
for each encoding specified. The "a=rid" line is specified in for each encoding specified. The "a=rid" line is specified in
[I-D.ietf-mmusic-rid], and its direction MUST be "send". If the [I-D.ietf-mmusic-rid], and its direction MUST be "send". If the
application has chosen a RID value, it MUST be used as the rid- application has chosen a RID value, it MUST be used as the rid-
identifier; otherwise a RID value MUST be generated by the identifier; otherwise a RID value MUST be generated by the
implementation. When generating RID values, it is RECOMMENDED implementation. RID values MUST be generated in a fashion that
that the values be 3 bytes or less, to allow them to efficiently does not leak user information, e.g., randomly or using a per-
fit into the RTP header extension defined in PeerConnection counter, and SHOULD be 3 bytes or less, to allow
[I-D.ietf-avtext-rid], Section 11. If no encodings have been them to efficiently fit into the RTP header extension defined in
[I-D.ietf-avtext-rid], Section 3. If no encodings have been
specified, or only one encoding is specified but without a RID specified, or only one encoding is specified but without a RID
value, then no "a=rid" lines are generated. value, then no "a=rid" lines are generated.
o If the RtpTransceiver has a sendrecv or sendonly direction and o If the RtpTransceiver has a sendrecv or sendonly direction and
more than one "a=rid" line has been generated, an "a=simulcast" more than one "a=rid" line has been generated, an "a=simulcast"
line, with direction "send", as defined in line, with direction "send", as defined in
[I-D.ietf-mmusic-sdp-simulcast], Section 6.2. The list of RIDs [I-D.ietf-mmusic-sdp-simulcast], Section 6.2. The list of RIDs
MUST include all of the RID identifiers used in the "a=rid" lines MUST include all of the RID identifiers used in the "a=rid" lines
for this m= section. for this m= section.
skipping to change at page 40, line 9 skipping to change at page 41, line 32
clarified for use in DTLS-SRTP scenarios in [RFC5763], Section 5. clarified for use in DTLS-SRTP scenarios in [RFC5763], Section 5.
The role value in the offer MUST be "actpass". The role value in the offer MUST be "actpass".
o An "a=dtls-id" line, as specified in [I-D.ietf-mmusic-dtls-sdp] o An "a=dtls-id" line, as specified in [I-D.ietf-mmusic-dtls-sdp]
Section 5.2. Section 5.2.
o An "a=rtcp" line, as specified in [RFC3605], Section 2.1, o An "a=rtcp" line, as specified in [RFC3605], Section 2.1,
containing the dummy value "9 IN IP4 0.0.0.0", because no containing the dummy value "9 IN IP4 0.0.0.0", because no
candidates have yet been gathered. candidates have yet been gathered.
o An "a=rtcp-mux" line, as specified in [RFC5761], Section 5.1.1. o An "a=rtcp-mux" line, as specified in [RFC5761], Section 5.1.3.
o An "a=rtcp-rsize" line, as specified in [RFC5506], Section 5. o An "a=rtcp-rsize" line, as specified in [RFC5506], Section 5.
Lastly, if a data channel has been created, a m= section MUST be Lastly, if a data channel has been created, 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 "UDP/DTLS/SCTP" if the default and the <proto> field MUST be set to "UDP/DTLS/SCTP"
candidate uses UDP transport, or "TCP/DTLS/SCTP" if the default [I-D.ietf-mmusic-sctp-sdp]. The "fmt" value MUST be set to "webrtc-
candidate uses TCP transport [I-D.ietf-mmusic-sctp-sdp]. The "fmt" datachannel" as specified in [I-D.ietf-mmusic-sctp-sdp], Section 4.1.
value MUST be set to "webrtc-datachannel" as specified in
[I-D.ietf-mmusic-sctp-sdp], Section 4.1.
Within the data m= section, the "a=mid", "a=ice-ufrag", "a=ice-pwd", Within the data m= section, the "a=mid", "a=ice-ufrag", "a=ice-pwd",
"a=fingerprint", "a=dtls-id", and "a=setup" lines MUST be included as "a=fingerprint", "a=dtls-id", and "a=setup" lines MUST be included as
mentioned above, along with an "a=fmtp:webrtc-datachannel" line and mentioned above, along with an "a=fmtp:webrtc-datachannel" line and
an "a=sctp-port" line referencing the SCTP port number as defined in an "a=sctp-port" line referencing the SCTP port number as defined in
[I-D.ietf-mmusic-sctp-sdp], Section 4.1. [I-D.ietf-mmusic-sctp-sdp], Section 4.1.
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
skipping to change at page 41, line 33 skipping to change at page 43, line 8
for generating an initial offer should be followed, subject to the for generating an initial offer should be followed, subject to the
following restriction: following restriction:
o The fields of the "o=" line MUST stay the same except for the o The fields of the "o=" line MUST stay the same except for the
<session-version> field, which MUST increment by one on each call <session-version> field, which MUST increment by one on each call
to createOffer if the offer might differ from the output of the to createOffer if the offer might differ from the output of the
previous call to createOffer; implementations MAY opt to increment previous call to createOffer; implementations MAY opt to increment
<session-version> on every call. The value of the generated <session-version> on every call. The value of the generated
<session-version> is independent of the <session-version> of the <session-version> is independent of the <session-version> of the
current local description; in particular, in the case where the current local description; in particular, in the case where the
current version is N, an offer is created with version N+1, and current version is N, an offer is created and applied with version
then that offer is rolled back so that the current version is N+1, and then that offer is rolled back so that the current
again N, the next generated offer will still have version N+2. version is again N, the next generated offer will still have
version N+2.
Note that if the application creates an offer by reading Note that if the application creates an offer by reading
currentLocalDescription instead of calling createOffer, the returned currentLocalDescription instead of calling createOffer, the returned
SDP may be different than when setLocalDescription was originally SDP may be different than when setLocalDescription was originally
called, due to the addition of gathered ICE candidates, but the called, due to the addition of gathered ICE candidates, but the
<session-version> will not have changed. There are no known <session-version> will not have changed. There are no known
scenarios in which this causes problems, but if this is a concern, scenarios in which this causes problems, but if this is a concern,
the solution is simply to use createOffer to ensure a unique the solution is simply to use createOffer to ensure a unique
<session-version>. <session-version>.
skipping to change at page 42, line 49 skipping to change at page 44, line 26
o Each "a=ice-ufrag" and "a=ice-pwd" line MUST stay the same, unless o Each "a=ice-ufrag" and "a=ice-pwd" line MUST stay the same, unless
the ICE configuration has changed (either changes to the supported the ICE configuration has changed (either changes to the supported
STUN/TURN servers, or the ICE candidate policy), or the STUN/TURN servers, or the ICE candidate policy), or the
"IceRestart" option ( Section 5.2.3.1 was specified. If the m= "IceRestart" option ( Section 5.2.3.1 was specified. If the m=
section is bundled into another m= section, it still MUST NOT section is bundled into another m= section, it still MUST NOT
contain any ICE credentials. contain any ICE credentials.
o If the m= section is not bundled into another m= section, an o If the m= section is not bundled into another m= section, an
"a=rtcp" attribute line MUST be added with of the default RTCP "a=rtcp" attribute line MUST be added with of the default RTCP
candidate, as indicated in [RFC5761], section 5.1.3. candidate, as indicated in [RFC5761], Section 5.1.3.
o If the m= section is not bundled into another m= section, for each o If the m= section is not bundled into another m= section, for each
candidate that has been gathered during the most recent gathering candidate that has been gathered during the most recent gathering
phase (see Section 3.5.1), an "a=candidate" line MUST be added, as phase (see Section 3.5.1), an "a=candidate" line MUST be added, as
defined in [RFC5245], Section 4.3., paragraph 3. If candidate defined in [RFC5245], Section 4.3., paragraph 3. If candidate
gathering for the section has completed, an "a=end-of-candidates" gathering for the section has completed, an "a=end-of-candidates"
attribute MUST be added, as described in [I-D.ietf-ice-trickle], attribute MUST be added, as described in [I-D.ietf-ice-trickle],
Section 9.3. If the m= section is bundled into another m= Section 9.3. If the m= section is bundled into another m=
section, both "a=candidate" and "a=end-of-candidates" MUST be section, both "a=candidate" and "a=end-of-candidates" MUST be
omitted. omitted.
skipping to change at page 43, line 30 skipping to change at page 45, line 7
o If any RtpTransceiver has been stopped, the port MUST be set to o If any RtpTransceiver has been stopped, the port MUST be set to
zero and the "a=msid" line MUST be removed. zero and the "a=msid" line MUST be removed.
o If any RtpTransceiver has been added, and there exists a m= o If any RtpTransceiver has been added, and there exists a m=
section with a zero port in the current local description or the section with a zero port in the current local description or the
current remote description, that m= section MUST be recycled by current remote description, that m= section MUST be recycled by
generating a m= section for the added RtpTransceiver as if the m= generating a m= section for the added RtpTransceiver as if the m=
section were being added to session description, except that section were being added to session description, except that
instead of adding it, the generated m= section replaces the m= instead of adding it, the generated m= section replaces the m=
section with a zero port. section with a zero port. The new m= section MUST contain a new
MID.
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. the "local-offer" state above.
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, if
any:
o The m= line and corresponding "a=rtpmap" and "a=fmtp" lines MUST o The m= line and corresponding "a=rtpmap" and "a=fmtp" lines MUST
only include codecs present in the most recent answer which have only include codecs present in the most recent answer which have
not been excluded by the codec preferences of the associated not been excluded by the codec preferences of the associated
transceiver. transceiver. Note that non-JSEP endpoints are not subject to
these restrictions, and might offer media formats that were not
present in the most recent answer, as specified in [RFC3264],
Section 8. Therefore, JSEP endpoints MUST be prepared to receive
such offers.
o The media formats on the m= line MUST be generated in the same o The media formats on the m= line MUST be generated in the same
order as in the current local description. order as in the current local 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 most recent answer. in the most recent answer.
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 most recent answer. present in the most recent answer.
skipping to change at page 47, line 12 skipping to change at page 48, line 37
o No supported codec is present in the offer. o No supported codec is present in the offer.
o The bundle policy is "max-bundle", and this is not the first m= o The bundle policy is "max-bundle", and this is not the first m=
section or in the same bundle group as the first m= section. section or in the same bundle group as the first m= section.
o The bundle policy is "balanced", and this is not the first m= o The bundle policy is "balanced", and this is not the first m=
section for this media type or in the same bundle group as the section for this media type or in the same bundle group as the
first m= section for this media type. first m= section for this media type.
o The RTP/RTCP multiplexing policy is "require" and the m= section
doesn't contain an "a=rtcp-mux" attribute.
Otherwise, each m= section in the answer should then be generated as Otherwise, each m= section in the answer should then be generated as
specified in [RFC3264], Section 6.1. For the m= line itself, the specified in [RFC3264], Section 6.1. For the m= line itself, the
following rules must be followed: following rules must be followed:
o The port value would normally be set to the port of the default o The port value would normally be set to the port of the default
ICE candidate for this m= section, but given that no candidates ICE candidate for this m= section, but given that no candidates
have yet been gathered, the "dummy" port value of 9 (Discard) MUST are available yet, the "dummy" port value of 9 (Discard) MUST be
be used, as indicated in [I-D.ietf-ice-trickle], Section 5.1. used, as indicated in [I-D.ietf-ice-trickle], Section 5.1.
o The <proto> field MUST be set to exactly match the <proto> field o The <proto> field MUST be set to exactly match the <proto> field
for the corresponding m= line in the offer. for the corresponding m= line in the offer.
o If codec preferences have been set for the associated transceiver, o If codec preferences have been set for the associated transceiver,
media formats MUST be generated in the corresponding order, and media formats MUST be generated in the corresponding order, and
MUST exclude any codecs not present in the codec preferences or MUST exclude any codecs not present in the codec preferences or
not present in the offer. not present in the offer. Note that non-JSEP endpoints are not
subject to this restriction, and might add media formats in the
answer that are not present in the offer, as specified in
[RFC3264], Section 6.1. Therefore, JSEP endpoints MUST be
prepared to receive such answers.
o Unless excluded by the above restrictions, the media formats MUST o Unless excluded by the above restrictions, the media formats MUST
include the mandatory audio/video codecs as specified in include the mandatory audio/video codecs as specified in
[I-D.ietf-rtcweb-audio](see Section 3) and [I-D.ietf-rtcweb-audio](see Section 3) and
[I-D.ietf-rtcweb-video](see Section 5). [I-D.ietf-rtcweb-video](see Section 5).
The m= line MUST be followed immediately by a "c=" line, as specified The m= line MUST be followed immediately by a "c=" line, as specified
in [RFC4566], Section 5.7. Again, as no candidates have yet been in [RFC4566], Section 5.7. Again, as no candidates are available
gathered, the "c=" line must contain the "dummy" value "IN IP4 yet, the "c=" line must contain the "dummy" value "IN IP4 0.0.0.0",
0.0.0.0", as defined in [I-D.ietf-ice-trickle], Section 5.1. as defined in [I-D.ietf-ice-trickle], Section 5.1.
If the offer supports bundle, all m= sections to be bundled must use If the offer supports bundle, all m= sections to be bundled must use
the same ICE credentials and candidates; all m= sections not being the same ICE credentials and candidates; all m= sections not being
bundled must use unique ICE credentials and candidates. Each m= bundled must use unique ICE credentials and candidates. Each m=
section MUST contain the following attributes (which are of attribute section MUST contain the following attributes (which are of attribute
types other than IDENTICAL and TRANSPORT): types other than IDENTICAL and TRANSPORT):
o If and only if present in the offer, an "a=mid" line, as specified o If and only if present in the offer, an "a=mid" line, as specified
in [RFC5888], Section 9.1. The "mid" value MUST match that in [RFC5888], Section 9.1. The "mid" value MUST match that
specified in the offer. specified in the offer.
skipping to change at page 48, line 16 skipping to change at page 49, line 41
the offered direction specified in [RFC3264], Section 6.1, and the offered direction specified in [RFC3264], Section 6.1, and
then intersecting with the direction of the associated then intersecting with the direction of the associated
RtpTransceiver. For example, in the case where an m= section is RtpTransceiver. For example, in the case where an m= section is
offered as "sendonly", and the local transceiver is set to offered as "sendonly", and the local transceiver is set to
"sendrecv", the result in the answer is a "recvonly" direction. "sendrecv", the result in the answer is a "recvonly" direction.
o For each media format on the m= line, "a=rtpmap" and "a=fmtp" o For each media format on the m= line, "a=rtpmap" and "a=fmtp"
lines, as specified in [RFC4566], Section 6, and [RFC3264], lines, as specified in [RFC4566], Section 6, and [RFC3264],
Section 6.1. Section 6.1.
o If this m= section is for media with configurable frame sizes, o If this m= section is for media with configurable durations of
e.g. audio, an "a=maxptime" line, indicating the smallest of the media per packet, e.g., audio, an "a=maxptime" line, as described
maximum supported frame sizes out of all codecs included above, as in Section 5.2.
specified in [RFC4566], Section 6.
o If this m= section is for video media, and there are known o If this m= section is for video media, and there are known
limitations on the size of images which can be decoded, an limitations on the size of images which can be decoded, an
"a=imageattr" line, as specified in Section 3.6. "a=imageattr" line, as specified in Section 3.6.
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 of the primary "a=fmtp" line that references the payload type of the primary
codec, as specified in [RFC4588], Section 8.1. codec, as specified in [RFC4588], Section 8.1.
skipping to change at page 49, line 23 skipping to change at page 50, line 46
Section 15.4. Section 15.4.
o An "a=fingerprint" line for each of the endpoint's certificates, o An "a=fingerprint" line for each of the endpoint's certificates,
as specified in [RFC4572], Section 5; the digest algorithm used as specified in [RFC4572], Section 5; the digest algorithm used
for the fingerprint MUST match that used in the certificate for the fingerprint MUST match that used in the certificate
signature. 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. The role value MUST be consistent "active" role is RECOMMENDED.
with the existing DTLS connection, if one exists and is being
continued.
o An "a=dtls-id" line, as specified in [I-D.ietf-mmusic-dtls-sdp] o An "a=dtls-id" line, as specified in [I-D.ietf-mmusic-dtls-sdp]
Section 5.3. Section 5.3.
o If present in the offer, an "a=rtcp-mux" line, as specified in o If present in the offer, an "a=rtcp-mux" line, as specified in
[RFC5761], Section 5.1.1. Otherwise, an "a=rtcp" line, as [RFC5761], Section 5.1.3. Otherwise, an "a=rtcp" line, as
specified in [RFC3605], Section 2.1, containing the dummy value "9 specified in [RFC3605], Section 2.1, containing the dummy value "9
IN IP4 0.0.0.0" (because no candidates have yet been gathered). IN IP4 0.0.0.0" (because no candidates have yet been gathered).
o If present in the offer, an "a=rtcp-rsize" line, as specified in o If present in the offer, an "a=rtcp-rsize" line, as specified in
[RFC5506], Section 5. [RFC5506], Section 5.
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> and "fmt" fields MUST be set to exactly "application" and the <proto> and "fmt" fields MUST be set to exactly
match the fields in the offer. match the fields in the offer.
skipping to change at page 51, line 5 skipping to change at page 52, line 29
o The media formats on the m= line MUST be generated in the same o The media formats on the m= line MUST be generated in the same
order as in the current local description. order as in the current local description.
o Each "a=ice-ufrag" and "a=ice-pwd" line MUST stay the same, unless o Each "a=ice-ufrag" and "a=ice-pwd" line MUST stay the same, unless
the m= section is restarting, in which case new ICE credentials the m= section is restarting, in which case new ICE credentials
must be created as specified in [RFC5245], Section 9.2.1.1. If must be created as specified in [RFC5245], Section 9.2.1.1. If
the m= section is bundled into another m= section, it still MUST the m= section is bundled into another m= section, it still MUST
NOT contain any ICE credentials. NOT contain any ICE credentials.
o Each "a=setup" line MUST use an "active" or "passive" role value
consistent with the existing DTLS association, if the association
is being continued by the offerer.
o If the m= section is not bundled into another m= section and RTCP o If the m= section is not bundled into another m= section and RTCP
multiplexing is not active, an "a=rtcp" attribute line MUST be multiplexing is not active, an "a=rtcp" attribute line MUST be
filled in with the port and address of the default RTCP candidate. filled in with the port and address of the default RTCP candidate.
If no RTCP candidates have yet been gathered, dummy values MUST be If no RTCP candidates have yet been gathered, dummy values MUST be
used, as described in the initial answer section above. used, as described in the initial answer section above.
o If the m= section is not bundled into another m= section, for each o If the m= section is not bundled into another m= section, for each
candidate that has been gathered during the most recent gathering candidate that has been gathered during the most recent gathering
phase (see Section 3.5.1), an "a=candidate" line MUST be added, as phase (see Section 3.5.1), an "a=candidate" line MUST be added, as
defined in [RFC5245], Section 4.3., paragraph 3. If candidate defined in [RFC5245], Section 4.3., paragraph 3. If candidate
skipping to change at page 51, line 30 skipping to change at page 53, line 11
o For RtpTransceivers that are not stopped, the "a=msid" line MUST o For RtpTransceivers that are not stopped, the "a=msid" line MUST
stay the same. stay the same.
5.3.3. Options Handling 5.3.3. Options Handling
The createAnswer method takes as a parameter an RTCAnswerOptions The createAnswer method takes as a parameter an RTCAnswerOptions
object. The set of parameters for RTCAnswerOptions is different than object. The set of parameters for RTCAnswerOptions is different than
those supported in RTCOfferOptions; the IceRestart option is those supported in RTCOfferOptions; the IceRestart option is
unnecessary, as ICE credentials will automatically be changed for all unnecessary, as ICE credentials will automatically be changed for all
m= lines where the offerer chose to perform ICE restart. m= sections where the offerer chose to perform ICE restart.
The following options are supported in RTCAnswerOptions. The following options are supported in RTCAnswerOptions.
5.3.3.1. VoiceActivityDetection 5.3.3.1. VoiceActivityDetection
Silence suppression in the answer is handled as described in Silence suppression in the answer is handled as described in
Section 5.2.3.2, with one exception: if support for silence Section 5.2.3.2, with one exception: if support for silence
suppression was not indicated in the offer, the suppression was not indicated in the offer, the
VoiceActivityDetection parameter has no effect, and the answer should VoiceActivityDetection parameter has no effect, and the answer should
be generated as if VoiceActivityDetection was set to false. This is be generated as if VoiceActivityDetection was set to false. This is
done on a per-codec basis (e.g., if the offerer somehow offered done on a per-codec basis (e.g., if the offerer somehow offered
support for CN but set "usedtx=0" for Opus, setting support for CN but set "usedtx=0" for Opus, setting
VoiceActivityDetection to true would result in an answer with CN VoiceActivityDetection to true would result in an answer with CN
codecs and "usedtx=0"). codecs and "usedtx=0").
5.4. Modifying an Offer or Answer 5.4. Modifying an Offer or Answer
The SDP returned from createOffer or createAnswer MUST NOT be changed The SDP returned from createOffer or createAnswer MUST NOT be changed
before passing it to setLocalDescription. If precise control over before passing it to setLocalDescription. If precise control over
the SDP is needed, the aformentioned createOffer/createAnswer options the SDP is needed, the aforementioned createOffer/createAnswer
or RTPSender APIs MUST be used. options or RtpTransceiver APIs MUST be used.
Note that the application MAY modify the SDP to reduce the Note that the application MAY modify the SDP to reduce the
capabilities in the offer it sends to the far side (post- capabilities in the offer it sends to the far side (post-
setLocalDescription) or the offer that it installs from the far side setLocalDescription) or the offer that it installs from the far side
(pre-setRemoteDescription), as long as it remains a valid SDP offer (pre-setRemoteDescription), as long as it remains a valid SDP offer
and specifies a subset of what was in the original offer. This is and specifies a subset of what was in the original offer. This is
safe because the answer is not permitted to expand capabilities, and safe because the answer is not permitted to expand capabilities, and
therefore will just respond to what is present in the offer. therefore will just respond to what is present in the offer.
The application SHOULD NOT modify the SDP in the answer it transmits, The application SHOULD NOT modify the SDP in the answer it transmits,
skipping to change at page 57, line 12 skipping to change at page 58, line 35
o Any "a=ssrc" or "a=ssrc-group" attributes MUST be parsed as o Any "a=ssrc" or "a=ssrc-group" attributes MUST be parsed as
specified in [RFC5576], Sections 4.1-4.2, and their values stored. specified in [RFC5576], Sections 4.1-4.2, and their values stored.
o Any "a=extmap" attributes MUST be parsed as specified in o Any "a=extmap" attributes MUST be parsed as specified in
[RFC5285], Section 5, and their values stored. [RFC5285], Section 5, and their values stored.
o Any "a=rtcp-fb" attributes MUST be parsed as specified in o Any "a=rtcp-fb" attributes MUST be parsed as specified in
[RFC4585], Section 4.2., and their values stored. [RFC4585], Section 4.2., and their values stored.
o If present, a single "a=rtcp-mux" attribute MUST be parsed as o If present, a single "a=rtcp-mux" attribute MUST be parsed as
specified in [RFC5761], Section 5.1.1, and its presence or absence specified in [RFC5761], Section 5.1.3, and its presence or absence
flagged and stored. flagged and stored.
o If present, a single "a=rtcp-mux-only" attribute MUST be parsed as o If present, a single "a=rtcp-mux-only" attribute MUST be parsed as
specified in [I-D.ietf-mmusic-mux-exclusive], Section 3, and its specified in [I-D.ietf-mmusic-mux-exclusive], Section 3, and its
presence or absence flagged and stored. presence or absence flagged and stored.
o If present, a single "a=rtcp-rsize" attribute MUST be parsed as o If present, a single "a=rtcp-rsize" attribute MUST be parsed as
specified in [RFC5506], Section 5, and its presence or absence specified in [RFC5506], Section 5, and its presence or absence
flagged and stored. flagged and stored.
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* DTLS fingerprint values, where at least one fingerprint MUST be * DTLS fingerprint values, where at least one fingerprint MUST be
present. present.
o All RID values referenced in an "a=simulcast" line MUST exist as o All RID values referenced in an "a=simulcast" line MUST exist as
"a=rid" lines. "a=rid" lines.
o Each m= section is also checked to ensure prohibited features are o Each m= section is also checked to ensure prohibited features are
not used. If this is a local description, the "ice-lite" not used. If this is a local description, the "ice-lite"
attribute MUST NOT be specified. attribute MUST NOT be specified.
o If the RTP/RTCP multiplexing policy is "require", each m= section
MUST contain an "a=rtcp-mux" attribute.
If this session description is of type "pranswer" or "answer", the If this session description is of type "pranswer" or "answer", the
following additional checks are applied: following additional checks are applied:
o The session description must follow the rules defined in o The session description must follow the rules defined in
[RFC3264], Section 6, including the requirement that the number of [RFC3264], Section 6, including the requirement that the number of
m= sections MUST exactly match the number of m= sections in the m= sections MUST exactly match the number of m= sections in the
associated offer. associated offer.
o For each m= section, the media type and protocol values MUST o For each m= section, the media type and protocol values MUST
exactly match the media type and protocol values in the exactly match the media type and protocol values in the
corresponding m= section in the associated offer. corresponding m= section in the associated offer.
If any of the preceding checks failed, processing MUST stop and an
error MUST be returned.
5.8. Applying a Local Description 5.8. Applying a Local Description
The following steps are performed at the media engine level to apply The following steps are performed at the media engine level to apply
a local description. a local description.
First, the parsed parameters are checked to ensure that they have not First, the parsed parameters are checked to ensure that they are
been altered after their generation in createOffer/createAnswer, as identical to those generated in the last call to createOffer/
discussed in Section 5.4; otherwise, processing MUST stop and an createAnswer, and thus have not been altered, as discussed in
error MUST be returned. Section 5.4; otherwise, processing MUST stop and an error MUST be
returned.
Next, media sections are processed. For each media section, the Next, media sections are processed. For each media section, the
following steps MUST be performed; if any parameters are out of following steps MUST be performed; if any parameters are out of
bounds, or cannot be applied, processing MUST stop and an error MUST bounds, or cannot be applied, processing MUST stop and an error MUST
be returned. be returned.
o If this media section is new, begin gathering candidates for it, o If this media section is new, begin gathering candidates for it,
as defined in [RFC5245], Section 4.1.1, unless it has been marked as defined in [RFC5245], Section 4.1.1, unless it has been marked
as bundle-only. as bundle-only.
o Or, if the ICE ufrag and password values have changed, and it has o Or, if the ICE ufrag and password values have changed, and it has
not been marked as bundle-only, trigger the ICE Agent to start an not been marked as bundle-only, trigger the ICE Agent to start an
ICE restart, and begin gathering new candidates for the media ICE restart, and begin gathering new candidates for the media
section as described in [RFC5245], Section 9.1.1.1. If this section as described in [RFC5245], Section 9.1.1.1. If this
description is an answer, also start checks on that media section description is an answer, also start checks on that media section
as defined in [RFC5245], Section 9.3.1.1. as defined in [RFC5245], Section 9.3.1.1.
o If the media section proto value indicates use of RTP: o If the media section proto value indicates use of RTP:
* If there is no RtpTransceiver associated with this m= section * If there is no RtpTransceiver associated with this m= section
(which should only happen when applying an offer), find one and (which will only happen when applying an offer), find one and
associate it with this m= section according to the following associate it with this m= section according to the following
steps: steps:
+ Find the RtpTransceiver that corresponds to the m= section + Find the RtpTransceiver that corresponds to this m= section,
with the same MID in the created offer. using the mapping between transceivers and m= section
indices established when creating the offer.
+ Set the value of the RtpTransceiver's mid attribute to the + Set the value of this RtpTransceiver's mid property to the
MID of the m= section. MID of the m= section.
* If RTCP mux is indicated, prepare to demux RTP and RTCP from * If RTCP mux is indicated, prepare to demux RTP and RTCP from
the RTP ICE component, as specified in [RFC5761], the RTP ICE component, as specified in [RFC5761],
Section 5.1.1. If RTCP mux is not indicated, but was indicated Section 5.1.3. If RTCP mux is not indicated, but was
in a previous description, this MUST result in an error. previously negotiated, i.e., the RTCP ICE component no longer
exists, this MUST result in an error.
* For each specified RTP header extension, establish a mapping * For each specified RTP header extension, establish a mapping
between the extension ID and URI, as described in section 6 of between the extension ID and URI, as described in section 6 of
[RFC5285]. If any indicated RTP header extension is not [RFC5285]. If any indicated RTP header extension is not
supported, this MUST result in an error. supported, this MUST result in an error.
* If the MID header extension is supported, prepare to demux RTP * If the MID header extension is supported, prepare to demux RTP
data intended for this media section based on the MID header streams intended for this media section based on the MID header
extension, as described in [I-D.ietf-mmusic-msid], Section 3.2. extension, as described in
[I-D.ietf-mmusic-sdp-bundle-negotiation], Section 14.
* For each specified media format, establish a mapping between * For each specified media format, establish a mapping between
the payload type and the actual media format, as described in the payload type and the actual media format, as described in
[RFC3264], Section 6.1. If any indicated media format is not [RFC3264], Section 6.1. If any indicated media format is not
supported, this MUST result in an error. supported, this MUST result in an error.
* For each specified "rtx" media format, establish a mapping * For each specified "rtx" media format, establish a mapping
between the RTX payload type and its associated primary payload between the RTX payload type and its associated primary payload
type, as described in [RFC4588], Sections 8.6 and 8.7. If any type, as described in [RFC4588], Sections 8.6 and 8.7. If any
referenced primary payload types are not present, this MUST referenced primary payload types are not present, this MUST
skipping to change at page 60, line 48 skipping to change at page 62, line 29
The following steps are performed at the media engine level to apply The following steps are performed at the media engine level to apply
a remote description. a remote description.
The following steps MUST be performed for attributes at the session The following steps MUST be performed for attributes at the session
level; if any parameters are out of bounds, or cannot be applied, level; if any parameters are out of bounds, or cannot be applied,
processing MUST stop and an error MUST be returned. processing MUST stop and an error MUST be returned.
o For any specified "CT" bandwidth value, set this as the limit for o For any specified "CT" bandwidth value, set this as the limit for
the maximum total bitrate for all m= sections, as specified in the maximum total bitrate for all m= sections, as specified in
Section 5.8 of [RFC4566]. The implementation can decide how to Section 5.8 of [RFC4566]. Within this overall limit, the
allocate the available bandwidth between m= sections to implementation can dynamically decide how to best allocate the
simultaneously meet any limits on individual m= sections, as well available bandwidth between m= sections, respecting any specific
as this overall session limit. limits that have been specified for individual m= sections.
o For any specified "RR" or "RS" bandwidth values, handle as o For any specified "RR" or "RS" bandwidth values, handle as
specified in [RFC3556], Section 2. specified in [RFC3556], Section 2.
o Any "AS" bandwidth value MUST be ignored, as the meaning of this o Any "AS" bandwidth value MUST be ignored, as the meaning of this
construct at the session level is not well defined. construct at the session level is not well defined.
For each media section, the following steps MUST be performed; if any For each media section, the following steps MUST be performed; if any
parameters are out of bounds, or cannot be applied, processing MUST parameters are out of bounds, or cannot be applied, processing MUST
stop and an error MUST be returned. stop and an error MUST be returned.
skipping to change at page 61, line 41 skipping to change at page 63, line 23
connectivity checks with the appropriate credentials. connectivity checks with the appropriate credentials.
o If an "a=end-of-candidates" attribute is present, process the end- o If an "a=end-of-candidates" attribute is present, process the end-
of-candidates indication as described in [I-D.ietf-ice-trickle] of-candidates indication as described in [I-D.ietf-ice-trickle]
Section 11. Section 11.
o If the media section proto value indicates use of RTP: o If the media section proto value indicates use of RTP:
* If the m= section is being recycled (see Section 5.2.2), * If the m= section is being recycled (see Section 5.2.2),
dissociate the currently associated RtpTransceiver by setting dissociate the currently associated RtpTransceiver by setting
its mid attribute to null. its mid property to null, and discard the mapping between the
transceiver and its m= section index.
* If the m= section is not associated with any RtpTransceiver * If the m= section is not associated with any RtpTransceiver
(possibly because it was dissociated in the previous step), (possibly because it was dissociated in the previous step),
either find an RtpTransceiver or create one according to the either find an RtpTransceiver or create one according to the
following steps: following steps:
+ If the m= section is sendrecv or recvonly, and there are + If the m= section is sendrecv or recvonly, and there are
RtpTransceivers of the same type that were added to the RtpTransceivers of the same type that were added to the
PeerConnection by addTrack and are not associated with any PeerConnection by addTrack and are not associated with any
m= section and are not stopped, find the first (according to m= section and are not stopped, find the first (according to
the canonical order described in Section 5.2.1) such the canonical order described in Section 5.2.1) such
RtpTransceiver. RtpTransceiver.
+ If no RtpTransceiver was found in the previous step, create + If no RtpTransceiver was found in the previous step, create
one with a recvonly direction. one with a recvonly direction.
+ Associate the found or created RtpTransceiver with the m= + Associate the found or created RtpTransceiver with the m=
section by setting the value of the RtpTransceiver's mid section by setting the value of the RtpTransceiver's mid
attribute to the MID of the m= section. If the m= section property to the MID of the m= section, and establish a
does not include a MID (i.e., the remote side does not mapping between the transceiver and the index of the m=
support the MID extension), generate a value for the section. If the m= section does not include a MID (i.e.,
RtpTransceiver mid attribute, following the guidance for the remote endpoint does not support the MID extension),
"a=mid" mentioned in Section 5.2.1. generate a value for the RtpTransceiver mid property,
following the guidance for "a=mid" mentioned in
Section 5.2.1.
* For each specified media format that is also supported by the * For each specified media format that is also supported by the
local implementation, establish a mapping between the specified local implementation, establish a mapping between the specified
payload type and the media format, as described in [RFC3264], payload type and the media format, as described in [RFC3264],
Section 6.1. Specifically, this means that the implementation Section 6.1. Specifically, this means that the implementation
records the payload type to be used in outgoing RTP packets records the payload type to be used in outgoing RTP packets
when sending each specified media format, as well as the when sending each specified media format, as well as the
relative preference for each format that is indicated in their relative preference for each format that is indicated in their
ordering. If any indicated media format is not supported by ordering. If any indicated media format is not supported by
the local implementation, it MUST be ignored. the local implementation, it MUST be ignored.
skipping to change at page 63, line 16 skipping to change at page 64, line 49
constraint on the maximum RTP bitrate to be used when sending constraint on the maximum RTP bitrate to be used when sending
media, as specified in [RFC3890]. If a "TIAS" value is not media, as specified in [RFC3890]. If a "TIAS" value is not
present, but an "AS" value is specified, generate a "TIAS" present, but an "AS" value is specified, generate a "TIAS"
value using this formula: value using this formula:
TIAS = AS * 1000 * 0.95 - 50 * 40 * 8 TIAS = AS * 1000 * 0.95 - 50 * 40 * 8
The 50 is based on 50 packets per second, the 40 is based on an The 50 is based on 50 packets per second, the 40 is based on an
estimate of total header size, the 1000 changes the unit from estimate of total header size, the 1000 changes the unit from
kbps to bps (as required by TIAS), and the 0.95 is to allocate kbps to bps (as required by TIAS), and the 0.95 is to allocate
5% to RTCP. If more accurate control of bandwidth is needed, 5% to RTCP. "TIAS" is used in preference to "AS" because it
"TIAS" should be used instead of "AS". provides more accurate control of bandwidth.
* For any "RR" or "RS" bandwidth values, handle as specified in * For any "RR" or "RS" bandwidth values, handle as specified in
[RFC3556], Section 2. [RFC3556], Section 2.
* Any specified "CT" bandwidth value MUST be ignored, as the * Any specified "CT" bandwidth value MUST be ignored, as the
meaning of this construct at the media level is not well meaning of this construct at the media level is not well
defined. defined.
* If the media section is of type audio: * If the media section is of type audio:
skipping to change at page 64, line 15 skipping to change at page 65, line 46
5.10. Applying an Answer 5.10. Applying an Answer
In addition to the steps mentioned above for processing a local or In addition to the steps mentioned above for processing a local or
remote description, the following steps are performed when processing remote description, the following steps are performed when processing
a description of type "pranswer" or "answer". a description of type "pranswer" or "answer".
For each media section, the following steps MUST be performed: For each media section, the following steps MUST be performed:
o If the media section has been rejected (i.e. port is set to zero o If the media section has been rejected (i.e. port is set to zero
in the answer), stop any reception or transmission of media for in the answer), stop any reception or transmission of media for
this section, and discard any associated ICE components, as this section, and, unless a non-rejected media section is bundled
with this media section, discard any associated ICE components, as
described in Section 9.2.1.3 of [RFC5245]. described in Section 9.2.1.3 of [RFC5245].
o If the remote DTLS fingerprint has been changed or the dtls-id has o If the remote DTLS fingerprint has been changed or the dtls-id has
changed, tear down the DTLS connection. If a DTLS connection changed, tear down the DTLS connection. If a DTLS connection
needs to be torn down but the answer does not indicate an ICE needs to be torn down but the answer does not indicate an ICE
restart, an error MUST be generated. If an ICE restart is restart, an error MUST be generated. If an ICE restart is
performed without a change in dtls-id or fingerprint, then the performed without a change in dtls-id or fingerprint, then the
same DTLS connection is continued over the new ICE channel. same DTLS connection is continued over the new ICE channel.
o If no valid DTLS connection exists, prepare to start a DTLS o If no valid DTLS connection exists, prepare to start a DTLS
connection, using the specified roles and fingerprints, on any connection, using the specified roles and fingerprints, on any
underlying ICE components, once they are active. underlying ICE components, once they are active.
o If the media section proto value indicates use of RTP: o If the media section proto value indicates use of RTP:
* If the media section references any media formats, RTP header * If the media section references any media formats, RTP header
extensions, or RTCP feedback mechanisms that were not present extensions, or RTCP feedback mechanisms that were not present
in the corresponding media section in the offer, this indicates in the corresponding media section in the offer, this indicates
a negotiation problem and MUST result in an error. a negotiation problem and MUST result in an error.
* If the media section has RTCP mux enabled, discard any RTCP * If the media section has RTCP mux enabled, discard the RTCP ICE
component, and begin or continue muxing RTCP over the RTP component, if one exists, and begin or continue muxing RTCP
component, as specified in [RFC5761], Section 5.1.3. over the RTP ICE component, as specified in [RFC5761],
Otherwise, prepare to transmit RTCP over the RTCP component; if Section 5.1.3. Otherwise, prepare to transmit RTCP over the
no RTCP component exists, because RTCP mux was previously RTCP ICE component; if no RTCP ICE component exists, because
enabled, this MUST result in an error. RTCP mux was previously enabled, this MUST result in an error.
* If the media section has reduced-size RTCP enabled, configure * If the media section has reduced-size RTCP enabled, configure
the RTCP transmission for this media section to use reduced- the RTCP transmission for this media section to use reduced-
size RTCP, as specified in [RFC5506]. size RTCP, as specified in [RFC5506].
* If the directional attribute in the answer is of type * If the directional attribute in the answer is of type
"sendrecv" or "sendonly", choose the media format to send as "sendrecv" or "sendonly", choose the media format to send as
the most preferred media format from the remote description the most preferred media format from the remote description
that is also present in the answer, as described in [RFC3264], that is also present in the answer, as described in [RFC3264],
Sections 6.1 and 7, and start transmitting RTP media once the Sections 6.1 and 7, and start transmitting RTP media once the
skipping to change at page 65, line 39 skipping to change at page 67, line 23
using the specified format for resiliency purposes, as using the specified format for resiliency purposes, as
discussed in [I-D.ietf-rtcweb-fec], Section 3.2. Note that discussed in [I-D.ietf-rtcweb-fec], Section 3.2. Note that
unlike RTX or FEC media formats, the "red" format is unlike RTX or FEC media formats, the "red" format is
transmitted on the Source RTP Stream, not the Redundancy RTP transmitted on the Source RTP Stream, not the Redundancy RTP
Stream. Stream.
* Enable the RTCP feedback mechanisms referenced in the media * Enable the RTCP feedback mechanisms referenced in the media
section for all Source RTP Streams using the specified media section for all Source RTP Streams using the specified media
formats. Specifically, begin or continue sending the requested formats. Specifically, begin or continue sending the requested
feedback types and reacting to received feedback, as specified feedback types and reacting to received feedback, as specified
in [RFC4585], Section 4.2. When sending RTCP feedback, use the in [RFC4585], Section 4.2. When sending RTCP feedback, follow
SSRC of an outgoing Source RTP Stream as the RTCP sender SSRC; the rules and recommendations from
if no outgoing Source RTP Stream exists, choose a random one. [I-D.ietf-avtcore-rtp-multi-stream], Section 5.4.1 to select
which SSRC to use.
* If the directional attribute is of type "recvonly" or * If the directional attribute is of type "recvonly" or
"inactive", stop transmitting all RTP media, but continue "inactive", stop transmitting all RTP media, but continue
sending RTCP, as described in [RFC3264], Section 5.1. sending RTCP, as described in [RFC3264], Section 5.1.
o If the media section proto value indicates use of SCTP: o If the media section proto value indicates use of SCTP:
* If no SCTP association yet exists, prepare to initiate a SCTP * If no SCTP association yet exists, prepare to initiate a SCTP
association over the associated ICE component and DTLS association over the associated ICE component and DTLS
connection, using the local SCTP port value from the local connection, using the local SCTP port value from the local
description, and the remote SCTP port value from the remote description, and the remote SCTP port value from the remote
description, as described in [I-D.ietf-mmusic-sctp-sdp], description, as described in [I-D.ietf-mmusic-sctp-sdp],
Section 10.2. Section 10.2.
If the answer contains valid bundle groups, discard any ICE If the answer contains valid bundle groups, discard any ICE
components for the m= sections that will be bundled onto the primary components for the m= sections that will be bundled onto the primary
ICE components in each bundle, and begin muxing these m= sections ICE components in each bundle, and begin muxing these m= sections
accordingly, as described in accordingly, as described in
[I-D.ietf-mmusic-sdp-bundle-negotiation], Section 8.2. [I-D.ietf-mmusic-sdp-bundle-negotiation], Section 8.2.
6. Processing RTP/RTCP packets If the description is of type "answer", and there are still remaining
candidates in the ICE candidate pool, discard them.
Note: The following algorithm does not yet have WG consensus but is
included here as something concrete for the working group to discuss.
When an RTP packet is received by a transport and passes SRTP
authentication, that packet needs to be routed to the correct
RtpReceiver. For each transport, the following steps MUST be
followed to prepare to route packets:
Construct a table mapping MID to RtpReceiver for each RtpReceiver
configured to receive from this transport.
Construct a table mapping incoming SSRC to RtpReceiver for each
RtpReceiver configured to receive from this transport and for each
SSRC that RtpReceiver is configured to receive. Some of the SSRCs
may be present in the m= section corresponding to that RtpReceiver
in the remote description.
Construct a table mapping outgoing SSRC to RtpSender for each
RtpSender configured to transmit from this transport and for each
SSRC that RtpSender is configured to use when sending.
Construct a table mapping payload type to RtpReceiver for each
RtpReceiver configured to receive from this transport and for each
payload type that RtpReceiver is configured to receive. The
payload types of a given RtpReceiver are found in the m= section
corresponding to that RtpReceiver in the local description. If
any payload type could map to more than one RtpReceiver, map to
the RtpReceiver whose m= section appears earliest in the local
description.
As RtpTransceivers (and, thus, RtpReceivers) are added, removed,
stopped, or reconfigured, the tables above must also be updated.
For each RTP packet received, the following steps MUST be followed to
route the packet:
If the packet has a MID and that MID is not in the table mapping
MID to RtpReceiver, drop the packet and stop.
If the packet has a MID and that MID is in the table mapping MID
to RtpReceiver, update the incoming SSRC mapping table to include
an entry that maps the packet's SSRC to the RtpReceiver for that
MID.
If the packet's SSRC is in the incoming SSRC mapping table,
deliver the packet to the associated RtpReceiver and stop.
If the packet's payload type is in the payload type table, update
the the incoming SSRC mapping table to include an entry that maps
the packet's SSRC to the RtpReceiver for that payload type. In
addition, deliver the packet to the associated RtpReceiver and
stop.
Otherwise, drop the packet.
For each RTCP packet received (including each RTCP packet that is
part of a compound RTCP packet), the following type-specific handling
MUST be performed to route the packet:
If the packet is of type SR, and the sender SSRC for the packet is
found in the incoming SSRC table, deliver a copy of the packet to
the RtpReceiver associated with that SSRC. In addition, for each
report block in the report whose SSRC is found in the outgoing
SSRC table, deliver a copy of the RTCP packet to the RtpSender
associated with that SSRC.
If the packet is of type RR, for each report block in the packet
whose SSRC is found in the outgoing SSRC table, deliver a copy of
the RTCP packet to the RtpSender associated with that SSRC.
If the packet is of type SDES, and the sender SSRC for the packet
is found in the incoming SSRC table, deliver the packet to the
RtpReceiver associated with that SSRC. In addition, for each
chunk in the packet that contains a MID that is in the table
mapping MID to RtpReceiver, update the incoming SSRC mapping table
to include an entry that maps the SSRC for that chunk to the
RtpReceiver associated with that MID. (This case can occur when
RTCP for a source is received before any RTP packets.)
If the packet is of type BYE, for each SSRC indicated in the 6. Processing RTP/RTCP
packet that is found in the incoming SSRC table, deliver a copy of
the packet to the RtpReceiver associated with that SSRC.
If the packet is of type RTPFB or PSFB, as defined in [RFC4585], When bundling, associating incoming RTP/RTCP with the proper m=
and the media source SSRC for the packet is found in the outgoing section is defined in [I-D.ietf-mmusic-sdp-bundle-negotiation]. [The
SSRC table, deliver the packet to the RtpSender associated with BUNDLE draft does not currently contain the necessary text to
that SSRC. describe this demux, but when it does it will contain text like that
contained in Appendix B.] When not bundling, the proper m= section
is clear from the ICE component over which the RTP/RTCP is received.
After packets are routed to the RtpReceiver, further processing of Once the proper m= section(s) are known, RTP/RTCP is delivered to the
the RTP packets is done at the RtpReceiver level. This includes RtpTransceiver(s) associated with the m= section(s) and further
using [I-D.ietf-mmusic-rid] to distinguish between multiple Encoded processing of the RTP/RTCP is done at the RtpTransceiver level. This
Streams, as well as determine which Source RTP stream should be includes using RID [I-D.ietf-mmusic-rid] to distinguish between
repaired by a given Redundancy RTP stream. If the RTP packet's PT multiple Encoded Streams, as well as determine which Source RTP
does not match any codec in use by the RtpReceiver, the packet will stream should be repaired by a given Redundancy RTP stream.
be dropped.
7. Examples 7. Examples
Note that this example section shows several SDP fragments. To Note that this example section shows several SDP fragments. To
format in 72 columns, some of the lines in SDP have been split into format in 72 columns, some of the lines in SDP have been split into
multiple lines, where leading whitespace indicates that a line is a multiple lines, where leading whitespace indicates that a line is a
continuation of the previous line. In addition, some blank lines continuation of the previous line. In addition, some blank lines
have been added to improve readability but are not valid in SDP. have been added to improve readability but are not valid in SDP.
More examples of SDP for WebRTC call flows can be found in More examples of SDP for WebRTC call flows can be found in
skipping to change at page 82, line 11 skipping to change at page 82, line 11
Significant text incorporated in the draft as well and review was Significant text incorporated in the draft as well and review was
provided by Peter Thatcher, Taylor Brandstetter, Harald Alvestrand provided by Peter Thatcher, Taylor Brandstetter, Harald Alvestrand
and Suhas Nandakumar. Dan Burnett, Neil Stratford, Anant Narayanan, and Suhas Nandakumar. Dan Burnett, Neil Stratford, Anant Narayanan,
Andrew Hutton, Richard Ejzak, Adam Bergkvist and Matthew Kaufman all Andrew Hutton, Richard Ejzak, Adam Bergkvist and Matthew Kaufman all
provided valuable feedback on this proposal. provided valuable feedback on this proposal.
11. References 11. References
11.1. Normative References 11.1. Normative References
[I-D.ietf-avtcore-rtp-multi-stream]
Lennox, J., Westerlund, M., Wu, Q., and C. Perkins,
"Sending Multiple RTP Streams in a Single RTP Session",
draft-ietf-avtcore-rtp-multi-stream-11 (work in progress),
December 2015.
[I-D.ietf-avtext-rid] [I-D.ietf-avtext-rid]
Roach, A., Nandakumar, S., and P. Thatcher, "RTP Stream Roach, A., Nandakumar, S., and P. Thatcher, "RTP Stream
Identifier (RID) Source Description (SDES)", draft-ietf- Identifier (RID) Source Description (SDES)", draft-ietf-
avtext-rid-00 (work in progress), February 2016. avtext-rid-00 (work in progress), February 2016.
[I-D.ietf-ice-trickle] [I-D.ietf-ice-trickle]
Ivov, E., Rescorla, E., Uberti, J., and P. Saint-Andre, Ivov, E., Rescorla, E., Uberti, J., and P. Saint-Andre,
"Trickle ICE: Incremental Provisioning of Candidates for "Trickle ICE: Incremental Provisioning of Candidates for
the Interactive Connectivity Establishment (ICE) the Interactive Connectivity Establishment (ICE)
Protocol". Protocol".
skipping to change at page 84, line 5 skipping to change at page 84, line 10
[I-D.ietf-rtcweb-security-arch] [I-D.ietf-rtcweb-security-arch]
Rescorla, E., "WebRTC Security Architecture", draft-ietf- Rescorla, E., "WebRTC Security Architecture", draft-ietf-
rtcweb-security-arch-09 (work in progress), February 2014. rtcweb-security-arch-09 (work in progress), February 2014.
[I-D.ietf-rtcweb-video] [I-D.ietf-rtcweb-video]
Roach, A., "WebRTC Video Processing and Codec Roach, A., "WebRTC Video Processing and Codec
Requirements", draft-ietf-rtcweb-video-00 (work in Requirements", draft-ietf-rtcweb-video-00 (work in
progress), July 2014. progress), July 2014.
[I-D.nandakumar-mmusic-proto-iana-registration]
Nandakumar, S., "IANA registration of SDP 'proto'
attribute for transporting RTP Media over TCP under
various RTP profiles.", September 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, June
skipping to change at page 85, line 30 skipping to change at page 85, line 30
Attributes in the Session Description Protocol (SDP)", Attributes in the Session Description Protocol (SDP)",
RFC 6236, May 2011. RFC 6236, May 2011.
[RFC6347] Rescorla, E. and N. Modadugu, "Datagram Transport Layer [RFC6347] Rescorla, E. and N. Modadugu, "Datagram Transport Layer
Security Version 1.2", RFC 6347, January 2012. Security Version 1.2", RFC 6347, January 2012.
[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, April
2013. 2013.
[RFC7850] Nandakumar, S., "Registering Values of the SDP 'proto'
Field for Transporting RTP Media over TCP under Various
RTP Profiles", RFC 7850, DOI 10.17487/RFC7850, April 2016,
<http://www.rfc-editor.org/info/rfc7850>.
11.2. Informative References 11.2. Informative References
[I-D.ietf-rtcweb-ip-handling] [I-D.ietf-rtcweb-ip-handling]
Uberti, J. and G. Shieh, "WebRTC IP Address Handling Uberti, J. and G. Shieh, "WebRTC IP Address Handling
Recommendations", draft-ietf-rtcweb-ip-handling-01 (work Recommendations", draft-ietf-rtcweb-ip-handling-01 (work
in progress), March 2016. in progress), March 2016.
[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), July
2013. 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.
[RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V.
Jacobson, "RTP: A Transport Protocol for Real-Time
Applications", STD 64, RFC 3550, DOI 10.17487/RFC3550,
July 2003, <http://www.rfc-editor.org/info/rfc3550>.
[RFC3556] Casner, S., "Session Description Protocol (SDP) Bandwidth [RFC3556] Casner, S., "Session Description Protocol (SDP) Bandwidth
Modifiers for RTP Control Protocol (RTCP) Bandwidth", Modifiers for RTP Control Protocol (RTCP) Bandwidth",
RFC 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
skipping to change at page 86, line 41 skipping to change at page 87, line 5
[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.
[RFC6464] Lennox, J., Ed., Ivov, E., and E. Marocco, "A Real-time [RFC6464] Lennox, J., Ed., Ivov, E., and E. Marocco, "A Real-time
Transport Protocol (RTP) Header Extension for Client-to- Transport Protocol (RTP) Header Extension for Client-to-
Mixer Audio Level Indication", RFC 6464, Mixer Audio Level Indication", RFC 6464,
DOI 10.17487/RFC6464, December 2011, DOI 10.17487/RFC6464, December 2011,
<http://www.rfc-editor.org/info/rfc6464>. <http://www.rfc-editor.org/info/rfc6464>.
[RFC7656] Lennox, J., Gross, K., Nandakumar, S., Salgueiro, G., and
B. Burman, Ed., "A Taxonomy of Semantics and Mechanisms
for Real-Time Transport Protocol (RTP) Sources", RFC 7656,
DOI 10.17487/RFC7656, November 2015,
<http://www.rfc-editor.org/info/rfc7656>.
[W3C.WD-webrtc-20140617] [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-webrtc- Browsers", World Wide Web Consortium WD WD-webrtc-
20140617, June 2014, 20140617, June 2014,
<http://www.w3.org/TR/2011/WD-webrtc-20140617>. <http://www.w3.org/TR/2011/WD-webrtc-20140617>.
Appendix A. Appendix A Appendix A. Appendix A
For the syntax validation performed in Section 5.7, the following For the syntax validation performed in Section 5.7, the following
skipping to change at page 87, line 44 skipping to change at page 88, line 44
| option) | | | option) | |
| msid | [I-D.ietf-mmusic-msid] Section 2 | | msid | [I-D.ietf-mmusic-msid] Section 2 |
| rid | [I-D.ietf-mmusic-rid] Section 10 | | rid | [I-D.ietf-mmusic-rid] Section 10 |
| simulcast | [I-D.ietf-mmusic-sdp-simulcast]Section | | simulcast | [I-D.ietf-mmusic-sdp-simulcast]Section |
| | 6.1 | | | 6.1 |
| dtls-id | [I-D.ietf-mmusic-dtls-sdp]Section 4 | | dtls-id | [I-D.ietf-mmusic-dtls-sdp]Section 4 |
+-----------------------+-------------------------------------------+ +-----------------------+-------------------------------------------+
Table 1: SDP ABNF References Table 1: SDP ABNF References
Appendix B. Change log Appendix B. Appendix B
The following text is meant to completely replace section
"Associating RTP/RTCP Streams With Correct SDP Media Description" of
[I-D.ietf-mmusic-sdp-bundle-negotiation].
As described in [RFC3550], RTP/RTCP packets are associated with RTP
streams as defined in [RFC7656]. Each RTP stream is identified by an
SSRC value, and each RTP/RTCP packet carries an SSRC value that is
used to associate the packet with the correct RTP stream. An RTCP
packet can carry multiple SSRC values, and might therefore be
associated with multiple RTP streams.
In order to be able to process received RTP/RTCP packets correctly it
must be possible to associate an RTP stream with the correct "m="
line, as the "m=" line and SDP attributes associated with the "m="
line contain information needed to process the packets.
As all RTP streams associated with a BUNDLE group use the same
address:port combination for sending and receiving RTP/RTCP packets,
the local address:port combination cannot be used to associate an RTP
stream with the correct "m=" line. In addition, multiple RTP streams
might be associated with the same "m=" line.
An offerer and answerer can inform each other which SSRC values they
will use for an RTP stream by using the SDP 'ssrc' attribute
[RFC5576]. However, an offerer will not know which SSRC values the
answerer will use until the offerer has received the answer providing
that information. Due to this, before the offerer has received the
answer, the offerer will not be able to associate an RTP stream with
the correct "m=" line using the SSRC value associated with the RTP
stream. In addition, the offerer and answerer may start using new
SSRC values mid-session, without informing each other using the SDP
'ssrc' attribute.
In order for an offerer and answerer to always be able to associate
an RTP stream with the correct "m=" line, the offerer and answerer
using the BUNDLE extension MUST support the mechanism defined in
[I-D.ietf-mmusic-sdp-bundle-negotiation] section 14. where the
offerer and answerer insert the identification-tag associated with an
"m=" line (provided by the remote peer) into RTP and RTCP packets
associated with a BUNDLE group.
The mapping from an SSRC to an identification-tag is carried in RTCP
SDES packets or in RTP header extensions
([I-D.ietf-mmusic-sdp-bundle-negotiation] section 14). Since a
compound RTCP packet can contain multiple RTCP SDES packets, and each
RTCP SDES packet can contain multiple chunks, an RTCP packet can
contain several SSRC to identification-tag mappings. The offerer and
answerer maintain tables used for routing that are updated each time
an RTP/RTCP packet contains new information that affects how packets
should be routed.
To prepare for demultiplexing RTP packets to the correct "m=" line,
the following steps MUST be followed for each BUNDLE group.
Construct a table mapping MID to "m=" line for each "m=" line in
this BUNDLE group. Note that an "m=" line may only have one MID.
Construct a table mapping incoming SSRC to "m=" line for each "m="
line in this BUNDLE group and for each SSRC configured for
receiving in that "m=" line.
Construct a table mapping outgoing SSRC to "m=line" for each "m="
line in this BUNDLE group and for each SSRC configured for sending
in that "m=" line.
Construct a table mapping payload type to "m=" line for each "m="
line in the BUNDLE group and for each payload type configured for
receiving in that "m=" line. If any payload type is configured
for receiving in more than one "m=" line in the BUNDLE group, do
not it include it in the table.
Note that for each of these tables, there can only be one mapping
for any given key (MID, SSRC, or PT). In other words, the tables
are not multimaps.
As "m=" lines are added or removed from the BUNDLE groups, or their
configurations are changed, the tables above MUST also be updated.
For each RTP packet received, the following steps MUST be followed to
route the packet to the correct "m=" section within a BUNDLE group.
Note that the phrase 'deliver a packet to the "m=" line' means to
further process the packet as would normally happen with RTP/RTCP, if
it were received on a transport associated with that "m=" line
outside of a BUNDLE group (i.e., if the "m=" line were not BUNDLEd),
including dropping an RTP packet if the packet's PT does not match
any PT in the "m=" line.
If the packet has a MID and that MID is not in the table mapping
MID to "m=" line, drop the packet and stop.
If the packet has a MID and that MID is in the table mapping MID
to "m=" line, update the incoming SSRC mapping table to include an
entry that maps the packet's SSRC to the "m=" line for that MID.
If the packet's SSRC is in the incoming SSRC mapping table, route
the packet to the associated "m=" line and stop.
If the packet's payload type is in the payload type table, update
the the incoming SSRC mapping table to include an entry that maps
the packet's SSRC to the "m=" line for that payload type. In
addition, route the packet to the associated "m=" line and stop.
Otherwise, drop the packet.
For each RTCP packet received (including each RTCP packet that is
part of a compound RTCP packet), the packet MUST be routed to the
appropriate handler for the SSRCs it contains information about.
Some examples of such handling are given below.
If the packet is of type SR, and the sender SSRC for the packet is
found in the incoming SSRC table, deliver a copy of the packet to
the "m=" line associated with that SSRC. In addition, for each
report block in the report whose SSRC is found in the outgoing
SSRC table, deliver a copy of the RTCP packet to the "m=" line
associated with that SSRC.
If the packet is of type RR, for each report block in the packet
whose SSRC is found in the outgoing SSRC table, deliver a copy of
the RTCP packet to the "m=" line associated with that SSRC.
If the packet is of type SDES, and the sender SSRC for the packet
is found in the incoming SSRC table, deliver the packet to the
"m=" line associated with that SSRC. In addition, for each chunk
in the packet that contains a MID that is in the table mapping MID
to "m=" line, update the incoming SSRC mapping table to include an
entry that maps the SSRC for that chunk to the "m=" line
associated with that MID. (This case can occur when RTCP for a
source is received before any RTP packets.)
If the packet is of type BYE, for each SSRC indicated in the
packet that is found in the incoming SSRC table, deliver a copy of
the packet to the "m=" line associated with that SSRC.
If the packet is of type RTPFB or PSFB, as defined in [RFC4585],
and the media source SSRC for the packet is found in the outgoing
SSRC table, deliver the packet to the "m=" line associated with
that SSRC.
Appendix C. Change log
Note: This section will be removed by RFC Editor before publication. Note: This section will be removed by RFC Editor before publication.
Changes in draft-18:
o Update demux algorithm and move it to an appendix in preparation
for merging it into BUNDLE.
o Clarify why we can't handle an incoming offer to send simulcast.
o Expand IceCandidate object text.
o Further document use of ICE candidate pool.
o Document removeTrack.
o Update requirements to only accept the last generated offer/answer
as an argument to setLocalDescription.
o Allow round pixels.
o Fix code around default timing when AVPF is not specified.
o Clean up terminology around m= line and m=section.
o Provide a more realistic example for minimum decoder capabilities.
o Document behavior when rtcp-mux policy is require but rtcp-mux
attribute not provided.
o Expanded discussion of RtpSender and RtpReceiver.
o Add RtpTransceiver.currentDirection and document setDirection.
o Require imageattr x=0, y=0 to indicate that there are no valid
resolutions.
o Require a privacy-preserving MID/RID construction.
o Require support for RFC 3556 bandwidth modifiers.
o Update maxptime description.
o Note that endpoints may encounter extra codecs in answers and
subsequent offers from non-JSEP peers.
o Update references.
Changes in draft-17: Changes in draft-17:
o Split createOffer and createAnswer sections to clearly indicate o Split createOffer and createAnswer sections to clearly indicate
attributes which always appear and which only appear when not attributes which always appear and which only appear when not
bundled into another m= section. bundled into another m= section.
o Add descriptions of RtpTransceiver methods. o Add descriptions of RtpTransceiver methods.
o Describe how to process RTCP feedback attributes. o Describe how to process RTCP feedback attributes.
skipping to change at page 90, line 18 skipping to change at page 95, line 12
o Remove unused references. o Remove unused references.
o Remove text advocating use of unilateral PTs. o Remove text advocating use of unilateral PTs.
o Trigger an ICE restart even if the ICE candidate policy is being o Trigger an ICE restart even if the ICE candidate policy is being
made more strict. made more strict.
o Remove the 'public' ICE candidate policy. o Remove the 'public' ICE candidate policy.
o Move open issues/TODOs into GitHub issues. o Move open issues into GitHub issues.
o Split local/remote description accessors into current/pending. o Split local/remote description accessors into current/pending.
o Clarify a=imageattr handling. o Clarify a=imageattr handling.
o Add more detail on VoiceActivityDetection handling. o Add more detail on VoiceActivityDetection handling.
o Reference draft-shieh-rtcweb-ip-handling. o Reference draft-shieh-rtcweb-ip-handling.
o Make it clear when an ICE restart should occur. o Make it clear when an ICE restart should occur.
o Resolve reference TODOs. o Resolve changes needed in references.
o Remove MSID semantics. o Remove MSID semantics.
o ice-options are now at session level. o ice-options are now at session level.
o Default RTCP mux policy is now 'require'. o Default RTCP mux policy is now 'require'.
Changes in draft-12: Changes in draft-12:
o Filled in sections on applying local and remote descriptions. o Filled in sections on applying local and remote descriptions.
skipping to change at page 91, line 17 skipping to change at page 96, line 11
Changes in draft-11: Changes in draft-11:
o Clarified handling of RTP CNAMEs. o Clarified handling of RTP CNAMEs.
o Updated what SDP lines should be processed or ignored. o Updated what SDP lines should be processed or ignored.
o Specified how a=imageattr should be used. o Specified how a=imageattr should be used.
Changes in draft-10: Changes in draft-10:
o TODO o Described video size negotiation with imageattr.
o Clarified rejection of sections that do not have mux-only.
o Add handling of LS groups
Changes in draft-09: Changes in draft-09:
o Don't return null for {local,remote}Description after close(). o Don't return null for {local,remote}Description after close().
o Changed TCP/TLS to UDP/DTLS in RTP profile names. o Changed TCP/TLS to UDP/DTLS in RTP profile names.
o Separate out bundle and mux policy. o Separate out bundle and mux policy.
o Added specific references to FEC mechanisms. o Added specific references to FEC mechanisms.
 End of changes. 153 change blocks. 
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