draft-ietf-rtcweb-jsep-13.txt   draft-ietf-rtcweb-jsep-14.txt 
Network Working Group J. Uberti Network Working Group J. Uberti
Internet-Draft Google Internet-Draft Google
Intended status: Standards Track C. Jennings Intended status: Standards Track C. Jennings
Expires: September 10, 2016 Cisco Expires: September 22, 2016 Cisco
E. Rescorla, Ed. E. Rescorla, Ed.
Mozilla Mozilla
March 9, 2016 March 21, 2016
Javascript Session Establishment Protocol Javascript Session Establishment Protocol
draft-ietf-rtcweb-jsep-13 draft-ietf-rtcweb-jsep-14
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
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working documents as Internet-Drafts. The list of current Internet- working documents as Internet-Drafts. The list of current Internet-
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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 September 10, 2016. This Internet-Draft will expire on September 22, 2016.
Copyright Notice Copyright Notice
Copyright (c) 2016 IETF Trust and the persons identified as the Copyright (c) 2016 IETF Trust and the persons identified as the
document authors. All rights reserved. document authors. All rights reserved.
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Table of Contents Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 3 1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 3
1.1. General Design of JSEP . . . . . . . . . . . . . . . . . 3 1.1. General Design of JSEP . . . . . . . . . . . . . . . . . 4
1.2. Other Approaches Considered . . . . . . . . . . . . . . . 5 1.2. Other Approaches Considered . . . . . . . . . . . . . . . 5
2. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 6 2. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 6
3. Semantics and Syntax . . . . . . . . . . . . . . . . . . . . 6 3. Semantics and Syntax . . . . . . . . . . . . . . . . . . . . 6
3.1. Signaling Model . . . . . . . . . . . . . . . . . . . . . 6 3.1. Signaling Model . . . . . . . . . . . . . . . . . . . . . 6
3.2. Session Descriptions and State Machine . . . . . . . . . 7 3.2. Session Descriptions and State Machine . . . . . . . . . 7
3.3. Session Description Format . . . . . . . . . . . . . . . 10 3.3. Session Description Format . . . . . . . . . . . . . . . 10
3.4. ICE . . . . . . . . . . . . . . . . . . . . . . . . . . . 10 3.4. Session Description Control . . . . . . . . . . . . . . . 10
3.4.1. ICE Gathering Overview . . . . . . . . . . . . . . . 10 3.4.1. RtpTransceivers . . . . . . . . . . . . . . . . . . . 10
3.4.2. ICE Candidate Trickling . . . . . . . . . . . . . . . 11 3.4.2. RtpSenders . . . . . . . . . . . . . . . . . . . . . 11
3.4.2.1. ICE Candidate Format . . . . . . . . . . . . . . 12 3.4.3. RtpReceivers . . . . . . . . . . . . . . . . . . . . 11
3.4.3. ICE Candidate Policy . . . . . . . . . . . . . . . . 12 3.5. ICE . . . . . . . . . . . . . . . . . . . . . . . . . . . 11
3.4.4. ICE Candidate Pool . . . . . . . . . . . . . . . . . 13 3.5.1. ICE Gathering Overview . . . . . . . . . . . . . . . 11
3.5. Video Size Negotiation . . . . . . . . . . . . . . . . . 14 3.5.2. ICE Candidate Trickling . . . . . . . . . . . . . . . 12
3.5.1. Creating an imageattr Attribute . . . . . . . . . . . 14 3.5.2.1. ICE Candidate Format . . . . . . . . . . . . . . 12
3.5.2. Interpreting an imageattr Attribute . . . . . . . . . 15 3.5.3. ICE Candidate Policy . . . . . . . . . . . . . . . . 13
3.6. Interactions With Forking . . . . . . . . . . . . . . . . 16 3.5.4. ICE Candidate Pool . . . . . . . . . . . . . . . . . 14
3.6.1. Sequential Forking . . . . . . . . . . . . . . . . . 16 3.6. Video Size Negotiation . . . . . . . . . . . . . . . . . 14
3.6.2. Parallel Forking . . . . . . . . . . . . . . . . . . 17 3.6.1. Creating an imageattr Attribute . . . . . . . . . . . 14
3.6.2. Interpreting an imageattr Attribute . . . . . . . . . 15
3.7. Interactions With Forking . . . . . . . . . . . . . . . . 17
3.7.1. Sequential Forking . . . . . . . . . . . . . . . . . 17
3.7.2. Parallel Forking . . . . . . . . . . . . . . . . . . 18
4. Interface . . . . . . . . . . . . . . . . . . . . . . . . . . 18 4. Interface . . . . . . . . . . . . . . . . . . . . . . . . . . 18
4.1. Methods . . . . . . . . . . . . . . . . . . . . . . . . . 18 4.1. Methods . . . . . . . . . . . . . . . . . . . . . . . . . 19
4.1.1. Constructor . . . . . . . . . . . . . . . . . . . . . 18 4.1.1. Constructor . . . . . . . . . . . . . . . . . . . . . 19
4.1.2. createOffer . . . . . . . . . . . . . . . . . . . . . 20 4.1.2. createOffer . . . . . . . . . . . . . . . . . . . . . 21
4.1.3. createAnswer . . . . . . . . . . . . . . . . . . . . 21 4.1.3. createAnswer . . . . . . . . . . . . . . . . . . . . 22
4.1.4. SessionDescriptionType . . . . . . . . . . . . . . . 22 4.1.4. SessionDescriptionType . . . . . . . . . . . . . . . 22
4.1.4.1. Use of Provisional Answers . . . . . . . . . . . 22 4.1.4.1. Use of Provisional Answers . . . . . . . . . . . 23
4.1.4.2. Rollback . . . . . . . . . . . . . . . . . . . . 23 4.1.4.2. Rollback . . . . . . . . . . . . . . . . . . . . 24
4.1.5. setLocalDescription . . . . . . . . . . . . . . . . . 24 4.1.5. setLocalDescription . . . . . . . . . . . . . . . . . 25
4.1.6. setRemoteDescription . . . . . . . . . . . . . . . . 24 4.1.6. setRemoteDescription . . . . . . . . . . . . . . . . 25
4.1.7. currentLocalDescription . . . . . . . . . . . . . . . 25 4.1.7. currentLocalDescription . . . . . . . . . . . . . . . 26
4.1.8. pendingLocalDescription . . . . . . . . . . . . . . . 25 4.1.8. pendingLocalDescription . . . . . . . . . . . . . . . 26
4.1.9. currentRemoteDescription . . . . . . . . . . . . . . 25 4.1.9. currentRemoteDescription . . . . . . . . . . . . . . 26
4.1.10. pendingRemoteDescription . . . . . . . . . . . . . . 25 4.1.10. pendingRemoteDescription . . . . . . . . . . . . . . 26
4.1.11. canTrickleIceCandidates . . . . . . . . . . . . . . . 26 4.1.11. canTrickleIceCandidates . . . . . . . . . . . . . . . 27
4.1.12. setConfiguration . . . . . . . . . . . . . . . . . . 26 4.1.12. setConfiguration . . . . . . . . . . . . . . . . . . 27
4.1.13. addIceCandidate . . . . . . . . . . . . . . . . . . . 27 4.1.13. addIceCandidate . . . . . . . . . . . . . . . . . . . 28
5. SDP Interaction Procedures . . . . . . . . . . . . . . . . . 27 5. SDP Interaction Procedures . . . . . . . . . . . . . . . . . 28
5.1. Requirements Overview . . . . . . . . . . . . . . . . . . 27 5.1. Requirements Overview . . . . . . . . . . . . . . . . . . 28
5.1.1. Implementation Requirements . . . . . . . . . . . . . 28 5.1.1. Implementation Requirements . . . . . . . . . . . . . 29
5.1.2. Usage Requirements . . . . . . . . . . . . . . . . . 29 5.1.2. Usage Requirements . . . . . . . . . . . . . . . . . 30
5.1.3. Profile Names and Interoperability . . . . . . . . . 29 5.1.3. Profile Names and Interoperability . . . . . . . . . 30
5.2. Constructing an Offer . . . . . . . . . . . . . . . . . . 30 5.2. Constructing an Offer . . . . . . . . . . . . . . . . . . 31
5.2.1. Initial Offers . . . . . . . . . . . . . . . . . . . 30 5.2.1. Initial Offers . . . . . . . . . . . . . . . . . . . 31
5.2.2. Subsequent Offers . . . . . . . . . . . . . . . . . . 35 5.2.2. Subsequent Offers . . . . . . . . . . . . . . . . . . 37
5.2.3. Options Handling . . . . . . . . . . . . . . . . . . 38 5.2.3. Options Handling . . . . . . . . . . . . . . . . . . 40
5.2.3.1. OfferToReceiveAudio . . . . . . . . . . . . . . . 38 5.2.3.1. IceRestart . . . . . . . . . . . . . . . . . . . 40
5.2.3.2. OfferToReceiveVideo . . . . . . . . . . . . . . . 39 5.2.3.2. VoiceActivityDetection . . . . . . . . . . . . . 40
5.2.3.3. IceRestart . . . . . . . . . . . . . . . . . . . 39 5.2.4. Direction Attribute in Offers . . . . . . . . . . . . 41
5.2.3.4. VoiceActivityDetection . . . . . . . . . . . . . 39 5.3. Generating an Answer . . . . . . . . . . . . . . . . . . 41
5.3. Generating an Answer . . . . . . . . . . . . . . . . . . 40 5.3.1. Initial Answers . . . . . . . . . . . . . . . . . . . 42
5.3.1. Initial Answers . . . . . . . . . . . . . . . . . . . 40 5.3.2. Subsequent Answers . . . . . . . . . . . . . . . . . 46
5.3.2. Subsequent Answers . . . . . . . . . . . . . . . . . 45 5.3.3. Options Handling . . . . . . . . . . . . . . . . . . 47
5.3.3. Options Handling . . . . . . . . . . . . . . . . . . 46 5.3.3.1. VoiceActivityDetection . . . . . . . . . . . . . 47
5.3.3.1. VoiceActivityDetection . . . . . . . . . . . . . 46 5.3.4. Direction Attribute in Answers . . . . . . . . . . . 47
5.4. Processing a Local Description . . . . . . . . . . . . . 46 5.4. Processing a Local Description . . . . . . . . . . . . . 48
5.5. Processing a Remote Description . . . . . . . . . . . . . 47 5.5. Processing a Remote Description . . . . . . . . . . . . . 48
5.6. Parsing a Session Description . . . . . . . . . . . . . . 47 5.6. Parsing a Session Description . . . . . . . . . . . . . . 49
5.6.1. Session-Level Parsing . . . . . . . . . . . . . . . . 48 5.6.1. Session-Level Parsing . . . . . . . . . . . . . . . . 50
5.6.2. Media Section Parsing . . . . . . . . . . . . . . . . 50 5.6.2. Media Section Parsing . . . . . . . . . . . . . . . . 52
5.6.3. Semantics Verification . . . . . . . . . . . . . . . 52 5.6.3. Semantics Verification . . . . . . . . . . . . . . . 54
5.7. Applying a Local Description . . . . . . . . . . . . . . 53 5.7. Applying a Local Description . . . . . . . . . . . . . . 55
5.8. Applying a Remote Description . . . . . . . . . . . . . . 54 5.8. Applying a Remote Description . . . . . . . . . . . . . . 57
5.9. Applying an Answer . . . . . . . . . . . . . . . . . . . 56 5.9. Applying an Answer . . . . . . . . . . . . . . . . . . . 59
6. Configurable SDP Parameters . . . . . . . . . . . . . . . . . 57 6. Configurable SDP Parameters . . . . . . . . . . . . . . . . . 60
7. Examples . . . . . . . . . . . . . . . . . . . . . . . . . . 59 7. Examples . . . . . . . . . . . . . . . . . . . . . . . . . . 62
7.1. Simple Example . . . . . . . . . . . . . . . . . . . . . 59 7.1. Simple Example . . . . . . . . . . . . . . . . . . . . . 62
7.2. Normal Examples . . . . . . . . . . . . . . . . . . . . . 63 7.2. Normal Examples . . . . . . . . . . . . . . . . . . . . . 66
8. Security Considerations . . . . . . . . . . . . . . . . . . . 72 8. Security Considerations . . . . . . . . . . . . . . . . . . . 75
9. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 72 9. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 75
10. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 72 10. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 75
11. References . . . . . . . . . . . . . . . . . . . . . . . . . 73 11. References . . . . . . . . . . . . . . . . . . . . . . . . . 76
11.1. Normative References . . . . . . . . . . . . . . . . . . 73 11.1. Normative References . . . . . . . . . . . . . . . . . . 76
11.2. Informative References . . . . . . . . . . . . . . . . . 75 11.2. Informative References . . . . . . . . . . . . . . . . . 78
Appendix A. Change log . . . . . . . . . . . . . . . . . . . . . 77 Appendix A. Change log . . . . . . . . . . . . . . . . . . . . . 80
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 81 Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 85
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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to generate and consume that JSON. to generate and consume that JSON.
Other methods may be added to SessionDescription in the future to Other methods may be added to SessionDescription in the future to
simplify handling of SessionDescriptions from Javascript. In the simplify handling of SessionDescriptions from Javascript. In the
meantime, Javascript libraries can be used to perform these meantime, Javascript libraries can be used to perform these
manipulations. manipulations.
Note that most applications should be able to treat the Note that most applications should be able to treat the
SessionDescriptions produced and consumed by these various API calls SessionDescriptions produced and consumed by these various API calls
as opaque blobs; that is, the application will not need to read or as opaque blobs; that is, the application will not need to read or
change them. The W3C WebRTC API specification will provide change them.
appropriate APIs to allow the application to control various session
parameters, which will provide the necessary information to the
browser about what sort of SessionDescription to produce.
3.4. ICE 3.4. Session Description Control
3.4.1. ICE Gathering Overview In order to give the application control over various common session
parameters, JSEP provides control surfaces which tell the browser how
to generate session descriptions. This avoids the need for
Javascript to modify session descriptions in most cases.
Changes to these objects result in changes to the session
descriptions generated by subsequent createOffer/Answer calls.
3.4.1. RtpTransceivers
RtpTransceivers allow the application to control the RTP media
associated with one m= section. Each RtpTransceiver has an RtpSender
and an RtpReceiver, which an application can use to control the
sending and receiving of RTP media. The application may also modify
the RtpTransceiver directly, for instance, by stopping it.
RtpTransceivers generally have a 1:1 mapping with m= sections,
although there may be more RtpTransceivers than m= sections when
RtpTransceivers are created but not yet associated with a m= section,
or if RtpTransceivers have been stopped and disassociated from m=
sections. 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
implicitly by calling setRemoteDescription with an offer that adds
new m= sections.
3.4.2. RtpSenders
RtpSenders allow the application to control how RTP media is sent.
In particular, the application can control whether an RtpSender is
active or not, which affects the directionality attribute of the
associated m= section.
3.4.3. RtpReceivers
RtpReceivers allows the application to control how RTP media is
received. In particular, the application can control whether an
RtpReceiver is active or not, which affects the directionality
attribute of the associated m= section.
3.5. ICE
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= line to the
local session description, or new ICE credentials in the description, local session description, or new ICE credentials in the description,
indicating an ICE restart. Use of new ICE credentials can be indicating an ICE restart. Use of new ICE credentials can be
triggered explicitly by the application, or implicitly by the browser triggered explicitly by the application, or implicitly by the browser
in response to changes in the ICE configuration. 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
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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= lines; other m= lines continue to use
their existing candidates. Also, when bundling is active, candidates their existing candidates. Also, when bundling is active, candidates
are only gathered (and exchanged) for the m= lines referenced in are only gathered (and exchanged) for the m= lines referenced in
BUNDLE-tags, as described in BUNDLE-tags, as described in
[I-D.ietf-mmusic-sdp-bundle-negotiation]. [I-D.ietf-mmusic-sdp-bundle-negotiation].
3.4.2. 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
gather all possible candidates. This results in faster media setup gather all possible candidates. This results in faster media setup
in cases where gathering is not performed prior to initiating the in cases where gathering is not performed prior to initiating the
call. call.
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trickling can send the initial offer immediately and send individual trickling can send the initial offer immediately and send individual
candidates when they get the notified of a new candidate; candidates when they get the notified of a new candidate;
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.
3.4.2.1. ICE Candidate Format 3.5.2.1. ICE Candidate Format
As with session descriptions, the syntax of the IceCandidate object As with session descriptions, the syntax of the IceCandidate object
provides some abstraction, but can be easily converted to and from provides some abstraction, but can be easily converted to and from
the SDP candidate lines. the SDP candidate lines.
The candidate lines are the only SDP information that is contained The candidate lines are the only SDP information that is contained
within IceCandidate, as they represent the only information needed within IceCandidate, as they represent the only information needed
that is not present in the initial offer (i.e., for trickle that is not present in the initial offer (i.e., for trickle
candidates). This information is carried with the same syntax as the candidates). This information is carried with the same syntax as the
"candidate-attribute" field defined for ICE. For example: "candidate-attribute" field defined for ICE. For example:
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one of two ways; either by a m= line index, or a MID. The m= line one of two ways; either by a m= line index, or a MID. The m= line
index is a zero-based index, with index N referring to the N+1th m= index is a zero-based index, with index N referring to the N+1th m=
line in the SDP sent by the entity which sent the IceCandidate. The line in the SDP sent by the entity which sent the IceCandidate. The
MID uses the "media stream identification" attribute, as defined in MID uses the "media stream identification" attribute, as defined in
[RFC5888], Section 4, to identify the m= line. JSEP implementations [RFC5888], Section 4, to identify the m= line. JSEP implementations
creating an ICE Candidate object MUST populate both of these fields. creating an ICE Candidate object MUST populate both of these fields.
Implementations receiving an ICE Candidate object MUST use the MID if Implementations receiving an ICE Candidate object MUST use the MID if
present, or the m= line index, if not (as it could have come from a present, or the m= line index, if not (as it could have come from a
non-JSEP endpoint). non-JSEP endpoint).
3.4.3. 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 suppress the use of
host candidates, to avoid exposing information about the local host candidates, to avoid exposing information about the local
network, or go as far as only using relay candidates, to leak as network, or go as far as only using relay candidates, to leak as
little location information as possible (note that these choices come little location information as possible (note that these choices come
with corresponding operational costs). To accomplish this, the with corresponding operational costs). To accomplish this, the
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To administer the ICE candidate policy, the browser will determine To administer the ICE candidate policy, the browser will determine
the current setting at the start of each gathering phase. Then, the current setting at the start of each gathering phase. Then,
during the gathering phase, the browser MUST NOT expose candidates during the gathering phase, the browser MUST NOT expose candidates
disallowed by the current policy to the application, use them as the disallowed by the current policy to the application, use them as the
source of connectivity checks, or indirectly expose them via other source of connectivity checks, or indirectly expose them via other
fields, such as the raddr/rport attributes for other ICE candidates. fields, such as the raddr/rport attributes for other ICE candidates.
Later, if a different policy is specified by the application, the Later, if a different policy is specified by the application, the
application can apply it by kicking off a new gathering phase via an application can apply it by kicking off a new gathering phase via an
ICE restart. ICE restart.
3.4.4. ICE Candidate Pool 3.5.4. ICE Candidate Pool
JSEP applications typically inform the browser to begin ICE gathering JSEP applications typically inform the browser to begin ICE gathering
via the information supplied to setLocalDescription, as this is where via the information supplied to setLocalDescription, as this is where
the app specifies the number of media streams, and thereby ICE the app specifies the number of media streams, and thereby ICE
components, for which to gather candidates. However, to accelerate components, for which to gather candidates. However, to accelerate
cases where the application knows the number of ICE components to use cases where the application knows the number of ICE components to use
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
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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.
3.5. Video Size Negotiation 3.6. Video Size Negotiation
Video size negotiation is the process through which a receiver can Video size negotiation is the process through which a receiver can
use the "a=imageattr" SDP attribute [RFC6236] to indicate what video use the "a=imageattr" SDP attribute [RFC6236] to indicate what video
frame sizes it is capable of receiving. A receiver may have hard frame sizes it is capable of receiving. A receiver may have hard
limits on what its video decoder can process, or it may 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.
3.5.1. 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
call, and will affect the output of the ensuing createAnswer call. call, and will affect the output of the ensuing createAnswer call.
Any constraints set after setLocalDescription is used to set the Any constraints set after setLocalDescription is used to set the
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As an example, consider a system with a HD-capable, multiformat video As an example, consider a system with a HD-capable, multiformat video
decoder, where the application has constrained the received track to decoder, where the application has constrained the received track to
at most 360p. In this case, the implemention would generate this at most 360p. In this case, the implemention would generate this
attribute: attribute:
a=imageattr:* recv [x=[16:640],y=[16:360],q=1.0] a=imageattr:* recv [x=[16:640],y=[16:360],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 16x16 up to 640x360 pixels.
3.5.2. 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
to see if the original resolution meets the size criteria specified to see if the original resolution meets the size criteria specified
in the attribute, and adapt the resolution accordingly by scaling (if in the attribute, and adapt the resolution accordingly by scaling (if
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upscaling. The sender SHOULD NOT upscale in other cases, even if the upscaling. The sender SHOULD NOT upscale in other cases, even if the
policy permits it. Upscaling MUST NOT change the track aspect ratio. policy permits it. Upscaling MUST NOT change the track aspect ratio.
If there is no appropriate and permitted scaling mechanism that If there is no appropriate and permitted scaling mechanism that
allows the received size limits to be satisfied, the sender MUST NOT allows the received size limits to be satisfied, 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.
3.6. Interactions With Forking 3.7. Interactions With Forking
Some call signaling systems allow various types of forking where an Some call signaling systems allow various types of forking where an
SDP Offer may be provided to more than one device. For example, SIP SDP Offer may be provided to more than one device. For example, SIP
[RFC3261] defines both a "Parallel Search" and "Sequential Search". [RFC3261] defines both a "Parallel Search" and "Sequential Search".
Although these are primarily signaling level issues that are outside Although these are primarily signaling level issues that are outside
the scope of JSEP, they do have some impact on the configuration of the scope of JSEP, they do have some impact on the configuration of
the media plane that is relevant. When forking happens at the the media plane that is relevant. When forking happens at the
signaling layer, the Javascript application responsible for the signaling layer, the Javascript application responsible for the
signaling needs to make the decisions about what media should be sent signaling needs to make the decisions about what media should be sent
or received at any point of time, as well as which remote endpoint it or received at any point of time, as well as which remote endpoint it
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can make the RTP and media perform as required by the application. can make the RTP and media perform as required by the application.
The basic operations that the applications can have the media engine The basic operations that the applications can have the media engine
do are: do are:
o Start exchanging media with a given remote peer, but keep all the o Start exchanging media with a given remote peer, but keep all the
resources reserved in the offer. resources reserved in the offer.
o Start exchanging media with a given remote peer, and free any o Start exchanging media with a given remote peer, and free any
resources in the offer that are not being used. resources in the offer that are not being used.
3.6.1. Sequential Forking 3.7.1. Sequential Forking
Sequential forking involves a call being dispatched to multiple Sequential forking involves a call being dispatched to multiple
remote callees, where each callee can accept the call, but only one remote callees, where each callee can accept the call, but only one
active session ever exists at a time; no mixing of received media is active session ever exists at a time; no mixing of received media is
performed. performed.
JSEP handles sequential forking well, allowing the application to JSEP handles sequential forking well, allowing the application to
easily control the policy for selecting the desired remote endpoint. easily control the policy for selecting the desired remote endpoint.
When an answer arrives from one of the callees, the application can When an answer arrives from one of the callees, the application can
choose to apply it either as a provisional answer, leaving open the choose to apply it either as a provisional answer, leaving open the
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In a "first-one-wins" situation, the first answer will be applied as In a "first-one-wins" situation, the first answer will be applied as
a final answer, and the application will reject any subsequent a final answer, and the application will reject any subsequent
answers. In SIP parlance, this would be ACK + BYE. answers. In SIP parlance, this would be ACK + BYE.
In a "last-one-wins" situation, all answers would be applied as In a "last-one-wins" situation, all answers would be applied as
provisional answers, and any previous call leg will be terminated. provisional answers, and any previous call leg will be terminated.
At some point, the application will end the setup process, perhaps At some point, the application will end the setup process, perhaps
with a timer; at this point, the application could reapply the with a timer; at this point, the application could reapply the
pending remote description as a final answer. pending remote description as a final answer.
3.6.2. Parallel Forking 3.7.2. Parallel Forking
Parallel forking involves a call being dispatched to multiple remote Parallel forking involves a call being dispatched to multiple remote
callees, where each callee can accept the call, and multiple callees, where each callee can accept the call, and multiple
simultaneous active signaling sessions can be established as a simultaneous active signaling sessions can be established as a
result. If multiple callees send media at the same time, the result. If multiple callees send media at the same time, the
possibilities for handling this are described in Section 3.1 of possibilities for handling this are described in Section 3.1 of
[RFC3960]. Most SIP devices today only support exchanging media with [RFC3960]. Most SIP devices today only support exchanging media with
a single device at a time, and do not try to mix multiple early media a single device at a time, and do not try to mix multiple early media
audio sources, as that could result in a confusing situation. For audio sources, as that could result in a confusing situation. For
example, consider having a European ringback tone mixed together with example, consider having a European ringback tone mixed together with
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4.1.1. Constructor 4.1.1. Constructor
The PeerConnection constructor allows the application to specify The PeerConnection constructor allows the application to specify
global parameters for the media session, such as the STUN/TURN global parameters for the media session, such as the STUN/TURN
servers and credentials to use when gathering candidates, as well as servers and credentials to use when gathering candidates, as well as
the initial ICE candidate policy and pool size, and also the bundle the initial ICE candidate policy and pool size, and also the bundle
policy to use. policy to use.
If an ICE candidate policy is specified, it functions as described in If an ICE candidate policy is specified, it functions as described in
Section 3.4.3, causing the browser to only surface the permitted Section 3.5.3, causing the browser to only surface the permitted
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
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does not provide an "a=rtcp-mux" attribute. does not provide an "a=rtcp-mux" attribute.
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. createOffer 4.1.2. createOffer
The createOffer method generates a blob of SDP that contains a The createOffer method generates a blob of SDP that contains a
[RFC3264] offer with the supported configurations for the session, [RFC3264] offer with the supported configurations for the session,
including descriptions of the local MediaStreams attached to this including descriptions of the media added to this PeerConnection, the
PeerConnection, the codec/RTP/RTCP options supported by this codec/RTP/RTCP options supported by this implementation, and any
implementation, and any candidates that have been gathered by the ICE candidates that have been gathered by the ICE Agent. An options
Agent. An options parameter may be supplied to provide additional parameter may be supplied to provide additional control over the
control over the generated offer. This options parameter should generated offer. This options parameter allows an application to
allow for the following manipulations to be performed: trigger an ICE restart, for the purpose of reestablishing
connectivity.
o To indicate support for a media type even if no MediaStreamTracks
of that type have been added to the session (e.g., an audio call
that wants to receive video.)
o To trigger an ICE restart, for the purpose of reestablishing
connectivity.
In the initial offer, the generated SDP will contain all desired In the initial offer, the generated SDP will contain all desired
functionality for the session (functionality that is supported but functionality for the session (functionality that is supported but
not desired by default may be omitted); for each SDP line, the not desired by default may be omitted); for each SDP line, the
generation of the SDP will follow the process defined for generating generation of the SDP will follow the process defined for generating
an initial offer from the document that specifies the given SDP line. an initial offer from the document that specifies the given SDP line.
The exact handling of initial offer generation is detailed in The exact handling of initial offer generation is detailed in
Section 5.2.1 below. Section 5.2.1 below.
In the event createOffer is called after the session is established, In the event createOffer is called after the session is established,
createOffer will generate an offer to modify the current session createOffer will generate an offer to modify the current session
based on any changes that have been made to the session, e.g. adding based on any changes that have been made to the session, e.g., adding
or removing MediaStreams, or requesting an ICE restart. For each or stopping RtpTransceivers, or requesting an ICE restart. For each
existing stream, the generation of each SDP line must follow the existing stream, the generation of each SDP line must follow the
process defined for generating an updated offer from the RFC that process defined for generating an updated offer from the RFC that
specifies the given SDP line. For each new stream, the generation of specifies the given SDP line. For each new stream, the generation of
the SDP must follow the process of generating an initial offer, as the SDP must follow the process of generating an initial offer, as
mentioned above. If no changes have been made, or for SDP lines that mentioned above. If no changes have been made, or for SDP lines that
are unaffected by the requested changes, the offer will only contain are unaffected by the requested changes, the offer will only contain
the parameters negotiated by the last offer-answer exchange. The the parameters negotiated by the last offer-answer exchange. The
exact handling of subsequent offer generation is detailed in exact handling of subsequent offer generation is detailed in
Section 5.2.2. below. Section 5.2.2. below.
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credentials, but does not result in candidate gathering, or cause credentials, but does not result in candidate gathering, or cause
media to start or stop flowing. media to start or stop flowing.
4.1.3. createAnswer 4.1.3. createAnswer
The createAnswer method generates a blob of SDP that contains a The createAnswer method generates a blob of SDP that contains a
[RFC3264] SDP answer with the supported configuration for the session [RFC3264] SDP answer with the supported configuration for the session
that is compatible with the parameters supplied in the 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 local MediaStreams attached to this descriptions of the media added to this PeerConnection, the
PeerConnection, the codec/RTP/RTCP options negotiated for this codec/RTP/RTCP options negotiated for this session, and any
session, and any candidates that have been gathered by the ICE Agent. candidates that have been gathered by the ICE Agent. An options
An options parameter may be supplied to provide additional control parameter may be supplied to provide additional control over the
over the generated answer. generated answer.
As an answer, the generated SDP will contain a specific configuration As an answer, the generated SDP will contain a specific configuration
that specifies how the media plane should be established; for each that specifies how the media plane should be established; for each
SDP line, the generation of the SDP must follow the process defined SDP line, the generation of the SDP must follow the process defined
for generating an answer from the document that specifies the given for generating an answer from the 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
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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 back to null. Any resources or candidates
that were allocated by the abandoned local description are discarded; that were allocated by the abandoned local description are discarded;
any media that is received will be processed according to the any media that is received will be processed according to the
previous local and remote descriptions. Rollback can only be used to previous local and remote descriptions. Rollback can only be used to
cancel proposed changes; there is no support for rolling back from a cancel proposed changes; there is no support for rolling back from a
stable state to a previous stable state. Note that this implies that stable state to a previous stable state. Note that this implies that
once the answerer has performed setLocalDescription with his answer, once the answerer has performed setLocalDescription with his answer,
this cannot be rolled back. this cannot be rolled back.
A rollback will disassociate any RtpTransceivers that were associated
with m= sections by the application of the rolled-back session
description (see Section 5.8 and Section 5.7). This means that some
RtpTransceivers that were previously associated will no longer be
associated with any m= section; in such cases, the value of the
RtpTransceiver's mid attribute MUST be set to null. RtpTransceivers
that were created by applying a remote offer that was subsequently
rolled back MUST be removed. However, a RtpTransceiver MUST NOT be
removed if the RtpTransceiver's RtpSender was activated by the
addTrack method. This is so that an application may call addTrack,
then call 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).
4.1.5. setLocalDescription 4.1.5. 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
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null: No SDP has been received from the other side, so it is not null: No SDP has been received from the other side, so it is not
known if it can handle trickle. This is the initial value before known if it can handle trickle. This is the initial value before
setRemoteDescription() is called. setRemoteDescription() is called.
true: SDP has been received from the other side indicating that it true: SDP has been received from the other side indicating that it
can support trickle. can support trickle.
false: SDP has been received from the other side indicating that it false: SDP has been received from the other side indicating that it
cannot support trickle. cannot support trickle.
As described in Section 3.4.2, JSEP implementations always provide As described in Section 3.5.2, JSEP implementations always provide
candidates to the application individually, consistent with what is candidates to the application individually, consistent with what is
needed for Trickle ICE. However, applications can use the needed for Trickle ICE. However, applications can use the
canTrickleIceCandidates property to determine whether their peer can canTrickleIceCandidates property to determine whether their peer can
actually do Trickle ICE, i.e., whether it is safe to send an initial actually do Trickle ICE, i.e., whether it is safe to send an initial
offer or answer followed later by candidates as they are gathered. offer or answer followed later by candidates as they are gathered.
As "true" is the only value that definitively indicates remote As "true" is the only value that definitively indicates remote
Trickle ICE support, an application which compares Trickle ICE support, an application 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.
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4.1.12. setConfiguration 4.1.12. 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 3.4.1 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, any existing candidates will be discarded, and new
candidates will be gathered from the new servers. 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,
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may result in a change to media state if it results in connectivity may result in a change to media state if it results in connectivity
being established. being established.
4.1.13. addIceCandidate 4.1.13. 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. Connectivity checks will be sent to the new candidate. Trickle ICE. Connectivity checks will be sent to the new candidate.
This method can also be used to provide an end-of-candidates
indication (as defined in [I-D.ietf-ice-trickle]) to the ICE Agent
for all media descriptions in the last remote description.
This call will result in a change to the state of the ICE Agent, and This call will result in a change to the state of the ICE Agent, and
may result in a change to media state if it results in connectivity may result in a change to media state if it results in connectivity
being established. being established.
5. SDP Interaction Procedures 5. SDP Interaction Procedures
This section describes the specific procedures to be followed when This section describes the specific procedures to be followed when
creating and parsing SDP objects. creating and parsing SDP objects.
5.1. Requirements Overview 5.1. Requirements Overview
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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, for each MediaStreamTrack that has been added to the Section 5.14. An m= section is generated for each RtpTransceiver
PeerConnection via the addStream method. (Note that this method that has been added to the PeerConnection via the addTrack,
takes a MediaStream, which can contain multiple MediaStreamTracks, addTransceiver, and setRemoteDescription methods. [[OPEN ISSUE: move
and therefore multiple m= sections can be generated even if addStream discussion of setRemoteDescription to the subsequent-offer section.]]
is only called once.) m=sections MUST be sorted first by the order in This is done in the order that their associated RtpTransceivers were
which the MediaStreams were added to the PeerConnection, and then by added to the PeerConnection and excludes RtpTranscievers that are
the alphabetical ordering of the media type for the MediaStreamTrack. stopped and not associated with an m= section (either due to an m=
For example, if a MediaStream containing both an audio and a video section being recycled or an RtpTransceiver having been stopped
MediaStreamTrack is added to a PeerConnection, the resultant m=audio before being associated with an m= section) .
section will precede the m=video section. If a second MediaStream
containing an audio MediaStreamTrack was added, it would follow the
m=video section.
Each m= section, provided it is not 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 the certificate for the identity For DTLS, all m= sections MUST use the certificate for the identity
that has been specified for the PeerConnection; as a result, they that has been specified for the PeerConnection; as a result, they
MUST all have the same [RFC4572] fingerprint value, or this value MUST all have the same [RFC4572] fingerprint value, or this value
MUST be a session-level attribute. MUST be a session-level attribute.
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o An "a=mid" line, as specified in [RFC5888], Section 4. When o An "a=mid" line, as specified in [RFC5888], Section 4. When
generating mid values, it is RECOMMENDED that the values be 3 generating mid values, it is RECOMMENDED that the values be 3
bytes or less, to allow them to efficiently fit into the RTP bytes or less, to allow them to efficiently fit into the RTP
header extension defined in header extension defined in
[I-D.ietf-mmusic-sdp-bundle-negotiation], Section 11. [I-D.ietf-mmusic-sdp-bundle-negotiation], Section 11.
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=msid" line, as specified in [I-D.ietf-mmusic-msid], o A direction attribute for the associated RtpTransceiver as
Section 2. described by Section 5.2.4.
o An "a=sendrecv" line, as specified in [RFC3264], Section 5.1.
o For each supported codec, "a=rtpmap" and "a=fmtp" lines, as o For each supported codec, "a=rtpmap" and "a=fmtp" lines, as
specified in [RFC4566], Section 6. The audio and video codecs specified in [RFC4566], Section 6. The audio and video codecs
that MUST be supported are specified in [I-D.ietf-rtcweb-audio] that MUST be supported are specified in [I-D.ietf-rtcweb-audio]
(see Section 3) and [I-D.ietf-rtcweb-video] (see Section 5). (see Section 3) and [I-D.ietf-rtcweb-video] (see Section 5).
o If this m= section is for media with configurable frame sizes, o If this m= section is for media with configurable frame sizes,
e.g. audio, an "a=maxptime" line, indicating the smallest of the e.g. audio, an "a=maxptime" line, indicating the smallest of the
maximum supported frame sizes out of all codecs included above, as maximum supported frame sizes out of all codecs included above, as
specified in [RFC4566], Section 6. 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 3.5. "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.
o For each supported FEC mechanism, "a=rtpmap" and "a=fmtp" lines, o For each supported FEC mechanism, "a=rtpmap" and "a=fmtp" lines,
as specified in [RFC4566], Section 6. The FEC mechanisms that as specified in [RFC4566], Section 6. The FEC mechanisms that
MUST be supported are specified in [I-D.ietf-rtcweb-fec], MUST be supported are specified in [I-D.ietf-rtcweb-fec],
skipping to change at page 34, line 34 skipping to change at page 35, line 32
with the FEC SSRC, and an "a=ssrc-group" line with semantics set with the FEC SSRC, and an "a=ssrc-group" line with semantics set
to "FEC-FR" and including the primary and FEC SSRCs, as specified to "FEC-FR" and including the primary and FEC SSRCs, as specified
in [RFC5956], section 4.3. For simplicity, if both RTX and FEC in [RFC5956], section 4.3. For simplicity, if both RTX and FEC
are supported, the FEC SSRC MUST be the same as the RTX SSRC. are supported, the FEC SSRC MUST be the same as the RTX SSRC.
o If the bundle policy for this PeerConnection is set to "max- o If the bundle policy for this PeerConnection is set to "max-
bundle", and this is not the first m= section, or the bundle bundle", and this is not the first m= section, or the bundle
policy is set to "balanced", and this is not the first m= section policy is set to "balanced", and this is not the first m= section
for this media type, an "a=bundle-only" line. for this media type, an "a=bundle-only" line.
o If the RtpSender of the RtpTransceiver associated with this
m=section is active:
* An "a=msid" line, as specified in [I-D.ietf-mmusic-msid],
Section 2.
* An "a=ssrc" line, as specified in [RFC5576], Section 4.1,
indicating the SSRC to be used for sending media, along with
the mandatory "cname" source attribute, as specified in
Section 6.1, indicating the CNAME for the source. The CNAME
MUST be generated in accordance with Section 4.9 of
[I-D.ietf-rtcweb-rtp-usage].
* If RTX is supported for this media type, another "a=ssrc" line
with the RTX SSRC, and an "a=ssrc-group" line, as specified in
[RFC5576], section 4.2, with semantics set to "FID" and
including the primary and RTX SSRCs.
* If FEC is supported for this media type, another "a=ssrc" line
with the FEC SSRC, and an "a=ssrc-group" line with semantics
set to "FEC-FR" and including the primary and FEC SSRCs, as
specified in [RFC5956], section 4.3. For simplicity, if both
RTX and FEC are supported, the FEC SSRC MUST be the same as the
RTX SSRC.
o If the RtpTransceiver's RtpSender is active, and the application
has specified RID values or has specified more than one encoding
in the RtpSenders's parameters, an "a=rid" line for each encoding
specified. The "a=rid" line is specified in
[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-
identifier; otherwise a RID value MUST be generated by the
implementation. When generating RID values, it is RECOMMENDED
that the values be 3 bytes or less, to allow them to efficiently
fit into the RTP header extension defined in
[I-D.ietf-avtext-rid], Section 11. If no encodings have been
specified, or only one encoding is specified but without a RID
value, then no "a=rid" lines are generated.
o If the RtpTransceiver's RtpSender is active and more than one
"a=rid" line has been generated, an "a=simulcast" line, with
direction "send", as defined in [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 for this m= section.
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" if the default
candidate uses UDP transport, or "TCP/DTLS/SCTP" if the default candidate uses UDP transport, or "TCP/DTLS/SCTP" if the default
candidate uses TCP transport [I-D.ietf-mmusic-sctp-sdp]. The "fmt" candidate uses TCP transport [I-D.ietf-mmusic-sctp-sdp]. The "fmt"
value MUST be set to "webrtc-datachannel" as specified in value MUST be set to "webrtc-datachannel" as specified in
[I-D.ietf-mmusic-sctp-sdp], Section 4.1. [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", and "a=setup" lines MUST be included as mentioned "a=fingerprint", and "a=setup" lines MUST be included as mentioned
skipping to change at page 35, line 8 skipping to change at page 36, line 50
[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
each m= section. The effect of this is that the browser offers all each m= section. The effect of this is that the browser offers all
m= sections as one bundle group. However, whether the m= sections m= sections as one bundle group. However, whether the m= sections
are bundle-only or not depends on the bundle policy. are bundle-only or not depends on the bundle policy.
The next step is to generate session-level lip sync groups as defined The next step is to generate session-level lip sync groups as defined
in [RFC5888], Section 7. For each MediaStream with more than one in [RFC5888], Section 7. For each MediaStream referenced by more
MediaStreamTrack, a group of type "LS" MUST be added that contains than one RtpTransceiver (by passing those MediaStreams as arguments
the mid values for each MediaStreamTrack in that MediaStream. to the addTrack and addTransceiver methods), a group of type "LS"
MUST be added that contains the mid values for each RtpTransceiver.
Attributes which SDP permits to either be at the session level or the Attributes which SDP permits to either be at the session level or the
media level SHOULD generally be at the media level even if they are media level SHOULD generally be at the media level even if they are
identical. This promotes readability, especially if one of a set of identical. This promotes readability, especially if one of a set of
initially identical attributes is subsequently changed. initially identical attributes is subsequently changed.
Attributes other than the ones specified above MAY be included, Attributes other than the ones specified above MAY be included,
except for the following attributes which are specifically except for the following attributes which are specifically
incompatible with the requirements of [I-D.ietf-rtcweb-rtp-usage], incompatible with the requirements of [I-D.ietf-rtcweb-rtp-usage],
and MUST NOT be included: and MUST NOT be included:
skipping to change at page 36, line 13 skipping to change at page 38, line 9
candidates. candidates.
If the initial offer was applied using setLocalDescription, but an If the initial offer was applied using setLocalDescription, but an
answer from the remote side has not yet been applied, meaning the answer from the remote side has not yet been applied, meaning the
PeerConnection is still in the "local-offer" state, an offer is PeerConnection is still in the "local-offer" state, an offer is
generated by following the steps in the "stable" state above, along generated by following the steps in the "stable" state above, along
with these exceptions: with these exceptions:
o The "s=" and "t=" lines MUST stay the same. o The "s=" and "t=" lines MUST stay the same.
o If any RtpTransceiver has been added, and there exists an m=
section with a zero port in the current local description or the
current remote description, that m= section MUST be recycled by
generating an m= section for the added RtpTransceiver as if the m=
section were being added to the session description, placed at the
same index as the m= section with a zero port.
o If an RtpTransceiver is stopped and is not associated with an m=
section, an m= section MUST NOT be generated for it. This
prevents adding back RtpTransceivers whose m= sections were
recycled and used for a new RtpTransceiver in a previous offer/
answer exchange, as described above.
o If an RtpTransceiver has been stopped and is associated with an m=
section, and the m= section is not being recycled as described
above, an m= section MUST be generated for it with the port set to
zero and the "a=msid", "a=ssrc", and "a=ssrc-group" lines removed.
o For RtpTransceivers that are not stopped, the "a=msid", "a=ssrc",
and "a=ssrc-group" lines MUST stay the same if they are present in
the current description.
o Each "m=" and c=" line MUST be filled in with the port, protocol, o Each "m=" and c=" line MUST be filled in with the port, protocol,
and address of the default candidate for the m= section, as and address of the default candidate for the m= section, as
described in [RFC5245], Section 4.3. If ICE checking has already described in [RFC5245], Section 4.3. If ICE checking has already
completed for one or more candidate pairs and a candidate pair is completed for one or more candidate pairs and a candidate pair is
in active use, then that pair MUST be used, even if ICE has not in active use, then that pair MUST be used, even if ICE has not
yet completed. Note that this differs from the guidance in yet completed. Note that this differs from the guidance in
[RFC5245], Section 9.1.2.2, which only refers to offers created [RFC5245], Section 9.1.2.2, which only refers to offers created
when ICE has completed. Each "a=rtcp" attribute line MUST also be when ICE has completed. Each "a=rtcp" attribute line MUST also be
filled in with the port and address of the appropriate default filled in with the port and address of the appropriate default
candidate, either the default RTP or RTCP candidate, depending on candidate, either the default RTP or RTCP candidate, depending on
skipping to change at page 36, line 35 skipping to change at page 39, line 5
default RTCP candidate MUST be used, as indicated in [RFC5761], default RTCP candidate MUST be used, as indicated in [RFC5761],
section 5.1.3. In each case, if no candidates of the desired type section 5.1.3. In each case, if no candidates of the desired type
have yet been gathered, dummy values MUST be used, as described have yet been gathered, dummy values MUST be used, as described
above. above.
o Each "a=mid" line MUST stay the same. o Each "a=mid" line MUST stay the same.
o Each "a=ice-ufrag" and "a=ice-pwd" line MUST stay the same, unless o Each "a=ice-ufrag" and "a=ice-pwd" line MUST stay the same, 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 5.2.3.3 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, 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 3.4.1), 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.
o For MediaStreamTracks that are still present, the "a=msid", o For RtpTransceivers that are still present, the "a=msid",
"a=ssrc", and "a=ssrc-group" lines MUST stay the same. "a=ssrc", and "a=ssrc-group" lines MUST stay the same.
o If any MediaStreamTracks have been removed, either through the o For RtpTransceivers that are still present, the "a=rid" lines MUST
removeStream method or by removing them from an added MediaStream, stay the same.
their m= sections MUST be marked as recvonly by changing the value
of the [RFC3264] directional attribute to "a=recvonly". The
"a=msid", "a=ssrc", and "a=ssrc-group" lines MUST be removed from
the associated m= sections.
o If any MediaStreamTracks have been added, and there exist m= o For RtpTransceivers that are still present, any "a=simulcast" line
sections of the appropriate media type with no associated MUST stay the same.
MediaStreamTracks (i.e. as described in the preceding paragraph),
those m= sections MUST be recycled by adding the new o If any RtpTransceiver has been stopped, the port MUST be set to
MediaStreamTrack to the m= section. This is done by adding the zero and the "a=msid", "a=ssrc", and "a=ssrc-group" lines MUST be
necessary "a=msid", "a=ssrc", and "a=ssrc-group" lines to the removed.
recycled m= section, and removing the "a=recvonly" attribute.
o If any RtpTransceiver has been added, and there exists a m=
section with a zero port in the current local description or the
current remote description, that m= section MUST be recycled by
generating a m= section for the added RtpTransceiver as if the m=
section were being added to session description, except that
instead of adding it, the generated m= section replaces the m=
section with a zero port.
If the initial offer was applied using setLocalDescription, and an If the initial offer was applied using setLocalDescription, and an
answer from the remote side has been applied using answer from the remote side has been applied using
setRemoteDescription, meaning the PeerConnection is in the "remote- setRemoteDescription, meaning the PeerConnection is in the "remote-
pranswer" or "stable" states, an offer is generated based on the pranswer" or "stable" states, an offer is generated based on the
negotiated session descriptions by following the steps mentioned for negotiated session descriptions by following the steps mentioned for
the "local-offer" state above, along with these exceptions: the "local-offer" state above.
o If a m= section exists in the current local description, but does
not have an associated local MediaStreamTrack (possibly because
said MediaStreamTrack was removed since the last exchange), a m=
section MUST still be generated in the new offer, as indicated in
[RFC3264], Section 8. The disposition of this section will depend
on the state of the remote MediaStreamTrack associated with this
m= section. If one exists, and it is still in the "live" state,
the new m= section MUST be marked as "a=recvonly", with no
"a=msid" or related attributes present. If no remote
MediaStreamTrack exists, or it is in the "ended" state, the m=
section MUST be marked as rejected, by setting the port to zero,
as indicated in [RFC3264], Section 8.2.
o If any MediaStreamTracks have been added, and there exist recvonly
m= sections of the appropriate media type with no associated
MediaStreamTracks, or rejected m= sections of any media type,
those m= sections MUST be recycled, and a local MediaStreamTrack
associated with these recycled m= sections until all such existing
m= sections have been used. This includes any recvonly or
rejected m= sections created by the preceding paragraph.
In addition, for each non-recycled, non-rejected m= section in the In addition, for each non-recycled, non-rejected m= section in the
new offer, the following adjustments are made based on the contents new offer, the following adjustments are made based on the contents
of the corresponding m= section in the current remote description: of the corresponding m= section in the current remote description:
o The m= line and corresponding "a=rtpmap" and "a=fmtp" lines MUST o The m= line and corresponding "a=rtpmap" and "a=fmtp" lines MUST
only include codecs present in the remote description. only include codecs present in the remote description.
o The RTP header extensions MUST only include those that are present o The RTP header extensions MUST only include those that are present
in the remote description. in the remote description.
skipping to change at page 38, line 35 skipping to change at page 40, line 35
are still alive, as well as any new m= sections. are still alive, as well as any new m= sections.
The "LS" groups are generated in the same way as with initial offers. The "LS" groups are generated in the same way as with initial offers.
5.2.3. Options Handling 5.2.3. Options Handling
The createOffer method takes as a parameter an RTCOfferOptions The createOffer method takes as a parameter an RTCOfferOptions
object. Special processing is performed when generating a SDP object. Special processing is performed when generating a SDP
description if the following options are present. description if the following options are present.
5.2.3.1. OfferToReceiveAudio 5.2.3.1. IceRestart
If the "OfferToReceiveAudio" option is specified, with an integer
value of N, and M audio MediaStreamTracks have been added to the
PeerConnection, the offer MUST include N non-rejected m= sections
with media type "audio", even if N is greater than M. This allows
the offerer to receive audio, including multiple independent streams,
even when not sending it; accordingly, the directional attribute on
the N-M audio m= sections without associated MediaStreamTracks MUST
be set to recvonly.
If N is set to a value less than M, the offer MUST mark the m=
sections associated with the M-N most recently added (since the last
setLocalDescription) MediaStreamTracks as sendonly. This allows the
offerer to indicate that it does not want to receive audio on some or
all of its newly created streams. For m= sections that have
previously been negotiated, this setting has no effect. [TODO: refer
to RTCRtpSender in the future]
For backwards compatibility with pre-standard versions of this
specification, a value of "true" is interpreted as equivalent to N=1,
and "false" as N=0.
5.2.3.2. OfferToReceiveVideo
If the "OfferToReceiveVideo" option is specified, with an integer
value of N, and M video MediaStreamTracks have been added to the
PeerConnection, the offer MUST include N non-rejected m= sections
with media type "video", even if N is greater than M. This allows
the offerer to receive video, including multiple independent streams,
even when not sending it; accordingly, the directional attribute on
the N-M video m= sections without associated MediaStreamTracks MUST
be set to recvonly.
If N is set to a value less than M, the offer MUST mark the m=
sections associated with the M-N most recently added (since the last
setLocalDescription) MediaStreamTracks as sendonly. This allows the
offerer to indicate that it does not want to receive video on some or
all of its newly created streams. For m= sections that have
previously been negotiated, this setting has no effect. [TODO: refer
to RTCRtpSender in the future]
For backwards compatibility with pre-standard versions of this
specification, a value of "true" is interpreted as equivalent to N=1,
and "false" as N=0.
5.2.3.3. IceRestart
If the "IceRestart" option is specified, with a value of "true", the If the "IceRestart" option is specified, with a value of "true", the
offer MUST indicate an ICE restart by generating new ICE ufrag and offer MUST indicate an ICE restart by generating new ICE ufrag and
pwd attributes, as specified in [RFC5245], Section 9.1.1.1. If this pwd attributes, as specified in [RFC5245], Section 9.1.1.1. If this
option is specified on an initial offer, it has no effect (since a option is specified on an initial offer, it has no effect (since a
new ICE ufrag and pwd are already generated). Similarly, if the ICE new ICE ufrag and pwd are already generated). Similarly, if the ICE
configuration has changed, this option has no effect, since new ufrag configuration has changed, this option has no effect, since new ufrag
and pwd attributes will be generated automatically. This option is and pwd attributes will be generated automatically. This option is
primarily useful for reestablishing connectivity in cases where primarily useful for reestablishing connectivity in cases where
failures are detected by the application. failures are detected by the application.
5.2.3.4. VoiceActivityDetection 5.2.3.2. VoiceActivityDetection
If the "VoiceActivityDetection" option is specified, with a value of If the "VoiceActivityDetection" option is specified, with a value of
"true", the offer MUST indicate support for silence suppression in "true", the offer MUST indicate support for silence suppression in
the audio it receives by including comfort noise ("CN") codecs for the audio it receives by including comfort noise ("CN") codecs for
each offered audio codec, as specified in [RFC3389], Section 5.1, each offered audio codec, as specified in [RFC3389], Section 5.1,
except for codecs that have their own internal silence suppression except for codecs that have their own internal silence suppression
support. For codecs that have their own internal silence suppression support. For codecs that have their own internal silence suppression
support, the appropriate fmtp parameters for that codec MUST be support, the appropriate fmtp parameters for that codec MUST be
specified to indicate that silence suppression for received audio is specified to indicate that silence suppression for received audio is
desired. For example, when using the Opus codec, the "usedtx=1" desired. For example, when using the Opus codec, the "usedtx=1"
skipping to change at page 40, line 28 skipping to change at page 41, line 31
Note that setting the "VoiceActivityDetection" parameter when Note that setting the "VoiceActivityDetection" parameter when
generating an offer is a request to receive audio with silence generating an offer is a request to receive audio with silence
suppression. It has no impact on whether the local endpoint does suppression. It has no impact on whether the local endpoint does
silence suppression for the audio it sends. silence suppression for the audio it sends.
The "VoiceActivityDetection" option does not have any impact on the The "VoiceActivityDetection" option does not have any impact on the
setting of the "vad" value in the signaling of the client to mixer setting of the "vad" value in the signaling of the client to mixer
audio level header extension described in [RFC6464], Section 4. audio level header extension described in [RFC6464], Section 4.
5.2.4. Direction Attribute in Offers
[RFC3264] direction attributes (defined in Section 6.1) in offers are
chosen according to the states of the RtpSender and RtpReceiver of a
given RtpTransceiver, as follows:
+-----------+-------------+-----------------+
| RtpSender | RtpReceiver | offer direction |
+-----------+-------------+-----------------+
| active | active | sendrecv |
| active | inactive | sendonly |
| inactive | active | recvonly |
| inactive | inactive | inactive |
+-----------+-------------+-----------------+
5.3. Generating an Answer 5.3. Generating an Answer
When createAnswer is called, a new SDP description must be created When createAnswer is called, a new SDP description must be created
that is compatible with the supplied remote description as well as that is compatible with the supplied remote description as well as
the requirements specified in [I-D.ietf-rtcweb-rtp-usage]. The exact the requirements specified in [I-D.ietf-rtcweb-rtp-usage]. The exact
details of this process are explained below. details of this process are explained below.
5.3.1. Initial Answers 5.3.1. Initial Answers
When createAnswer is called for the first time after a remote When createAnswer is called for the first time after a remote
skipping to change at page 41, line 10 skipping to change at page 42, line 28
rejected. rejected.
The first step in generating an initial answer is to generate The first step in generating an initial answer is to generate
session-level attributes. The process here is identical to that session-level attributes. The process here is identical to that
indicated in the Initial Offers section above, except that the indicated in the Initial Offers section above, except that the
"a=ice-options" line, with the "trickle" option as specified in "a=ice-options" line, with the "trickle" option as specified in
[I-D.ietf-ice-trickle], Section 4, is only included if such an option [I-D.ietf-ice-trickle], Section 4, is only included if such an option
was present in the offer. was present in the offer.
The next step is to generate lip sync groups as defined in [RFC5888], The next step is to generate lip sync groups as defined in [RFC5888],
Section 7. For each MediaStream with more than one MediaStreamTrack, Section 7. For each MediaStream with more than one referenced
a group of type "LS" MUST be added that contains the mid values for RtpTransceiver, a group of type "LS" MUST be added that contains the
each MediaStreamTrack in that MediaStream. In some cases this may mid values for each RtpTransceiver added with that MediaStream. In
result in adding a mid to a given LS group that was not in that LS some cases this may result in adding a mid to a given LS group that
group in the associated offer. Although this is not allowed by was not in that LS group in the associated offer. Although this is
[RFC5888], it is allowed when implementing this specification. not allowed by [RFC5888], it is allowed when implementing this
[[OPEN ISSUE: This is still under discussion. See: specification. [[OPEN ISSUE: This is still under discussion. See:
https://github.com/rtcweb-wg/jsep/issues/162.]] https://github.com/rtcweb-wg/jsep/issues/162.]]
The next step is to generate m= sections for each m= section that is The next step is to generate m= sections for each m= section that is
present in the remote offer, as specified in [RFC3264], Section 6. present in the remote offer, as specified in [RFC3264], Section 6.
For the purposes of this discussion, any session-level attributes in For the purposes of this discussion, any session-level attributes in
the offer that are also valid as media-level attributes SHALL be the offer that are also valid as media-level attributes SHALL be
considered to be present in each m= section. considered to be present in each m= section.
The next step is to go through each offered m= section. If there is The next step is to go through each offered m= section. Each offered
a local MediaStreamTrack of the same type which has been added to the m= section will have an associated RtpTransceiver, as described in
PeerConnection via addStream and not yet associated with a m= Section 5.8. If there are more RtpTransceivers than there are m=
section, and the specific m= section is either sendrecv or recvonly, sections, the unmatched RtpTransceivers will need to be associated in
the MediaStreamTrack will be associated with the m= section at this a subsequent offer.
time. MediaStreamTracks are assigned to m= sections using the
canonical order described in Section 5.2.1. If there are more m=
sections of a certain type than MediaStreamTracks, some m= sections
will not have an associated MediaStreamTrack. If there are more
MediaStreamTracks of a certain type than compatible m= sections, only
the first N MediaStreamTracks will be able to be associated in the
constructed answer. The remainder will need to be associated in a
subsequent offer.
For each offered m= section, if the associated remote For each offered m= section, if the associated RtpTransceiver has
MediaStreamTrack has been stopped, and is therefore in state "ended", been stopped, the corresponding m= section in the answer MUST be
and no local MediaStreamTrack has been associated, the corresponding marked as rejected by setting the port in the m= line to zero, as
m= section in the answer MUST be marked as rejected by setting the indicated in [RFC3264], Section 6., and further processing for this
port in the m= line to zero, as indicated in [RFC3264], Section 6., m= section can be skipped.
and further processing for this m= section can be skipped.
Provided that is not the case, each m= section in the answer should Provided that is not the case, each m= section in the answer should
then be generated as specified in [RFC3264], Section 6.1. For the m= then be generated as specified in [RFC3264], Section 6.1. For the m=
line itself, the following rules must be followed: line itself, the 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 have yet been gathered, the "dummy" port value of 9 (Discard) MUST
be used, as indicated in [I-D.ietf-ice-trickle], Section 5.1. be used, as indicated in [I-D.ietf-ice-trickle], Section 5.1.
skipping to change at page 42, line 28 skipping to change at page 43, line 37
section MUST include the following: section MUST include the following:
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.
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 If a local MediaStreamTrack has been associated, an "a=msid" line, o A direction attribute for the associated RtpTransceiver described
as specified in [I-D.ietf-mmusic-msid], Section 2. by Section 5.3.4.
o Depending on the directionality of the offer, the disposition of
any associated remote MediaStreamTrack, and the presence of an
associated local MediaStreamTrack, the appropriate directionality
attribute, as specified in [RFC3264], Section 6.1. If the offer
was sendrecv, and the remote MediaStreamTrack is still "live", and
there is a local MediaStreamTrack that has been associated, the
directionality MUST be set as sendrecv. If the offer was
sendonly, and the remote MediaStreamTrack is still "live", the
directionality MUST be set as recvonly. If the offer was
recvonly, and a local MediaStreamTrack has been associated, the
directionality MUST be set as sendonly. If the offer was
inactive, the directionality MUST be set as inactive.
o For each supported codec that is present in the offer, "a=rtpmap" o For each supported codec that is present in the offer, "a=rtpmap"
and "a=fmtp" lines, as specified in [RFC4566], Section 6, and and "a=fmtp" lines, as specified in [RFC4566], Section 6, and
[RFC3264], Section 6.1. The audio and video codecs that MUST be [RFC3264], Section 6.1. The audio and video codecs that MUST be
supported are specified in [I-D.ietf-rtcweb-audio] (see Section 3) supported are specified in [I-D.ietf-rtcweb-audio] (see Section 3)
and [I-D.ietf-rtcweb-video] (see Section 5). and [I-D.ietf-rtcweb-video] (see Section 5).
o If this m= section is for media with configurable frame sizes, o If this m= section is for media with configurable frame sizes,
e.g. audio, an "a=maxptime" line, indicating the smallest of the e.g. audio, an "a=maxptime" line, indicating the smallest of the
maximum supported frame sizes out of all codecs included above, as maximum supported frame sizes out of all codecs included above, as
specified in [RFC4566], Section 6. 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 3.5. "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.
o For each supported FEC mechanism, "a=rtpmap" and "a=fmtp" lines, o For each supported FEC mechanism, "a=rtpmap" and "a=fmtp" lines,
as specified in [RFC4566], Section 6. The FEC mechanisms that as specified in [RFC4566], Section 6. The FEC mechanisms that
MUST be supported are specified in [I-D.ietf-rtcweb-fec], MUST be supported are specified in [I-D.ietf-rtcweb-fec],
skipping to change at page 44, line 11 skipping to change at page 45, line 7
specified in [I-D.ietf-rtcweb-rtp-usage], Section 5.2. Any header specified in [I-D.ietf-rtcweb-rtp-usage], Section 5.2. Any header
extensions that require encryption MUST be specified as indicated extensions that require encryption MUST be specified as indicated
in [RFC6904], Section 4. in [RFC6904], Section 4.
o For each supported RTCP feedback mechanism that is present in the o For each supported RTCP feedback mechanism that is present in the
offer, an "a=rtcp-fb" mechanism, as specified in [RFC4585], offer, an "a=rtcp-fb" mechanism, as specified in [RFC4585],
Section 4.2. The list of RTCP feedback mechanisms that SHOULD/ Section 4.2. The list of RTCP feedback mechanisms that SHOULD/
MUST be supported is specified in [I-D.ietf-rtcweb-rtp-usage], MUST be supported is specified in [I-D.ietf-rtcweb-rtp-usage],
Section 5.1. Section 5.1.
o If a local MediaStreamTrack has been associated, an "a=ssrc" line, o If the RtpSender of the RtpTransceiver associated with this
as specified in [RFC5576], Section 4.1, indicating the SSRC to be m=section is active:
used for sending media, along with the mandatory "cname" source
attribute, as specified in Section 6.1, indicating the CNAME for
the source. The CNAME MUST be generated in accordance with
Section 4.9 of [I-D.ietf-rtcweb-rtp-usage].
o If a local MediaStreamTrack has been associated, and RTX has been * An "a=msid" line, as specified in [I-D.ietf-mmusic-msid],
negotiated for this m= section, another "a=ssrc" line with the RTX Section 2.
SSRC, and an "a=ssrc-group" line, as specified in [RFC5576],
section 4.2, with semantics set to "FID" and including the primary
and RTX SSRCs.
o If a local MediaStreamTrack has been associated, and FEC has been * An "a=ssrc" line, as specified in [RFC5576], Section 4.1,
negotiated for this m= section, another "a=ssrc" line with the FEC indicating the SSRC to be used for sending media, along with
SSRC, and an "a=ssrc-group" line with semantics set to "FEC-FR" the mandatory "cname" source attribute, as specified in
and including the primary and FEC SSRCs, as specified in Section 6.1, indicating the CNAME for the source. The CNAME
[RFC5956], section 4.3. For simplicity, if both RTX and FEC are MUST be generated in accordance with Section 4.9 of
supported, the FEC SSRC MUST be the same as the RTX SSRC. [I-D.ietf-rtcweb-rtp-usage].
* If RTX has been negotiated for this m= section, another
"a=ssrc" line with the RTX SSRC, and an "a=ssrc-group" line, as
specified in [RFC5576], section 4.2, with semantics set to
"FID" and including the primary and RTX SSRCs.
* If FEC has been negotiated for this m= section, another
"a=ssrc" line with the FEC SSRC, and an "a=ssrc-group" line
with semantics set to "FEC-FR" and including the primary and
FEC SSRCs, as specified in [RFC5956], section 4.3. For
simplicity, if both RTX and FEC are supported, the FEC SSRC
MUST be the same as the RTX SSRC.
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.
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=candidate", "a=fingerprint", and "a=setup" lines MUST be included "a=candidate", "a=fingerprint", and "a=setup" lines MUST be included
as mentioned above, along with an "a=fmtp:webrtc-datachannel" line as mentioned above, along with an "a=fmtp:webrtc-datachannel" line
and an "a=sctp-port" line referencing the SCTP port number as defined and an "a=sctp-port" line referencing the SCTP port number as defined
skipping to change at page 45, line 51 skipping to change at page 47, line 7
above. above.
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 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 3.4.1), 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.
o For MediaStreamTracks that are still present, the "a=msid", o For RtpTransceivers that are not stopped, the "a=msid", "a=ssrc",
"a=ssrc", and "a=ssrc-group" lines MUST stay the same. and "a=ssrc-group" lines MUST 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 OfferToReceiveAudio, those supported in RTCOfferOptions; the IceRestart option is
OfferToReceiveVideo, and IceRestart options mentioned in unnecessary, as ICE credentials will automatically be changed for all
Section 5.2.3 are meaningless in the context of generating an answer, m= lines where the offerer chose to perform ICE restart.
as there is no need to generate extra m= lines in an answer, and ICE
credentials will automatically be changed for all m= lines 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 5.2.3.4, 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").
5.3.4. Direction Attribute in Answers
[RFC3264] direction attributes (defined in Section 6.1) in answers
are chosen according to the direction attribute in the remote offer
and the states of the RtpSender and RtpReceiver of the corresponding
RtpTransceiver, as follows:
+-----------------+-----------+-------------+------------------+
| offer direction | RtpSender | RtpReceiver | answer direction |
+-----------------+-----------+-------------+------------------+
| sendrecv | active | active | sendrecv |
| sendrecv | active | inactive | sendonly |
| sendrecv | inactive | active | recvonly |
| sendrecv | inactive | inactive | inactive |
| sendonly | * | active | recvonly |
| sendonly | * | inactive | inactive |
| recvonly | active | * | sendonly |
| recvonly | inactive | * | inactive |
| inactive | * | * | inactive |
+-----------------+-----------+-------------+------------------+
5.4. Processing a Local Description 5.4. Processing a Local Description
When a SessionDescription is supplied to setLocalDescription, the When a SessionDescription is supplied to setLocalDescription, the
following steps MUST be performed: following steps MUST be performed:
o First, the type of the SessionDescription is checked against the o First, the type of the SessionDescription is checked against the
current state of the PeerConnection: current state of the PeerConnection:
* If the type is "offer", the PeerConnection state MUST be either * If the type is "offer", the PeerConnection state MUST be either
"stable" or "have-local-offer". "stable" or "have-local-offer".
skipping to change at page 52, line 25 skipping to change at page 54, line 18
o Any "a=remote-candidates" attributes MUST be parsed as specified o Any "a=remote-candidates" attributes MUST be parsed as specified
in [RFC5245], Section 4.3, but their values are ignored. in [RFC5245], Section 4.3, but their values are ignored.
o If present, a single "a=end-of-candidates" attribute MUST be o If present, a single "a=end-of-candidates" attribute MUST be
parsed as specified in [I-D.ietf-ice-trickle], Section 8.2, and parsed as specified in [I-D.ietf-ice-trickle], Section 8.2, and
its presence or absence flagged and stored. its presence or absence flagged and stored.
o Any "a=imageattr" attributes MUST be parsed as specified in o Any "a=imageattr" attributes MUST be parsed as specified in
[RFC6236], Section 3, and their values stored. [RFC6236], Section 3, and their values stored.
o Any "a=rid" lines MUST be parsed as specified in
[I-D.ietf-mmusic-rid], Section 10, and their values stored.
o If present, a single "a=simulcast" line MUST be parsed as
specified in [I-D.ietf-mmusic-sdp-simulcast], and its values
stored.
Otherwise, if the "m=" proto value indicates use of SCTP, the Otherwise, if the "m=" proto value indicates use of SCTP, the
following attribute lines MUST be processed: following attribute lines MUST be processed:
o The "m=" fmt value MUST be parsed as specified in o The "m=" fmt value MUST be parsed as specified in
[I-D.ietf-mmusic-sctp-sdp], Section 4.3, and the application [I-D.ietf-mmusic-sctp-sdp], Section 4.3, and the application
protocol value stored. protocol value stored.
o An "a=sctp-port" attribute MUST be present, and it MUST be parsed o An "a=sctp-port" attribute MUST be present, and it MUST be parsed
as specified in [I-D.ietf-mmusic-sctp-sdp], Section 5.2, and the as specified in [I-D.ietf-mmusic-sctp-sdp], Section 5.2, and the
value stored. value stored.
skipping to change at page 53, line 16 skipping to change at page 55, line 16
limits specified in [RFC5245], Section 15.4. limits specified in [RFC5245], Section 15.4.
* DTLS setup value, which MUST be set according to the rules * DTLS setup value, which MUST be set according to the rules
specified in [RFC5763], Section 5, and MUST be consistent with specified in [RFC5763], Section 5, and MUST be consistent with
the selected role of the current DTLS connection, if one the selected role of the current DTLS connection, if one
exists.[TODO: may need revision, i.e., use of actpass exists.[TODO: may need revision, i.e., use of actpass
* 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
"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.
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
skipping to change at page 54, line 9 skipping to change at page 56, line 12
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, trigger the o Or, if the ICE ufrag and password values have changed, trigger the
ICE Agent to start an ICE restart and begin gathering new ICE Agent to start an ICE restart and begin gathering new
candidates for the media section, as defined in [RFC5245], candidates for the media section, as defined in [RFC5245],
Section 9.1.1.1, unless it has been marked as bundle-only. Section 9.1.1.1, unless it has been marked as bundle-only.
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
(which should only happen when applying an offer), find one and
associate it with this m= section according to the following
steps:
+ Find the RtpTransceiver that corresponds to the m= section
with the same MID in the created offer.
+ Set the value of the RtpTransceiver's mid attribute to the
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 5.1.1. If RTCP mux is not indicated, but was indicated
in a previous description, this MUST result in an error. in a previous description, 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 unknown, [RFC5285]. If any indicated RTP header extension is unknown,
this MUST result in an error. this MUST result in an error.
skipping to change at page 55, line 35 skipping to change at page 57, line 48
restart is needed. restart is needed.
o Configure the ICE components associated with this media section to o Configure the ICE components associated with this media section to
use the supplied ICE remote ufrag and password for their use the supplied ICE remote ufrag and password for their
connectivity checks. connectivity checks.
o Pair any supplied ICE candidates with any gathered local o Pair any supplied ICE candidates with any gathered local
candidates, as described in Section 5.7 of [RFC5245] and start candidates, as described in Section 5.7 of [RFC5245] and start
connectivity checks with the appropriate credentials. connectivity checks with the appropriate credentials.
o If an "a=end-of-candidates" attribute is present, process the end-
of-candidates indication as described in [I-D.ietf-ice-trickle]
Section 11.
o If the media section proto value indicates use of RTP: o If the media section proto value indicates use of RTP:
* [TODO: header extensions] * [TODO: header extensions]
* If the m= section is being recycled (see Section 5.2.2),
dissociate the currently associated RtpTransceiver by setting
its mid attribute to null.
* If the m= section is not associated with any RtpTransceiver
(possibly because it was dissociated in the previous step),
either find an RtpTransceiver or create one according to the
following steps:
+ If the m= section is sendrecv or recvonly, and there are
RtpTransceivers of the same type that were added to the
PeerConnection by addTrack and are not associated with any
m= section and are not stopped, find the first (according to
the canonical order described in Section 5.2.1) such
RtpTransceiver.
+ If no RtpTransceiver was found in the previous step, create
one with an inactive RtpSender and active RtpReceiver.
+ Associate the found or created RtpTransceiver with the m=
section by setting the value of the RtpTransceiver's mid
attribute to the MID of the m= section.
* For each specified payload type that is also supported by the * For each specified payload type that is also supported by the
local implementation, establish a mapping between the payload local implementation, establish a mapping between the payload
type ID and the actual media format. [TODO - Justin to add type ID and the actual media format. [TODO - Justin to add
more to explain mapping.] If any indicated payload type is more to explain mapping.] If any indicated payload type is
unknown, it MUST be ignored. [TODO: should fail on answers] unknown, it MUST be ignored. [TODO: should fail on answers]
* 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]. If any referenced primary type, as described in [RFC4588]. If any referenced primary
payload types are not present, this MUST result in an error. payload types are not present, this MUST result in an error.
skipping to change at page 57, line 29 skipping to change at page 60, line 22
MUST result in an error. 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", prepare to start transmitting media "sendrecv" or "sendonly", prepare to start transmitting media
using the specified primary SSRC and one of the selected using the specified primary SSRC and one of the selected
payload types, once the underlying transport layers have been payload types, once the underlying transport layers have been
established. Otherwise, stop transmitting RTP media, although established. If RID values are specified, include the RID
RTCP should still be sent, as described in [RFC3264], header extension in the RTP streams, as indicated in
Section 5.1. [I-D.ietf-mmusic-rid], Section 4). If simulcast is negotiated,
send the number of Source RTP Streams as specified in
[I-D.ietf-mmusic-sdp-simulcast], Section 6.2.2. If the
directional attribute is of type "recvonly" or "inactive", stop
transmitting RTP media, although RTCP should still be sent, 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.
skipping to change at page 58, line 19 skipping to change at page 61, line 17
Changes MUST NOT be silently ignored. Changes MUST NOT be silently ignored.
The following elements of the session description MUST NOT be changed The following elements of the session description MUST NOT be changed
between the createOffer and the setLocalDescription (or between the between the createOffer and the setLocalDescription (or between the
createAnswer and the setLocalDescription), since they reflect createAnswer and the setLocalDescription), since they reflect
transport attributes that are solely under browser control, and the transport attributes that are solely under browser control, and the
browser MUST NOT honor an attempt to change them: browser MUST NOT honor an attempt to change them:
o The number, type and port number of m= lines. o The number, type and port number of m= lines.
o The generated MID attributes (a=mid).
o The generated ICE credentials (a=ice-ufrag and a=ice-pwd). o The generated ICE credentials (a=ice-ufrag and a=ice-pwd).
o The set of ICE candidates and their parameters (a=candidate). o The set of ICE candidates and their parameters (a=candidate).
o The DTLS fingerprint(s) (a=fingerprint). o The DTLS fingerprint(s) (a=fingerprint).
o The contents of bundle groups, bundle-only parameters, or "a=rtcp- o The contents of bundle groups, bundle-only parameters, or "a=rtcp-
mux" parameters. mux" parameters.
The following modifications, if done by the browser to a description The following modifications, if done by the browser to a description
skipping to change at page 60, line 7 skipping to change at page 63, line 7
The flow shows Alice's browser initiating the session to Bob's The flow shows Alice's browser initiating the session to Bob's
browser. The messages from Alice's JS to Bob's JS are assumed to browser. The messages from Alice's JS to Bob's JS are assumed to
flow over some signaling protocol via a web server. The JS on both flow over some signaling protocol via a web server. The JS on both
Alice's side and Bob's side waits for all candidates before sending Alice's side and Bob's side waits for all candidates before sending
the offer or answer, so the offers and answers are complete. Trickle the offer or answer, so the offers and answers are complete. Trickle
ICE is not used. Both Alice and Bob are using the default policy of ICE is not used. Both Alice and Bob are using the default policy of
balanced. balanced.
// set up local media state // set up local media state
AliceJS->AliceUA: create new PeerConnection AliceJS->AliceUA: create new PeerConnection
AliceJS->AliceUA: addStream with stream containing audio and video AliceJS->AliceUA: addTrack with two tracks: one for audio and one for video
AliceJS->AliceUA: createOffer to get offer AliceJS->AliceUA: createOffer to get offer
AliceJS->AliceUA: setLocalDescription with offer AliceJS->AliceUA: setLocalDescription with offer
AliceUA->AliceJS: multiple onicecandidate events with candidates AliceUA->AliceJS: multiple onicecandidate events with candidates
// wait for ICE gathering to complete // wait for ICE gathering to complete
AliceUA->AliceJS: onicecandidate event with null candidate AliceUA->AliceJS: onicecandidate event with null candidate
AliceJS->AliceUA: get |offer-A1| from pendingLocalDescription AliceJS->AliceUA: get |offer-A1| from pendingLocalDescription
// |offer-A1| is sent over signaling protocol to Bob // |offer-A1| is sent over signaling protocol to Bob
AliceJS->WebServer: signaling with |offer-A1| AliceJS->WebServer: signaling with |offer-A1|
WebServer->BobJS: signaling with |offer-A1| WebServer->BobJS: signaling with |offer-A1|
// |offer-A1| arrives at Bob // |offer-A1| arrives at Bob
BobJS->BobUA: create a PeerConnection BobJS->BobUA: create a PeerConnection
BobJS->BobUA: setRemoteDescription with |offer-A1| BobJS->BobUA: setRemoteDescription with |offer-A1|
BobUA->BobJS: onaddstream event with remoteStream BobUA->BobJS: onaddstream event with remoteStream
// Bob accepts call // Bob accepts call
BobJS->BobUA: addStream with local media BobJS->BobUA: addTrack with local tracks
BobJS->BobUA: createAnswer BobJS->BobUA: createAnswer
BobJS->BobUA: setLocalDescription with answer BobJS->BobUA: setLocalDescription with answer
BobUA->BobJS: multiple onicecandidate events with candidates BobUA->BobJS: multiple onicecandidate events with candidates
// wait for ICE gathering to complete // wait for ICE gathering to complete
BobUA->BobJS: onicecandidate event with null candidate BobUA->BobJS: onicecandidate event with null candidate
BobJS->BobUA: get |answer-A1| from currentLocalDescription BobJS->BobUA: get |answer-A1| from currentLocalDescription
// |answer-A1| is sent over signaling protocol to Alice // |answer-A1| is sent over signaling protocol to Alice
BobJS->WebServer: signaling with |answer-A1| BobJS->WebServer: signaling with |answer-A1|
skipping to change at page 63, line 39 skipping to change at page 66, line 39
browsers setting up an audio channel and a data channel. Trickle ICE browsers setting up an audio channel and a data channel. Trickle ICE
is used in full trickle mode with a bundle policy of max-bundle, an is used in full trickle mode with a bundle policy of max-bundle, an
RTCP mux policy of require, and a single TURN server. Later, two RTCP mux policy of require, and a single TURN server. Later, two
video flows, one for the presenter and one for screen sharing, are video flows, one for the presenter and one for screen sharing, are
added to the session. This example shows Alice's browser initiating added to the session. This example shows Alice's browser initiating
the session to Bob's browser. The messages from Alice's JS to Bob's the session to Bob's browser. The messages from Alice's JS to Bob's
JS are assumed to flow over some signaling protocol via a web server. JS are assumed to flow over some signaling protocol via a web server.
// set up local media state // set up local media state
AliceJS->AliceUA: create new PeerConnection AliceJS->AliceUA: create new PeerConnection
AliceJS->AliceUA: addStream that contains audio track AliceJS->AliceUA: addTrack with an audio track
AliceJS->AliceUA: createDataChannel to get data channel AliceJS->AliceUA: createDataChannel to get data channel
AliceJS->AliceUA: createOffer to get |offer-B1| AliceJS->AliceUA: createOffer to get |offer-B1|
AliceJS->AliceUA: setLocalDescription with |offer-B1| AliceJS->AliceUA: setLocalDescription with |offer-B1|
// |offer-B1| is sent over signaling protocol to Bob // |offer-B1| is sent over signaling protocol to Bob
AliceJS->WebServer: signaling with |offer-B1| AliceJS->WebServer: signaling with |offer-B1|
WebServer->BobJS: signaling with |offer-B1| WebServer->BobJS: signaling with |offer-B1|
// |offer-B1| arrives at Bob // |offer-B1| arrives at Bob
BobJS->BobUA: create a PeerConnection BobJS->BobUA: create a PeerConnection
skipping to change at page 64, line 16 skipping to change at page 67, line 16
AliceJS->WebServer: signaling with |candidate-B1| AliceJS->WebServer: signaling with |candidate-B1|
AliceUA->AliceJS: onicecandidate event with |candidate-B2| (srflx) AliceUA->AliceJS: onicecandidate event with |candidate-B2| (srflx)
AliceJS->WebServer: signaling with |candidate-B2| AliceJS->WebServer: signaling with |candidate-B2|
WebServer->BobJS: signaling with |candidate-B1| WebServer->BobJS: signaling with |candidate-B1|
BobJS->BobUA: addIceCandidate with |candidate-B1| BobJS->BobUA: addIceCandidate with |candidate-B1|
WebServer->BobJS: signaling with |candidate-B2| WebServer->BobJS: signaling with |candidate-B2|
BobJS->BobUA: addIceCandidate with |candidate-B2| BobJS->BobUA: addIceCandidate with |candidate-B2|
// Bob accepts call // Bob accepts call
BobJS->BobUA: addStream with local audio stream BobJS->BobUA: addTrack with local audio
BobJS->BobUA: createDataChannel to get data channel BobJS->BobUA: createDataChannel to get data channel
BobJS->BobUA: createAnswer to get |answer-B1| BobJS->BobUA: createAnswer to get |answer-B1|
BobJS->BobUA: setLocalDescription with |answer-B1| BobJS->BobUA: setLocalDescription with |answer-B1|
// |answer-B1| is sent to Alice // |answer-B1| is sent to Alice
BobJS->WebServer: signaling with |answer-B1| BobJS->WebServer: signaling with |answer-B1|
WebServer->AliceJS: signaling with |answer-B1| WebServer->AliceJS: signaling with |answer-B1|
AliceJS->AliceUA: setRemoteDescription with |answer-B1| AliceJS->AliceUA: setRemoteDescription with |answer-B1|
AliceUA->AliceJS: onaddstream event with audio track from Bob AliceUA->AliceJS: onaddstream event with audio track from Bob
skipping to change at page 64, line 50 skipping to change at page 67, line 50
AliceUA->AliceJS: ondatachannel event AliceUA->AliceJS: ondatachannel event
BobUA->BobJS: onopen BobUA->BobJS: onopen
AliceUA->AliceJS: onopen AliceUA->AliceJS: onopen
// media is flowing between browsers // media is flowing between browsers
BobUA->AliceUA: audio+data sent from Bob to Alice BobUA->AliceUA: audio+data sent from Bob to Alice
AliceUA->BobUA: audio+data sent from Alice to Bob AliceUA->BobUA: audio+data sent from Alice to Bob
// some time later Bob adds two video streams // some time later Bob adds two video streams
// note, no candidates exchanged, because of bundle // note, no candidates exchanged, because of bundle
BobJS->BobUA: addStream with first video stream BobJS->BobUA: addTrack with first video stream
BobJS->BobUA: addStream with second video stream BobJS->BobUA: addTrack with second video stream
BobJS->BobUA: createOffer to get |offer-B2| BobJS->BobUA: createOffer to get |offer-B2|
BobJS->BobUA: setLocalDescription with |offer-B2| BobJS->BobUA: setLocalDescription with |offer-B2|
// |offer-B2| is sent to Alice // |offer-B2| is sent to Alice
BobJS->WebServer: signaling with |offer-B2| BobJS->WebServer: signaling with |offer-B2|
WebServer->AliceJS: signaling with |offer-B2| WebServer->AliceJS: signaling with |offer-B2|
AliceJS->AliceUA: setRemoteDescription with |offer-B2| AliceJS->AliceUA: setRemoteDescription with |offer-B2|
AliceUA->AliceJS: onaddstream event with first video stream AliceUA->AliceJS: onaddstream event with first video stream
AliceUA->AliceJS: onaddstream event with second video stream AliceUA->AliceJS: onaddstream event with second video stream
AliceJS->AliceUA: createAnswer to get |answer-B2| AliceJS->AliceUA: createAnswer to get |answer-B2|
skipping to change at page 75, line 48 skipping to change at page 78, line 48
[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.
11.2. Informative References 11.2. Informative References
[I-D.ietf-avtext-rid]
Roach, A., Nandakumar, S., and P. Thatcher, "RTP Stream
Identifier (RID) Source Description (SDES)", draft-ietf-
avtext-rid-00 (work in progress), February 2016.
[I-D.ietf-mmusic-rid]
Thatcher, P., Zanaty, M., Nandakumar, S., Burman, B.,
Roach, A., and B. Campen, "RTP Payload Format
Constraints", draft-ietf-mmusic-rid-04 (work in progress),
February 2016.
[I-D.ietf-mmusic-sdp-simulcast]
Burman, B., Westerlund, M., Nandakumar, S., and M. Zanaty,
"Using Simulcast in SDP and RTP Sessions", draft-ietf-
mmusic-sdp-simulcast-04 (work in progress), February 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.
[I-D.shieh-rtcweb-ip-handling] [I-D.shieh-rtcweb-ip-handling]
Shieh, G. and J. Uberti, "WebRTC IP Address Handling Shieh, G. and J. Uberti, "WebRTC IP Address Handling
Recommendations", draft-shieh-rtcweb-ip-handling-00 (work Recommendations", draft-shieh-rtcweb-ip-handling-00 (work
in progress), October 2015. in progress), October 2015.
skipping to change at page 77, line 22 skipping to change at page 80, line 35
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. Change log Appendix A. 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-14:
o Added discussion of RtpTransceivers + RtpSenders + RtpReceivers,
and how they interact with createOffer/createAnswer.
o Removed obsolete OfferToReceiveX options.
o Explained how addIceCandidate can be used for end-of-candidates.
Changes in draft-13: Changes in draft-13:
o Clarified which SDP lines can be ignored. o Clarified which SDP lines can be ignored.
o Clarified how to handle various received attributes. o Clarified how to handle various received attributes.
o Revised how atttributes should be generated for bundled m= lines. o Revised how atttributes should be generated for bundled m= lines.
o Remove unused references. o Remove unused references.
skipping to change at page 81, line 37 skipping to change at page 85, line 11
o Suggested API and examples have been moved to an appendix. o Suggested API and examples have been moved to an appendix.
Changes in draft -00: Changes in draft -00:
o Migrated from draft-uberti-rtcweb-jsep-02. o Migrated from draft-uberti-rtcweb-jsep-02.
Authors' Addresses Authors' Addresses
Justin Uberti Justin Uberti
Google Google
747 6th Ave S 747 6th St S
Kirkland, WA 98033 Kirkland, WA 98033
USA USA
Email: justin@uberti.name Email: justin@uberti.name
Cullen Jennings Cullen Jennings
Cisco Cisco
170 West Tasman Drive 170 West Tasman Drive
San Jose, CA 95134 San Jose, CA 95134
USA USA
 End of changes. 72 change blocks. 
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