draft-ietf-rtcweb-jsep-18.txt   draft-ietf-rtcweb-jsep-19.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: July 20, 2017 Cisco Expires: September 11, 2017 Cisco
E. Rescorla, Ed. E. Rescorla, Ed.
Mozilla Mozilla
January 16, 2017 March 10, 2017
Javascript Session Establishment Protocol Javascript Session Establishment Protocol
draft-ietf-rtcweb-jsep-18 draft-ietf-rtcweb-jsep-19
Abstract Abstract
This document describes the mechanisms for allowing a Javascript This document describes the mechanisms for allowing a Javascript
application to control the signaling plane of a multimedia session application to control the signaling plane of a multimedia session
via the interface specified in the W3C RTCPeerConnection API, and via the interface specified in the W3C RTCPeerConnection API, and
discusses how this relates to existing signaling protocols. discusses how this relates to existing signaling protocols.
Status of This Memo Status of This Memo
skipping to change at page 1, line 36 skipping to change at page 1, line 36
Internet-Drafts are working documents of the Internet Engineering Internet-Drafts are working documents of the Internet Engineering
Task Force (IETF). Note that other groups may also distribute Task Force (IETF). Note that other groups may also distribute
working documents as Internet-Drafts. The list of current Internet- working documents as Internet-Drafts. The list of current Internet-
Drafts is at http://datatracker.ietf.org/drafts/current/. Drafts is at http://datatracker.ietf.org/drafts/current/.
Internet-Drafts are draft documents valid for a maximum of six months Internet-Drafts are draft documents valid for a maximum of six months
and may be updated, replaced, or obsoleted by other documents at any and may be updated, replaced, or obsoleted by other documents at any
time. It is inappropriate to use Internet-Drafts as reference time. It is inappropriate to use Internet-Drafts as reference
material or to cite them other than as "work in progress." material or to cite them other than as "work in progress."
This Internet-Draft will expire on July 20, 2017. This Internet-Draft will expire on September 11, 2017.
Copyright Notice Copyright Notice
Copyright (c) 2017 IETF Trust and the persons identified as the Copyright (c) 2017 IETF Trust and the persons identified as the
document authors. All rights reserved. document authors. All rights reserved.
This document is subject to BCP 78 and the IETF Trust's Legal This document is subject to BCP 78 and the IETF Trust's Legal
Provisions Relating to IETF Documents Provisions Relating to IETF Documents
(http://trustee.ietf.org/license-info) in effect on the date of (http://trustee.ietf.org/license-info) in effect on the date of
publication of this document. Please review these documents publication of this document. Please review these documents
skipping to change at page 2, line 16 skipping to change at page 2, line 16
Table of Contents Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 4 1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 4
1.1. General Design of JSEP . . . . . . . . . . . . . . . . . 4 1.1. General Design of JSEP . . . . . . . . . . . . . . . . . 4
1.2. Other Approaches Considered . . . . . . . . . . . . . . . 5 1.2. Other Approaches Considered . . . . . . . . . . . . . . . 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 . . . . . . . . . . . . . . . 11
3.4. Session Description Control . . . . . . . . . . . . . . . 10 3.4. Session Description Control . . . . . . . . . . . . . . . 11
3.4.1. RtpTransceivers . . . . . . . . . . . . . . . . . . . 10 3.4.1. RtpTransceivers . . . . . . . . . . . . . . . . . . . 11
3.4.2. RtpSenders . . . . . . . . . . . . . . . . . . . . . 11 3.4.2. RtpSenders . . . . . . . . . . . . . . . . . . . . . 12
3.4.3. RtpReceivers . . . . . . . . . . . . . . . . . . . . 11 3.4.3. RtpReceivers . . . . . . . . . . . . . . . . . . . . 12
3.5. ICE . . . . . . . . . . . . . . . . . . . . . . . . . . . 11 3.5. ICE . . . . . . . . . . . . . . . . . . . . . . . . . . . 12
3.5.1. ICE Gathering Overview . . . . . . . . . . . . . . . 11 3.5.1. ICE Gathering Overview . . . . . . . . . . . . . . . 12
3.5.2. ICE Candidate Trickling . . . . . . . . . . . . . . . 12 3.5.2. ICE Candidate Trickling . . . . . . . . . . . . . . . 13
3.5.2.1. ICE Candidate Format . . . . . . . . . . . . . . 12 3.5.2.1. ICE Candidate Format . . . . . . . . . . . . . . 13
3.5.3. ICE Candidate Policy . . . . . . . . . . . . . . . . 13 3.5.3. ICE Candidate Policy . . . . . . . . . . . . . . . . 14
3.5.4. ICE Candidate Pool . . . . . . . . . . . . . . . . . 14 3.5.4. ICE Candidate Pool . . . . . . . . . . . . . . . . . 15
3.6. Video Size Negotiation . . . . . . . . . . . . . . . . . 15 3.6. Video Size Negotiation . . . . . . . . . . . . . . . . . 16
3.6.1. Creating an imageattr Attribute . . . . . . . . . . . 15 3.6.1. Creating an imageattr Attribute . . . . . . . . . . . 16
3.6.2. Interpreting an imageattr Attribute . . . . . . . . . 16 3.6.2. Interpreting an imageattr Attribute . . . . . . . . . 17
3.7. Simulcast . . . . . . . . . . . . . . . . . . . . . . . . 17 3.7. Simulcast . . . . . . . . . . . . . . . . . . . . . . . . 18
3.8. Interactions With Forking . . . . . . . . . . . . . . . . 18 3.8. Interactions With Forking . . . . . . . . . . . . . . . . 19
3.8.1. Sequential Forking . . . . . . . . . . . . . . . . . 19 3.8.1. Sequential Forking . . . . . . . . . . . . . . . . . 20
3.8.2. Parallel Forking . . . . . . . . . . . . . . . . . . 19 3.8.2. Parallel Forking . . . . . . . . . . . . . . . . . . 20
4. Interface . . . . . . . . . . . . . . . . . . . . . . . . . . 20 4. Interface . . . . . . . . . . . . . . . . . . . . . . . . . . 21
4.1. PeerConnection . . . . . . . . . . . . . . . . . . . . . 20 4.1. PeerConnection . . . . . . . . . . . . . . . . . . . . . 21
4.1.1. Constructor . . . . . . . . . . . . . . . . . . . . . 20 4.1.1. Constructor . . . . . . . . . . . . . . . . . . . . . 21
4.1.2. addTrack . . . . . . . . . . . . . . . . . . . . . . 22 4.1.2. addTrack . . . . . . . . . . . . . . . . . . . . . . 23
4.1.3. removeTrack . . . . . . . . . . . . . . . . . . . . . 23 4.1.3. removeTrack . . . . . . . . . . . . . . . . . . . . . 24
4.1.4. addTransceiver . . . . . . . . . . . . . . . . . . . 23 4.1.4. addTransceiver . . . . . . . . . . . . . . . . . . . 24
4.1.5. createDataChannel . . . . . . . . . . . . . . . . . . 23 4.1.5. createDataChannel . . . . . . . . . . . . . . . . . . 24
4.1.6. createOffer . . . . . . . . . . . . . . . . . . . . . 24 4.1.6. createOffer . . . . . . . . . . . . . . . . . . . . . 24
4.1.7. createAnswer . . . . . . . . . . . . . . . . . . . . 25 4.1.7. createAnswer . . . . . . . . . . . . . . . . . . . . 25
4.1.8. SessionDescriptionType . . . . . . . . . . . . . . . 25 4.1.8. SessionDescriptionType . . . . . . . . . . . . . . . 26
4.1.8.1. Use of Provisional Answers . . . . . . . . . . . 26 4.1.8.1. Use of Provisional Answers . . . . . . . . . . . 27
4.1.8.2. Rollback . . . . . . . . . . . . . . . . . . . . 27 4.1.8.2. Rollback . . . . . . . . . . . . . . . . . . . . 28
4.1.9. setLocalDescription . . . . . . . . . . . . . . . . . 28 4.1.9. setLocalDescription . . . . . . . . . . . . . . . . . 29
4.1.10. setRemoteDescription . . . . . . . . . . . . . . . . 28 4.1.10. setRemoteDescription . . . . . . . . . . . . . . . . 29
4.1.11. currentLocalDescription . . . . . . . . . . . . . . . 29 4.1.11. currentLocalDescription . . . . . . . . . . . . . . . 30
4.1.12. pendingLocalDescription . . . . . . . . . . . . . . . 29 4.1.12. pendingLocalDescription . . . . . . . . . . . . . . . 30
4.1.13. currentRemoteDescription . . . . . . . . . . . . . . 29 4.1.13. currentRemoteDescription . . . . . . . . . . . . . . 30
4.1.14. pendingRemoteDescription . . . . . . . . . . . . . . 29 4.1.14. pendingRemoteDescription . . . . . . . . . . . . . . 30
4.1.15. canTrickleIceCandidates . . . . . . . . . . . . . . . 30 4.1.15. canTrickleIceCandidates . . . . . . . . . . . . . . . 31
4.1.16. setConfiguration . . . . . . . . . . . . . . . . . . 30 4.1.16. setConfiguration . . . . . . . . . . . . . . . . . . 31
4.1.17. addIceCandidate . . . . . . . . . . . . . . . . . . . 31 4.1.17. addIceCandidate . . . . . . . . . . . . . . . . . . . 32
4.2. RtpTransceiver . . . . . . . . . . . . . . . . . . . . . 32 4.2. RtpTransceiver . . . . . . . . . . . . . . . . . . . . . 33
4.2.1. stop . . . . . . . . . . . . . . . . . . . . . . . . 32 4.2.1. stop . . . . . . . . . . . . . . . . . . . . . . . . 33
4.2.2. stopped . . . . . . . . . . . . . . . . . . . . . . . 32 4.2.2. stopped . . . . . . . . . . . . . . . . . . . . . . . 33
4.2.3. setDirection . . . . . . . . . . . . . . . . . . . . 32 4.2.3. setDirection . . . . . . . . . . . . . . . . . . . . 33
4.2.4. direction . . . . . . . . . . . . . . . . . . . . . . 32 4.2.4. direction . . . . . . . . . . . . . . . . . . . . . . 33
4.2.5. currentDirection . . . . . . . . . . . . . . . . . . 33 4.2.5. currentDirection . . . . . . . . . . . . . . . . . . 34
4.2.6. setCodecPreferences . . . . . . . . . . . . . . . . . 33 4.2.6. setCodecPreferences . . . . . . . . . . . . . . . . . 34
5. SDP Interaction Procedures . . . . . . . . . . . . . . . . . 33 5. SDP Interaction Procedures . . . . . . . . . . . . . . . . . 34
5.1. Requirements Overview . . . . . . . . . . . . . . . . . . 34 5.1. Requirements Overview . . . . . . . . . . . . . . . . . . 35
5.1.1. Implementation Requirements . . . . . . . . . . . . . 34 5.1.1. Usage Requirements . . . . . . . . . . . . . . . . . 35
5.1.2. Usage Requirements . . . . . . . . . . . . . . . . . 35 5.1.2. Profile Names and Interoperability . . . . . . . . . 35
5.1.3. Profile Names and Interoperability . . . . . . . . . 36 5.2. Constructing an Offer . . . . . . . . . . . . . . . . . . 36
5.2. Constructing an Offer . . . . . . . . . . . . . . . . . . 37 5.2.1. Initial Offers . . . . . . . . . . . . . . . . . . . 36
5.2.1. Initial Offers . . . . . . . . . . . . . . . . . . . 37 5.2.2. Subsequent Offers . . . . . . . . . . . . . . . . . . 43
5.2.2. Subsequent Offers . . . . . . . . . . . . . . . . . . 42 5.2.3. Options Handling . . . . . . . . . . . . . . . . . . 47
5.2.3. Options Handling . . . . . . . . . . . . . . . . . . 46 5.2.3.1. IceRestart . . . . . . . . . . . . . . . . . . . 47
5.2.3.1. IceRestart . . . . . . . . . . . . . . . . . . . 46 5.2.3.2. VoiceActivityDetection . . . . . . . . . . . . . 47
5.2.3.2. VoiceActivityDetection . . . . . . . . . . . . . 46 5.3. Generating an Answer . . . . . . . . . . . . . . . . . . 48
5.3. Generating an Answer . . . . . . . . . . . . . . . . . . 47 5.3.1. Initial Answers . . . . . . . . . . . . . . . . . . . 48
5.3.1. Initial Answers . . . . . . . . . . . . . . . . . . . 47 5.3.2. Subsequent Answers . . . . . . . . . . . . . . . . . 54
5.3.2. Subsequent Answers . . . . . . . . . . . . . . . . . 51 5.3.3. Options Handling . . . . . . . . . . . . . . . . . . 55
5.3.3. Options Handling . . . . . . . . . . . . . . . . . . 53 5.3.3.1. VoiceActivityDetection . . . . . . . . . . . . . 56
5.3.3.1. VoiceActivityDetection . . . . . . . . . . . . . 53 5.4. Modifying an Offer or Answer . . . . . . . . . . . . . . 56
5.4. Modifying an Offer or Answer . . . . . . . . . . . . . . 53 5.5. Processing a Local Description . . . . . . . . . . . . . 57
5.5. Processing a Local Description . . . . . . . . . . . . . 54 5.6. Processing a Remote Description . . . . . . . . . . . . . 57
5.6. Processing a Remote Description . . . . . . . . . . . . . 54 5.7. Parsing a Session Description . . . . . . . . . . . . . . 58
5.7. Parsing a Session Description . . . . . . . . . . . . . . 55 5.7.1. Session-Level Parsing . . . . . . . . . . . . . . . . 58
5.7.1. Session-Level Parsing . . . . . . . . . . . . . . . . 55 5.7.2. Media Section Parsing . . . . . . . . . . . . . . . . 60
5.7.2. Media Section Parsing . . . . . . . . . . . . . . . . 57 5.7.3. Semantics Verification . . . . . . . . . . . . . . . 62
5.7.3. Semantics Verification . . . . . . . . . . . . . . . 59 5.8. Applying a Local Description . . . . . . . . . . . . . . 64
5.8. Applying a Local Description . . . . . . . . . . . . . . 60 5.9. Applying a Remote Description . . . . . . . . . . . . . . 65
5.9. Applying a Remote Description . . . . . . . . . . . . . . 62 5.10. Applying an Answer . . . . . . . . . . . . . . . . . . . 69
5.10. Applying an Answer . . . . . . . . . . . . . . . . . . . 65 6. Processing RTP/RTCP . . . . . . . . . . . . . . . . . . . . . 71
6. Processing RTP/RTCP . . . . . . . . . . . . . . . . . . . . . 68 7. Examples . . . . . . . . . . . . . . . . . . . . . . . . . . 71
7. Examples . . . . . . . . . . . . . . . . . . . . . . . . . . 68 7.1. Simple Example . . . . . . . . . . . . . . . . . . . . . 72
7.1. Simple Example . . . . . . . . . . . . . . . . . . . . . 68 7.2. Detailed Example . . . . . . . . . . . . . . . . . . . . 77
7.2. Normal Examples . . . . . . . . . . . . . . . . . . . . . 72 7.3. Early Transport Warmup Example . . . . . . . . . . . . . 86
8. Security Considerations . . . . . . . . . . . . . . . . . . . 81 8. Security Considerations . . . . . . . . . . . . . . . . . . . 94
9. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 81 9. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 95
10. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 81 10. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 95
11. References . . . . . . . . . . . . . . . . . . . . . . . . . 82 11. References . . . . . . . . . . . . . . . . . . . . . . . . . 95
11.1. Normative References . . . . . . . . . . . . . . . . . . 82 11.1. Normative References . . . . . . . . . . . . . . . . . . 95
11.2. Informative References . . . . . . . . . . . . . . . . . 85 11.2. Informative References . . . . . . . . . . . . . . . . . 99
Appendix A. Appendix A . . . . . . . . . . . . . . . . . . . . . 87 Appendix A. Appendix A . . . . . . . . . . . . . . . . . . . . . 101
Appendix B. Appendix B . . . . . . . . . . . . . . . . . . . . . 88 Appendix B. Appendix B . . . . . . . . . . . . . . . . . . . . . 102
Appendix C. Change log . . . . . . . . . . . . . . . . . . . . . 91 Appendix C. Change log . . . . . . . . . . . . . . . . . . . . . 107
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 99 Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 116
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
skipping to change at page 4, line 28 skipping to change at page 4, line 28
applications may prefer to use different protocols, such as the applications may prefer to use different protocols, such as the
existing SIP or Jingle call signaling protocols, or something custom existing SIP or Jingle call signaling protocols, or something custom
to the particular application, perhaps for a novel use case. In this to the particular application, perhaps for a novel use case. In this
approach, the key information that needs to be exchanged is the approach, the key information that needs to be exchanged is the
multimedia session description, which specifies the necessary multimedia session description, which specifies the necessary
transport and media configuration information necessary to establish transport and media configuration information necessary to establish
the media plane. the media plane.
With these considerations in mind, this document describes the With these considerations in mind, this document describes the
Javascript Session Establishment Protocol (JSEP) that allows for full Javascript Session Establishment Protocol (JSEP) that allows for full
control of the signaling state machine from Javascript. JSEP removes control of the signaling state machine from Javascript. As described
the browser almost entirely from the core signaling flow, which is above, JSEP assumes a model in which a Javascript application
instead handled by the Javascript making use of two interfaces: (1) executes inside a runtime containing WebRTC APIs (the "JSEP
passing in local and remote session descriptions and (2) interacting implementation"). The JSEP implementation is almost entirely
with the ICE state machine. divorced from the core signaling flow, which is instead handled by
the Javascript making use of two interfaces: (1) passing in local and
remote session descriptions and (2) interacting with the ICE state
machine. The combination of the JSEP implementation and the
Javascript application is referred to throughout this document as a
"JSEP endpoint".
In this document, the use of JSEP is described as if it always occurs In this document, the use of JSEP is described as if it always occurs
between two browsers. Note though in many cases it will actually be between two JSEP endpoints. Note though in many cases it will
between a browser and some kind of server, such as a gateway or MCU. actually be between a JSEP endpoint and some kind of server, such as
This distinction is invisible to the browser; it just follows the a gateway or MCU. This distinction is invisible to the JSEP
instructions it is given via the API. endpoint; it just follows the instructions it is given via the API.
JSEP's handling of session descriptions is simple and JSEP's handling of session descriptions is simple and
straightforward. Whenever an offer/answer exchange is needed, the straightforward. Whenever an offer/answer exchange is needed, the
initiating side creates an offer by calling a createOffer() API. The initiating side creates an offer by calling a createOffer() API. The
application then uses that offer to set up its local config via the application then uses that offer to set up its local config via the
setLocalDescription() API. The offer is finally sent off to the setLocalDescription() API. The offer is finally sent off to the
remote side over its preferred signaling mechanism (e.g., remote side over its preferred signaling mechanism (e.g.,
WebSockets); upon receipt of that offer, the remote party installs it WebSockets); upon receipt of that offer, the remote party installs it
using the setRemoteDescription() API. using the setRemoteDescription() API.
To complete the offer/answer exchange, the remote party uses the To complete the offer/answer exchange, the remote party uses the
createAnswer() API to generate an appropriate answer, applies it createAnswer() API to generate an appropriate answer, applies it
using the setLocalDescription() API, and sends the answer back to the using the setLocalDescription() API, and sends the answer back to the
initiator over the signaling channel. When the initiator gets that initiator over the signaling channel. When the initiator gets that
answer, it installs it using the setRemoteDescription() API, and answer, it installs it using the setRemoteDescription() API, and
initial setup is complete. This process can be repeated for initial setup is complete. This process can be repeated for
additional offer/answer exchanges. additional offer/answer exchanges.
Regarding ICE [RFC5245], JSEP decouples the ICE state machine from Regarding ICE [RFC5245], JSEP decouples the ICE state machine from
the overall signaling state machine, as the ICE state machine must the overall signaling state machine, as the ICE state machine must
remain in the browser, because only the browser has the necessary remain in the JSEP implementation, because only the implementation
knowledge of candidates and other transport info. Performing this has the necessary knowledge of candidates and other transport info.
separation also provides additional flexibility; in protocols that Performing this separation also provides additional flexibility; in
decouple session descriptions from transport, such as Jingle, the protocols that decouple session descriptions from transport, such as
session description can be sent immediately and the transport Jingle, the session description can be sent immediately and the
information can be sent when available. In protocols that don't, transport information can be sent when available. In protocols that
such as SIP, the information can be used in the aggregated form. don't, such as SIP, the information can be used in the aggregated
Sending transport information separately can allow for faster ICE and form. Sending transport information separately can allow for faster
DTLS startup, since ICE checks can start as soon as any transport ICE and DTLS startup, since ICE checks can start as soon as any
information is available rather than waiting for all of it. transport information is available rather than waiting for all of it.
Through its abstraction of signaling, the JSEP approach does require Through its abstraction of signaling, the JSEP approach does require
the application to be aware of the signaling process. While the the application to be aware of the signaling process. While the
application does not need to understand the contents of session application does not need to understand the contents of session
descriptions to set up a call, the application must call the right descriptions to set up a call, the application must call the right
APIs at the right times, convert the session descriptions and ICE APIs at the right times, convert the session descriptions and ICE
information into the defined messages of its chosen signaling information into the defined messages of its chosen signaling
protocol, and perform the reverse conversion on the messages it protocol, and perform the reverse conversion on the messages it
receives from the other side. receives from the other side.
skipping to change at page 5, line 46 skipping to change at page 6, line 4
the JSEP API into an API suitable for a SIP or XMPP. Thus, JSEP the JSEP API into an API suitable for a SIP or XMPP. Thus, JSEP
provides greater control for the experienced developer without provides greater control for the experienced developer without
forcing any additional complexity on the novice developer. forcing any additional complexity on the novice developer.
1.2. Other Approaches Considered 1.2. Other Approaches Considered
One approach that was considered instead of JSEP was to include a One approach that was considered instead of JSEP was to include a
lightweight signaling protocol. Instead of providing session lightweight signaling protocol. Instead of providing session
descriptions to the API, the API would produce and consume messages descriptions to the API, the API would produce and consume messages
from this protocol. While providing a more high-level API, this put from this protocol. While providing a more high-level API, this put
more control of signaling within the browser, forcing the browser to more control of signaling within the JSEP implementation, forcing it
have to understand and handle concepts like signaling glare. In to have to understand and handle concepts like signaling glare. In
addition, it prevented the application from driving the state machine addition, it prevented the application from driving the state machine
to a desired state, as is needed in the page reload case. to a desired state, as is needed in the page reload case.
A second approach that was considered but not chosen was to decouple A second approach that was considered but not chosen was to decouple
the management of the media control objects from session the management of the media control objects from session
descriptions, instead offering APIs that would control each component descriptions, instead offering APIs that would control each component
directly. This was rejected based on a feeling that requiring directly. This was rejected based on a feeling that requiring
exposure of this level of complexity to the application programmer exposure of this level of complexity to the application programmer
would not be beneficial; it would result in an API where even a would not be beneficial; it would result in an API where even a
simple example would require a significant amount of code to simple example would require a significant amount of code to
orchestrate all the needed interactions, as well as creating a large orchestrate all the needed interactions, as well as creating a large
API surface that needed to be agreed upon and documented. In API surface that needed to be agreed upon and documented. In
addition, these API points could be called in any order, resulting in addition, these API points could be called in any order, resulting in
a more complex set of interactions with the media subsystem than the a more complex set of interactions with the media subsystem than the
JSEP approach, which specifies how session descriptions are to be JSEP approach, which specifies how session descriptions are to be
evaluated and applied. evaluated and applied.
One variation on JSEP that was considered was to keep the basic One variation on JSEP that was considered was to keep the basic
session description-oriented API, but to move the mechanism for session description-oriented API, but to move the mechanism for
generating offers and answers out of the browser. Instead of generating offers and answers out of the JSEP implementation.
providing createOffer/createAnswer methods within the browser, this Instead of providing createOffer/createAnswer methods within the
approach would instead expose a getCapabilities API which would implementation, this approach would instead expose a getCapabilities
provide the application with the information it needed in order to API which would provide the application with the information it
generate its own session descriptions. This increases the amount of needed in order to generate its own session descriptions. This
work that the application needs to do; it needs to know how to increases the amount of work that the application needs to do; it
generate session descriptions from capabilities, and especially how needs to know how to generate session descriptions from capabilities,
to generate the correct answer from an arbitrary offer and the and especially how to generate the correct answer from an arbitrary
supported capabilities. While this could certainly be addressed by offer and the supported capabilities. While this could certainly be
using a library like the one mentioned above, it basically forces the addressed by using a library like the one mentioned above, it
use of said library even for a simple example. Providing basically forces the use of said library even for a simple example.
createOffer/createAnswer avoids this problem, but still allows Providing createOffer/createAnswer avoids this problem.
applications to generate their own offers/answers (to a large extent)
if they choose, using the description generated by createOffer as an
indication of the browser's capabilities.
2. Terminology 2. Terminology
The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT", The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
"SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
document are to be interpreted as described in [RFC2119]. document are to be interpreted as described in [RFC2119].
3. Semantics and Syntax 3. Semantics and Syntax
3.1. Signaling Model 3.1. Signaling Model
JSEP does not specify a particular signaling model or state machine, JSEP does not specify a particular signaling model or state machine,
other than the generic need to exchange session descriptions in the other than the generic need to exchange session descriptions in the
fashion described by [RFC3264](offer/answer) in order for both sides fashion described by [RFC3264](offer/answer) in order for both sides
of the session to know how to conduct the session. JSEP provides of the session to know how to conduct the session. JSEP provides
mechanisms to create offers and answers, as well as to apply them to mechanisms to create offers and answers, as well as to apply them to
a session. However, the browser is totally decoupled from the actual a session. However, the JSEP implementation is totally decoupled
mechanism by which these offers and answers are communicated to the from the actual mechanism by which these offers and answers are
remote side, including addressing, retransmission, forking, and glare communicated to the remote side, including addressing,
handling. These issues are left entirely up to the application; the retransmission, forking, and glare handling. These issues are left
application has complete control over which offers and answers get entirely up to the application; the application has complete control
handed to the browser, and when. over which offers and answers get handed to the implementation, and
when.
+-----------+ +-----------+ +-----------+ +-----------+
| Web App |<--- App-Specific Signaling -->| Web App | | Web App |<--- App-Specific Signaling -->| Web App |
+-----------+ +-----------+ +-----------+ +-----------+
^ ^ ^ ^
| SDP | SDP | SDP | SDP
V V V V
+-----------+ +-----------+ +-----------+ +-----------+
| Browser |<----------- Media ------------>| Browser | | JSEP |<----------- Media ------------>| JSEP |
| Impl. | | Impl. |
+-----------+ +-----------+ +-----------+ +-----------+
Figure 1: JSEP Signaling Model Figure 1: JSEP Signaling Model
3.2. Session Descriptions and State Machine 3.2. Session Descriptions and State Machine
In order to establish the media plane, the user agent needs specific In order to establish the media plane, the user agent needs specific
parameters to indicate what to transmit to the remote side, as well parameters to indicate what to transmit to the remote side, as well
as how to handle the media that is received. These parameters are as how to handle the media that is received. These parameters are
determined by the exchange of session descriptions in offers and determined by the exchange of session descriptions in offers and
answers, and there are certain details to this process that must be answers, and there are certain details to this process that must be
handled in the JSEP APIs. handled in the JSEP APIs.
Whether a session description applies to the local side or the remote Whether a session description applies to the local side or the remote
side affects the meaning of that description. For example, the list side affects the meaning of that description. For example, the list
of codecs sent to a remote party indicates what the local side is of codecs sent to a remote party indicates what the local side is
willing to receive, which, when intersected with the set of codecs willing to receive, which, when intersected with the set of codecs
the remote side supports, specifies what the remote side should send. the remote side supports, specifies what the remote side should send.
However, not all parameters follow this rule; for example, the DTLS- However, not all parameters follow this rule; for example, the
SRTP parameters [RFC5763] sent to a remote party indicate what fingerprints [I-D.ietf-mmusic-4572-update] sent to a remote party are
certificate the local side will use in DTLS setup, and thereby what calculated based on the local certificate(s) offered; the remote
the remote party should expect to receive; the remote party will have party MUST either accept these parameters or reject them altogether,
to accept these parameters, with no option to choose different with no option to choose different values.
values.
In addition, various RFCs put different conditions on the format of In addition, various RFCs put different conditions on the format of
offers versus answers. For example, an offer may propose an offers versus answers. For example, an offer may propose an
arbitrary number of media streams (i.e. m= sections), but an answer arbitrary number of m= sections (i.e., media descriptions as
must contain the exact same number as the offer. described in [RFC4566], Section 5.14), but an answer must contain the
exact same number as the offer.
Lastly, while the exact media parameters are only known only after an Lastly, while the exact media parameters are only known only after an
offer and an answer have been exchanged, it is possible for the offer and an answer have been exchanged, it is possible for the
offerer to receive media after they have sent an offer and before offerer to receive media after they have sent an offer and before
they have received an answer. To properly process incoming media in they have received an answer. To properly process incoming media in
this case, the offerer's media handler must be aware of the details this case, the offerer's media handler must be aware of the details
of the offer before the answer arrives. of the offer before the answer arrives.
Therefore, in order to handle session descriptions properly, the user Therefore, in order to handle session descriptions properly, the user
agent needs: agent needs:
skipping to change at page 10, line 34 skipping to change at page 11, line 34
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. change them.
3.4. Session Description Control 3.4. Session Description Control
In order to give the application control over various common session In order to give the application control over various common session
parameters, JSEP provides control surfaces which tell the browser how parameters, JSEP provides control surfaces which tell the JSEP
to generate session descriptions. This avoids the need for implementation how to generate session descriptions. This avoids the
Javascript to modify session descriptions in most cases. need for Javascript to modify session descriptions in most cases.
Changes to these objects result in changes to the session Changes to these objects result in changes to the session
descriptions generated by subsequent createOffer/Answer calls. descriptions generated by subsequent createOffer/Answer calls.
3.4.1. RtpTransceivers 3.4.1. RtpTransceivers
RtpTransceivers allow the application to control the RTP media RtpTransceivers allow the application to control the RTP media
associated with one m= section. Each RtpTransceiver has an RtpSender associated with one m= section. Each RtpTransceiver has an RtpSender
and an RtpReceiver, which an application can use to control the and an RtpReceiver, which an application can use to control the
sending and receiving of RTP media. The application may also modify sending and receiving of RTP media. The application may also modify
skipping to change at page 11, line 44 skipping to change at page 12, line 44
3.5. ICE 3.5. ICE
3.5.1. ICE Gathering Overview 3.5.1. ICE Gathering Overview
JSEP gathers ICE candidates as needed by the application. Collection JSEP gathers ICE candidates as needed by the application. Collection
of ICE candidates is referred to as a gathering phase, and this is of ICE candidates is referred to as a gathering phase, and this is
triggered either by the addition of a new or recycled m= section to triggered either by the addition of a new or recycled m= section to
the local session description, or new ICE credentials in the the local session description, or new ICE credentials in the
description, indicating an ICE restart. Use of new ICE credentials description, indicating an ICE restart. Use of new ICE credentials
can be triggered explicitly by the application, or implicitly by the can be triggered explicitly by the application, or implicitly by the
browser in response to changes in the ICE configuration. JSEP implementation in response to changes in the ICE configuration.
When the ICE configuration changes in a way that requires a new When the ICE configuration changes in a way that requires a new
gathering phase, a 'needs-ice-restart' bit is set. When this bit is gathering phase, a 'needs-ice-restart' bit is set. When this bit is
set, calls to the createOffer API will generate new ICE credentials. set, calls to the createOffer API will generate new ICE credentials.
This bit is cleared by a call to the setLocalDescription API with new This bit is cleared by a call to the setLocalDescription API with new
ICE credentials from either an offer or an answer, i.e., from either ICE credentials from either an offer or an answer, i.e., from either
a local- or remote-initiated ICE restart. a local- or remote-initiated ICE restart.
When a new gathering phase starts, the ICE Agent will notify the When a new gathering phase starts, the ICE agent will notify the
application that gathering is occurring through an event. Then, when application that gathering is occurring through an event. Then, when
each new ICE candidate becomes available, the ICE Agent will supply each new ICE candidate becomes available, the ICE agent will supply
it to the application via an additional event; these candidates will it to the application via an additional event; these candidates will
also automatically be added to the current and/or pending local also automatically be added to the current and/or pending local
session description. Finally, when all candidates have been session description. Finally, when all candidates have been
gathered, an event will be dispatched to signal that the gathering gathered, an event will be dispatched to signal that the gathering
process is complete. process is complete.
Note that gathering phases only gather the candidates needed by Note that gathering phases only gather the candidates needed by
new/recycled/restarting m= sections; other m= sections continue to new/recycled/restarting m= sections; other m= sections continue to
use their existing candidates. Also, when bundling is active, use their existing candidates. Also, when bundling is active,
candidates are only gathered (and exchanged) for the m= sections candidates are only gathered (and exchanged) for the m= sections
skipping to change at page 12, line 43 skipping to change at page 13, line 43
JSEP supports optional candidate trickling by providing APIs, as JSEP supports optional candidate trickling by providing APIs, as
described above, that provide control and feedback on the ICE described above, that provide control and feedback on the ICE
candidate gathering process. Applications that support candidate candidate gathering process. Applications that support candidate
trickling can send the initial offer immediately and send individual trickling can send the initial offer immediately and send individual
candidates when they get the notified of a new candidate; candidates when they get 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.
3.5.2.1. ICE Candidate Format 3.5.2.1. ICE Candidate Format
In JSEP, ICE candidates are abstracted by an IceCandidate object, and In JSEP, ICE candidates are abstracted by an IceCandidate object, and
as with session descriptions, SDP syntax is used for the internal as with session descriptions, SDP syntax is used for the internal
representation. representation.
The candidate details are specified in an IceCandidate field, using The candidate details are specified in an IceCandidate field, using
the same SDP syntax as the "candidate-attribute" field defined in the same SDP syntax as the "candidate-attribute" field defined in
skipping to change at page 13, line 38 skipping to change at page 14, line 38
answerer when interacting with a non-JSEP endpoint that does not answerer when interacting with a non-JSEP endpoint that does not
support the MID attribute, as discussed in Section 5.9 below). If support the MID attribute, as discussed in Section 5.9 below). If
the MID field is present in a received IceCandidate, it MUST be used the MID field is present in a received IceCandidate, it MUST be used
for identification; otherwise, the m= section index is used instead. for identification; otherwise, the m= section index is used instead.
When creating an IceCandidate object, JSEP implementations MUST When creating an IceCandidate object, JSEP implementations MUST
populate all of these fields. populate all of these fields.
3.5.3. ICE Candidate Policy 3.5.3. ICE Candidate Policy
Typically, when gathering ICE candidates, the browser will gather all Typically, when gathering ICE candidates, the JSEP implementation
possible forms of initial candidates - host, server reflexive, and will gather all possible forms of initial candidates - host, server
relay. However, in certain cases, applications may want to have more reflexive, and relay. However, in certain cases, applications may
specific control over the gathering process, due to privacy or want to have more specific control over the gathering process, due to
related concerns. For example, one may want to only use relay privacy or related concerns. For example, one may want to only use
candidates, to leak as little location information as possible relay candidates, to leak as little location information as possible
(keeping in mind that this choice comes with corresponding (keeping in mind that this choice comes with corresponding
operational costs). To accomplish this, JSEP allows the application operational costs). To accomplish this, JSEP allows the application
to restrict which ICE candidates are used in a session. Note that to restrict which ICE candidates are used in a session. Note that
this filtering is applied on top of any restrictions the browser this filtering is applied on top of any restrictions the
chooses to enforce regarding which IP addresses are permitted for the implementation chooses to enforce regarding which IP addresses are
application, as discussed in [I-D.ietf-rtcweb-ip-handling]. permitted for the application, as discussed in
[I-D.ietf-rtcweb-ip-handling].
There may also be cases where the application wants to change which There may also be cases where the application wants to change which
types of candidates are used while the session is active. A prime types of candidates are used while the session is active. A prime
example is where a callee may initially want to use only relay example is where a callee may initially want to use only relay
candidates, to avoid leaking location information to an arbitrary candidates, to avoid leaking location information to an arbitrary
caller, but then change to use all candidates (for lower operational caller, but then change to use all candidates (for lower operational
cost) once the user has indicated they want to take the call. For cost) once the user has indicated they want to take the call. For
this scenario, the browser MUST allow the candidate policy to be this scenario, the JSEP implementation MUST allow the candidate
changed in mid-session, subject to the aforementioned interactions policy to be changed in mid-session, subject to the aforementioned
with local policy. interactions with local policy.
To administer the ICE candidate policy, the browser will determine To administer the ICE candidate policy, the JSEP implementation will
the current setting at the start of each gathering phase. Then, determine the current setting at the start of each gathering phase.
during the gathering phase, the browser MUST NOT expose candidates Then, during the gathering phase, the implementation MUST NOT expose
disallowed by the current policy to the application, use them as the candidates disallowed by the current policy to the application, use
source of connectivity checks, or indirectly expose them via other them as the source of connectivity checks, or indirectly expose them
fields, such as the raddr/rport attributes for other ICE candidates. via other fields, such as the raddr/rport attributes for other ICE
Later, if a different policy is specified by the application, the candidates. Later, if a different policy is specified by the
application can apply it by kicking off a new gathering phase via an application, the application can apply it by kicking off a new
ICE restart. gathering phase via an ICE restart.
3.5.4. ICE Candidate Pool 3.5.4. ICE Candidate Pool
JSEP applications typically inform the browser to begin ICE gathering JSEP applications typically inform the JSEP implementation to begin
via the information supplied to setLocalDescription, as this is where ICE gathering via the information supplied to setLocalDescription, as
the app specifies the number of media streams, and thereby ICE this is where the app specifies the number of media streams, and
components, for which to gather candidates. However, to accelerate thereby ICE components, for which to gather candidates. However, to
cases where the application knows the number of ICE components to use accelerate cases where the application knows the number of ICE
ahead of time, it may ask the browser to gather a pool of potential components to use ahead of time, it may ask the implementation to
ICE candidates to help ensure rapid media setup. gather a pool of potential ICE candidates to help ensure rapid media
setup.
When setLocalDescription is eventually called, and the browser goes When setLocalDescription is eventually called, and the JSEP
to gather the needed ICE candidates, it SHOULD start by checking if implementation goes to gather the needed ICE candidates, it SHOULD
any candidates are available in the pool. If there are candidates in start by checking if any candidates are available in the pool. If
the pool, they SHOULD be handed to the application immediately via there are candidates in the pool, they SHOULD be handed to the
the ICE candidate event. If the pool becomes depleted, either application immediately via the ICE candidate event. If the pool
because a larger-than-expected number of ICE components is used, or becomes depleted, either because a larger-than-expected number of ICE
because the pool has not had enough time to gather candidates, the components is used, or because the pool has not had enough time to
remaining candidates are gathered as usual. This only occurs for the gather candidates, the remaining candidates are gathered as usual.
first offer/answer exchange, after which the candidate pool is This only occurs for the first offer/answer exchange, after which the
emptied and no longer used. candidate pool is emptied and no longer used.
One example of where this concept is useful is an application that One example of where this concept is useful is an application that
expects an incoming call at some point in the future, and wants to expects an incoming call at some point in the future, and wants to
minimize the time it takes to establish connectivity, to avoid minimize the time it takes to establish connectivity, to avoid
clipping of initial media. By pre-gathering candidates into the clipping of initial media. By pre-gathering candidates into the
pool, it can exchange and start sending connectivity checks from pool, it can exchange and start sending connectivity checks from
these candidates almost immediately upon receipt of a call. Note these candidates almost immediately upon receipt of a call. Note
though that by holding on to these pre-gathered candidates, which though that by holding on to these pre-gathered candidates, which
will be kept alive as long as they may be needed, the application will be kept alive as long as they may be needed, the application
will consume resources on the STUN/TURN servers it is using. will consume resources on the STUN/TURN servers it is using.
3.6. Video Size Negotiation 3.6. Video Size Negotiation
Video size negotiation is the process through which a receiver can Video size negotiation is the process through which a receiver can
use the "a=imageattr" SDP attribute [RFC6236] to indicate what video use the "a=imageattr" SDP attribute [RFC6236] to indicate what video
frame sizes it is capable of receiving. A receiver may have hard frame sizes it is capable of receiving. A receiver may have hard
limits on what its video decoder can process, or it may wish to limits on what its video decoder can process, or it may have some
constrain what it receives due to application preferences, e.g. a maximum set by policy.
specific size for the window in which the video will be displayed.
Note that certain codecs support transmission of samples with aspect Note that certain codecs support transmission of samples with aspect
ratios other than 1.0 (i.e., non-square pixels). JSEP ratios other than 1.0 (i.e., non-square pixels). JSEP
implementations will not transmit non-square pixels, but SHOULD implementations will not transmit non-square pixels, but SHOULD
receive and render such video with the correct aspect ratio. receive and render such video with the correct aspect ratio.
However, sample aspect ratio has no impact on the size negotiation However, sample aspect ratio has no impact on the size negotiation
described below; all dimensions are measured in pixels, whether described below; all dimensions are measured in pixels, whether
square or not. square or not.
3.6.1. Creating an imageattr Attribute 3.6.1. Creating an imageattr Attribute
In order to determine the limits on what video resolution a receiver The receiver will first intersect any known local limits (e.g.,
wants to receive, it will intersect its decoder hard limits with any hardware decoder capababilities, local policy) to determine the
mandatory constraints that have been applied to the associated absolute minimum and maximum sizes it can receive. If there are no
MediaStreamTrack. If the decoder limits are unknown, e.g. when using known local limits, the "a=imageattr" attribute SHOULD be omitted.
a software decoder, the mandatory constraints are used directly. For
the answerer, these mandatory constraints can be applied to the
remote MediaStreamTracks that are created by a setRemoteDescription
call, and will affect the output of the ensuing createAnswer call.
Any constraints set after setLocalDescription is used to set the
answer will result in a new offer-answer exchange. For the offerer,
because it does not know about any remote MediaStreamTracks until it
receives the answer, the offer can only reflect decoder hard limits.
If the offerer wishes to set mandatory constraints on video
resolution, it must do so after receiving the answer, and the result
will be a new offer-answer to communicate them.
If there are no known decoder limits or mandatory constraints, the
"a=imageattr" attribute SHOULD be omitted.
Otherwise, an "a=imageattr" attribute is created with "recv" Otherwise, an "a=imageattr" attribute is created with "recv"
direction, and the resulting resolution space formed by intersecting direction, and the resulting resolution space formed from the
the decoder limits and constraints is used to specify its minimum and aforementioned intersection is used to specify its minimum and
maximum x= and y= values. If the intersection is the null set, i.e., maximum x= and y= values. If the intersection is the null set, i.e.,
there are no resolutions that are permitted by both the decoder and the degenerate case of no permitted resolutions, this MUST be
the mandatory constraints, this MUST be represented by x=0 and y=0 represented by x=0 and y=0 values.
values.
The rules here express a single set of preferences, and therefore, The rules here express a single set of preferences, and therefore,
the "a=imageattr" q= value is not important. It SHOULD be set to the "a=imageattr" q= value is not important. It SHOULD be set to
1.0. 1.0.
The "a=imageattr" field is payload type specific. When all video The "a=imageattr" field is payload type specific. When all video
codecs supported have the same capabilities, use of a single codecs supported have the same capabilities, use of a single
attribute, with the wildcard payload type (*), is RECOMMENDED. attribute, with the wildcard payload type (*), is RECOMMENDED.
However, when the supported video codecs have differing capabilities, However, when the supported video codecs have different limitations,
specific "a=imageattr" attributes MUST be inserted for each payload specific "a=imageattr" attributes MUST be inserted for each payload
type. type.
As an example, consider a system with a multiformat video decoder, As an example, consider a system with a multiformat video decoder,
which is capable of decoding any resolution from 48x48 to 720p, and which is capable of decoding any resolution from 48x48 to 720p, In
where the application has constrained the received track to at most this case, the implementation would generate this attribute:
360p. In this case, the implementation would generate this
attribute:
a=imageattr:* recv [x=[48:640],y=[48:360],q=1.0] a=imageattr:* recv [x=[48:1280],y=[48:720],q=1.0]
This declaration indicates that the receiver is capable of decoding This declaration indicates that the receiver is capable of decoding
any image resolution from 48x48 up to 640x360 pixels. any image resolution from 48x48 up to 1280x720 pixels.
3.6.2. Interpreting an imageattr Attribute 3.6.2. Interpreting an imageattr Attribute
[RFC6236] defines "a=imageattr" to be an advisory field. This means [RFC6236] defines "a=imageattr" to be an advisory field. This means
that it does not absolutely constrain the video formats that the that it does not absolutely constrain the video formats that the
sender can use, but gives an indication of the preferred values. sender can use, but gives an indication of the preferred values.
This specification prescribes more specific behavior. When a sender This specification prescribes more specific behavior. When a 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
appropriate). Note that when considering a MediaStreamTrack that is appropriate). Note that when considering a MediaStreamTrack that is
producing rotated video, the unrotated resolution MUST be used. This producing rotated video, the unrotated resolution MUST be used. This
is required regardless of whether the receiver supports performing is required regardless of whether the receiver supports performing
receive-side rotation (e.g., through CVO), as it significantly receive-side rotation (e.g., through CVO [TS26.114]), as it
simplifies the matching logic. significantly simplifies the matching logic.
For the purposes of resolution negotiation, only size limits are For the purposes of resolution negotiation, only size limits are
considered. Any other values, e.g. picture or sample aspect ratio, considered. Any other values, e.g. picture or sample aspect ratio,
MUST be ignored. MUST be ignored.
When communicating with a non-JSEP endpoint, multiple relevant When communicating with a non-JSEP endpoint, multiple relevant
"a=imageattr recv" attributes may be present in a received m= "a=imageattr recv" attributes may be present in a received m=
section. If this occurs, attributes other than the one with the section. If this occurs, attributes other than the one with the
highest "q=" value MUST be ignored. If multiple attributes have the highest "q=" value MUST be ignored. If multiple attributes have the
same "q=" value, those that appear after the first such attribute in same "q=" value, those that appear after the first such attribute in
skipping to change at page 20, line 51 skipping to change at page 21, line 36
4.1.1. Constructor 4.1.1. Constructor
The PeerConnection constructor allows the application to specify The PeerConnection constructor allows the application to specify
global parameters for the media session, such as the STUN/TURN global parameters for the media session, such as the STUN/TURN
servers and credentials to use when gathering candidates, as well as servers and credentials to use when gathering candidates, as well as
the initial ICE candidate policy and pool size, and also the bundle the initial ICE candidate policy and pool size, and also the bundle
policy to use. policy to use.
If an ICE candidate policy is specified, it functions as described in If an ICE candidate policy is specified, it functions as described in
Section 3.5.3, causing the browser to only surface the permitted Section 3.5.3, causing the JSEP implementation to only surface the
candidates (including any internal browser filtering) to the permitted candidates (including any implementation-internal
application, and only use those candidates for connectivity checks. filtering) to the application, and only use those candidates for
The set of available policies is as follows: connectivity checks. The set of available policies is as follows:
all: All candidates permitted by browser policy will be gathered and all: All candidates permitted by implementation policy will be
used. gathered and used.
relay: All candidates except relay candidates will be filtered out. relay: All candidates except relay candidates will be filtered out.
This obfuscates the location information that might be ascertained This obfuscates the location information that might be ascertained
by the remote peer from the received candidates. Depending on how by the remote peer from the received candidates. Depending on how
the application deploys and chooses relay servers, this could the application deploys and chooses relay servers, this could
obfuscate location to a metro or possibly even global level. obfuscate location to a metro or possibly even global level.
The default ICE candidate policy MUST be set to "all" as this is The default ICE candidate policy MUST be set to "all" as this is
generally the desired policy, and also typically reduces use of generally the desired policy, and also typically reduces use of
application TURN server resources significantly. application TURN server resources significantly.
skipping to change at page 21, line 32 skipping to change at page 22, line 20
number of ICE components to pre-gather candidates for. Because pre- number of ICE components to pre-gather candidates for. Because pre-
gathering results in utilizing STUN/TURN server resources for gathering results in utilizing STUN/TURN server resources for
potentially long periods of time, this must only occur upon potentially long periods of time, this must only occur upon
application request, and therefore the default candidate pool size application request, and therefore the default candidate pool size
MUST be zero. MUST be zero.
The application can specify its preferred policy regarding use of The application can specify its preferred policy regarding use of
bundle, the multiplexing mechanism defined in bundle, the multiplexing mechanism defined in
[I-D.ietf-mmusic-sdp-bundle-negotiation]. Regardless of policy, the [I-D.ietf-mmusic-sdp-bundle-negotiation]. Regardless of policy, the
application will always try to negotiate bundle onto a single application will always try to negotiate bundle onto a single
transport, and will offer a single bundle group across all media transport, and will offer a single bundle group across all m=
section; use of this single transport is contingent upon the answerer sections; use of this single transport is contingent upon the
accepting bundle. However, by specifying a policy from the list answerer accepting bundle. However, by specifying a policy from the
below, the application can control exactly how aggressively it will list below, the application can control exactly how aggressively it
try to bundle media streams together, which affects how it will will try to bundle media streams together, which affects how it will
interoperate with a non-bundle-aware endpoint. When negotiating with interoperate with a non-bundle-aware endpoint. When negotiating with
a non-bundle-aware endpoint, only the streams not marked as bundle- a non-bundle-aware endpoint, only the streams not marked as bundle-
only streams will be established. only streams will be established.
The set of available policies is as follows: The set of available policies is as follows:
balanced: The first media section of each type (audio, video, or balanced: The first m= section of each type (audio, video, or
application) will contain transport parameters, which will allow application) will contain transport parameters, which will allow
an answerer to unbundle that section. The second and any an answerer to unbundle that section. The second and any
subsequent media section of each type will be marked bundle-only. subsequent m= section of each type will be marked bundle-only.
The result is that if there are N distinct media types, then The result is that if there are N distinct media types, then
candidates will be gathered for for N media streams. This policy candidates will be gathered for for N media streams. This policy
balances desire to multiplex with the need to ensure basic audio balances desire to multiplex with the need to ensure basic audio
and video can still be negotiated in legacy cases. When acting as and video can still be negotiated in legacy cases. When acting as
answerer, if there is no bundle group in the offer, the answerer, if there is no bundle group in the offer, the
implementation will reject all but the first m= section of each implementation will reject all but the first m= section of each
type. type.
max-compat: All media sections will contain transport parameters; max-compat: All m= sections will contain transport parameters; none
none will be marked as bundle-only. This policy will allow all will be marked as bundle-only. This policy will allow all streams
streams to be received by non-bundle-aware endpoints, but require to be received by non-bundle-aware endpoints, but require separate
separate candidates to be gathered for each media stream. candidates to be gathered for each media stream.
max-bundle: Only the first media section will contain transport max-bundle: Only the first m= section will contain transport
parameters; all streams other than the first will be marked as parameters; all streams other than the first will be marked as
bundle-only. This policy aims to minimize candidate gathering and bundle-only. This policy aims to minimize candidate gathering and
maximize multiplexing, at the cost of less compatibility with maximize multiplexing, at the cost of less compatibility with
legacy endpoints. When acting as answerer, the implementation legacy endpoints. When acting as answerer, the implementation
will reject any m= sections other than the first m= section, will reject any m= sections other than the first m= section,
unless they are in the same bundle group as that m= section. unless they are in the same bundle group as that m= section.
As it provides the best tradeoff between performance and As it provides the best tradeoff between performance and
compatibility with legacy endpoints, the default bundle policy MUST compatibility with legacy endpoints, the default bundle policy MUST
be set to "balanced". be set to "balanced".
The application can specify its preferred policy regarding use of The application can specify its preferred policy regarding use of
RTP/RTCP multiplexing [RFC5761] using one of the following policies: RTP/RTCP multiplexing [RFC5761] using one of the following policies:
negotiate: The browser will gather both RTP and RTCP candidates but negotiate: The JSEP implementation will gather both RTP and RTCP
also will offer "a=rtcp-mux", thus allowing for compatibility with candidates but also will offer "a=rtcp-mux", thus allowing for
either multiplexing or non-multiplexing endpoints. compatibility with either multiplexing or non-multiplexing
endpoints.
require: The browser will only gather RTP candidates. This halves require: The JSEP implementation will only gather RTP candidates and
the number of candidates that the offerer needs to gather. will insert an "a=rtcp-mux-only" indication into any new m=
Applying a description with an m= section that does not contain an sections in offers it generates. This halves the number of
"a=rtcp-mux" attribute will cause an error to be returned. candidates that the offerer needs to gather. Applying a
description with an m= section that does not contain an "a=rtcp-
mux" attribute will cause an error to be returned.
The default multiplexing policy MUST be set to "require". The default multiplexing policy MUST be set to "require".
Implementations MAY choose to reject attempts by the application to Implementations MAY choose to reject attempts by the application to
set the multiplexing policy to "negotiate". set the multiplexing policy to "negotiate".
4.1.2. addTrack 4.1.2. addTrack
The addTrack method adds a MediaStreamTrack to the PeerConnection, The addTrack method adds a MediaStreamTrack to the PeerConnection,
using the MediaStream argument to associate the track with other using the MediaStream argument to associate the track with other
tracks in the same MediaStream, so that they can be added to the same tracks in the same MediaStream, so that they can be added to the same
skipping to change at page 23, line 17 skipping to change at page 24, line 11
call to setRemoteDescription() and does not have a local track. call to setRemoteDescription() and does not have a local track.
Otherwise, a new transceiver will be created, as described in Otherwise, a new transceiver will be created, as described in
Section 4.1.4. Section 4.1.4.
4.1.3. removeTrack 4.1.3. removeTrack
The removeTrack method removes a MediaStreamTrack from the The removeTrack method removes a MediaStreamTrack from the
PeerConnection, using the RtpSender argument to indicate which sender PeerConnection, using the RtpSender argument to indicate which sender
should have its track removed. The sender's track is cleared, and should have its track removed. The sender's track is cleared, and
the sender stops sending. Future calls to createOffer will mark the the sender stops sending. Future calls to createOffer will mark the
media description associated with the sender as recvonly (if m= section associated with the sender as recvonly (if
transceiver.currentDirection is sendrecv) or as inactive (if transceiver.currentDirection is sendrecv) or as inactive (if
transceiver.currentDirection is sendonly). transceiver.currentDirection is sendonly).
4.1.4. addTransceiver 4.1.4. addTransceiver
The addTransceiver method adds a new RtpTransceiver to the The addTransceiver method adds a new RtpTransceiver to the
PeerConnection. If a MediaStreamTrack argument is provided, then the PeerConnection. If a MediaStreamTrack argument is provided, then the
transceiver will be configured with that media type and the track transceiver will be configured with that media type and the track
will be attached to the transceiver. Otherwise, the application MUST will be attached to the transceiver. Otherwise, the application MUST
explicitly specify the type; this mode is useful for creating explicitly specify the type; this mode is useful for creating
skipping to change at page 24, line 11 skipping to change at page 24, line 50
The createDataChannel method also includes a number of arguments The createDataChannel method also includes a number of arguments
which are used by the PeerConnection (e.g., maxPacketLifetime) but which are used by the PeerConnection (e.g., maxPacketLifetime) but
are not reflected in the SDP and do not affect the JSEP state. are not reflected in the SDP and do not affect the JSEP state.
4.1.6. createOffer 4.1.6. createOffer
The createOffer method generates a blob of SDP that contains a The createOffer method generates a blob of SDP that contains a
[RFC3264] offer with the supported configurations for the session, [RFC3264] offer with the supported configurations for the session,
including descriptions of the media added to this PeerConnection, the including descriptions of the media added to this PeerConnection, the
codec/RTP/RTCP options supported by this implementation, and any codec/RTP/RTCP options supported by this implementation, and any
candidates that have been gathered by the ICE Agent. An options candidates that have been gathered by the ICE agent. An options
parameter may be supplied to provide additional control over the parameter may be supplied to provide additional control over the
generated offer. This options parameter allows an application to generated offer. This options parameter allows an application to
trigger an ICE restart, for the purpose of reestablishing trigger an ICE restart, for the purpose of reestablishing
connectivity. connectivity.
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.
skipping to change at page 24, line 46 skipping to change at page 25, line 36
exact handling of subsequent offer generation is detailed in exact handling of subsequent offer generation is detailed in
Section 5.2.2. below. Section 5.2.2. below.
Session descriptions generated by createOffer must be immediately Session descriptions generated by createOffer must be immediately
usable by setLocalDescription; if a system has limited resources usable by setLocalDescription; if a system has limited resources
(e.g. a finite number of decoders), createOffer should return an (e.g. a finite number of decoders), createOffer should return an
offer that reflects the current state of the system, so that offer that reflects the current state of the system, so that
setLocalDescription will succeed when it attempts to acquire those setLocalDescription will succeed when it attempts to acquire those
resources. resources.
Calling this method may do things such as generate new ICE Calling this method may do things such as generating new ICE
credentials, but does not result in candidate gathering, or cause credentials, but does not result in candidate gathering, or cause
media to start or stop flowing. media to start or stop flowing. Specifically, the offer is not
applied, and does not become the pending local description, until
setLocalDescription is called.
4.1.7. createAnswer 4.1.7. createAnswer
The createAnswer method generates a blob of SDP that contains a The createAnswer method generates a blob of SDP that contains a
[RFC3264] SDP answer with the supported configuration for the session [RFC3264] SDP answer with the supported configuration for the session
that is compatible with the parameters supplied in the most recent that is compatible with the parameters supplied in the most recent
call to setRemoteDescription, which MUST have been called prior to call to setRemoteDescription, which MUST have been called prior to
calling createAnswer. Like createOffer, the returned blob contains calling createAnswer. Like createOffer, the returned blob contains
descriptions of the media added to this PeerConnection, the descriptions of the media added to this PeerConnection, the
codec/RTP/RTCP options negotiated for this session, and any codec/RTP/RTCP options negotiated for this session, and any
candidates that have been gathered by the ICE Agent. An options candidates that have been gathered by the ICE agent. An options
parameter may be supplied to provide additional control over the parameter may be supplied to provide additional control over the
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
description should reflect the current state of the system. description should reflect the current state of the system.
Calling this method may do things such as generate new ICE Calling this method may do things such as generating new ICE
credentials, but does not trigger candidate gathering or change media credentials, but does not trigger candidate gathering or cause a
state. media state change. Specifically, the answer is not applied, and
does not become the pending local description, until
setLocalDescription is called.
4.1.8. SessionDescriptionType 4.1.8. SessionDescriptionType
Session description objects (RTCSessionDescription) may be of type Session description objects (RTCSessionDescription) may be of type
"offer", "pranswer", "answer" or "rollback". These types provide "offer", "pranswer", "answer" or "rollback". These types provide
information as to how the description parameter should be parsed, and information as to how the description parameter should be parsed, and
how the media state should be changed. how the media state should be changed.
"offer" indicates that a description should be parsed as an offer; "offer" indicates that a description should be parsed as an offer;
said description may include many possible media configurations. A said description may include many possible media configurations. A
skipping to change at page 26, line 29 skipping to change at page 27, line 20
answers as provisional answers, and only apply an answer as final answers as provisional answers, and only apply an answer as final
when it receives one that meets its criteria (e.g. a live user when it receives one that meets its criteria (e.g. a live user
instead of voicemail). instead of voicemail).
"rollback" is a special session description type implying that the "rollback" is a special session description type implying that the
state machine should be rolled back to the previous stable state, as state machine should be rolled back to the previous stable state, as
described in Section 4.1.8.2. The contents MUST be empty. described in Section 4.1.8.2. The contents MUST be empty.
4.1.8.1. Use of Provisional Answers 4.1.8.1. Use of Provisional Answers
Most web applications will not need to create answers using the Most applications will not need to create answers using the
"pranswer" type. While it is good practice to send an immediate "pranswer" type. While it is good practice to send an immediate
response to an "offer", in order to warm up the session transport and response to an offer, in order to warm up the session transport and
prevent media clipping, the preferred handling for a web application prevent media clipping, the preferred handling for a JSEP application
would be to create and send an "inactive" final answer immediately is to create and send a "sendonly" final answer with a null
after receiving the offer. Later, when the called user actually MediaStreamTrack immediately after receiving the offer, which will
accepts the call, the application can create a new "sendrecv" offer prevent media from being sent by the caller, and allow media to be
to update the previous offer/answer pair and start the media flow. sent immediately upon answer by the callee. Later, when the callee
While this could also be done with an inactive "pranswer", followed actually accepts the call, the application can plug in the real
by a sendrecv "answer", the initial "pranswer" leaves the offer- MediaStreamTrack and create a new "sendrecv" offer to update the
answer exchange open, which means that neither side can send an previous offer/answer pair and start bidirectional media flow. While
updated offer during this time. this could also be done with a "sendonly" pranswer, followed by a
"sendrecv" answer, the initial pranswer leaves the offer-answer
exchange open, which means that the caller cannot send an updated
offer during this time.
As an example, consider a typical web application that will set up a As an example, consider a typical JSEP application that wants to set
data channel, an audio channel, and a video channel. When an up audio and video as quickly as possible. When the callee receives
endpoint receives an offer with these channels, it could send an an offer with audio and video MediaStreamTracks, it will send an
answer accepting the data channel for two-way data, and accepting the immediate answer accepting these tracks as sendonly (meaning that the
audio and video tracks as inactive or receive-only. It could then caller will not send the callee any media yet, and because the callee
ask the user to accept the call, acquire the local media streams, and has not yet added its own MediaStreamTracks, the callee will not send
send a new offer to the remote side moving the audio and video to be any media either). It will then ask the user to accept the call and
two-way media. By the time the human has accepted the call and acquire the needed local tracks. Upon acceptance by the user, the
triggered the new offer, it is likely that the ICE and DTLS application will plug in the tracks it has acquired, which, because
handshaking for all the channels will already have finished. ICE and DTLS handshaking have likely completed by this point, can
start transmitting immediately. The application will also send a new
offer to the remote side indicating call acceptance and moving the
audio and video to be two-way media. A detailed example flow along
these lines is shown in Section 7.3.
Of course, some applications may not be able to perform this double Of course, some applications may not be able to perform this double
offer-answer exchange, particularly ones that are attempting to offer-answer exchange, particularly ones that are attempting to
gateway to legacy signaling protocols. In these cases, "pranswer" gateway to legacy signaling protocols. In these cases, pranswer can
can still provide the application with a mechanism to warm up the still provide the application with a mechanism to warm up the
transport. transport.
4.1.8.2. Rollback 4.1.8.2. Rollback
In certain situations it may be desirable to "undo" a change made to In certain situations it may be desirable to "undo" a change made to
setLocalDescription or setRemoteDescription. Consider a case where a setLocalDescription or setRemoteDescription. Consider a case where a
call is ongoing, and one side wants to change some of the session call is ongoing, and one side wants to change some of the session
parameters; that side generates an updated offer and then calls parameters; that side generates an updated offer and then calls
setLocalDescription. However, the remote side, either before or setLocalDescription. However, the remote side, either before or
after setRemoteDescription, decides it does not want to accept the after setRemoteDescription, decides it does not want to accept the
skipping to change at page 29, line 18 skipping to change at page 30, line 18
If setLocalDescription was previously called with an offer, and If setLocalDescription was previously called with an offer, and
setRemoteDescription is called with an answer (provisional or final), setRemoteDescription is called with an answer (provisional or final),
and the media directions are compatible, and media is available to and the media directions are compatible, and media is available to
send, this will result in the starting of media transmission. send, this will result in the starting of media transmission.
4.1.11. currentLocalDescription 4.1.11. currentLocalDescription
The currentLocalDescription method returns the current negotiated The currentLocalDescription method returns the current negotiated
local description - i.e., the local description from the last local description - i.e., the local description from the last
successful offer/answer exchange - in addition to any local successful offer/answer exchange - in addition to any local
candidates that have been generated by the ICE Agent since the local candidates that have been generated by the ICE agent since the local
description was set. description was set.
A null object will be returned if an offer/answer exchange has not A null object will be returned if an offer/answer exchange has not
yet been completed. yet been completed.
4.1.12. pendingLocalDescription 4.1.12. pendingLocalDescription
The pendingLocalDescription method returns a copy of the local The pendingLocalDescription method returns a copy of the local
description currently in negotiation - i.e., a local offer set description currently in negotiation - i.e., a local offer set
without any corresponding remote answer - in addition to any local without any corresponding remote answer - in addition to any local
candidates that have been generated by the ICE Agent since the local candidates that have been generated by the ICE agent since the local
description was set. description was set.
A null object will be returned if the state of the PeerConnection is A null object will be returned if the state of the PeerConnection is
"stable" or "have-remote-offer". "stable" or "have-remote-offer".
4.1.13. currentRemoteDescription 4.1.13. currentRemoteDescription
The currentRemoteDescription method returns a copy of the current The currentRemoteDescription method returns a copy of the current
negotiated remote description - i.e., the remote description from the negotiated remote description - i.e., the remote description from the
last successful offer/answer exchange - in addition to any remote last successful offer/answer exchange - in addition to any remote
skipping to change at page 31, line 29 skipping to change at page 32, line 29
if decreased, the now-superfluous candidates are discarded. if decreased, the now-superfluous candidates are discarded.
o The bundle and RTCP-multiplexing policies MUST NOT be changed o The bundle and RTCP-multiplexing policies MUST NOT be changed
after the construction of the PeerConnection. after the construction of the PeerConnection.
This call may result in a change to the state of the ICE Agent. This call may result in a change to the state of the ICE Agent.
4.1.17. addIceCandidate 4.1.17. addIceCandidate
The addIceCandidate method provides a remote candidate to the ICE The addIceCandidate method provides a remote candidate to the ICE
Agent, which, if parsed successfully, will be added to the current agent, which, if parsed successfully, will be added to the current
and/or pending remote description according to the rules defined for and/or pending remote description according to the rules defined for
Trickle ICE. The pair of MID and ufrag is used to determine the m= Trickle ICE. The pair of MID and ufrag is used to determine the m=
section and ICE candidate generation to which the candidate belongs. section and ICE candidate generation to which the candidate belongs.
If the MID is not present, the m= section index is used to look up If the MID is not present, the m= section index is used to look up
the locally generated MID (see Section 5.9), which is used in place the locally generated MID (see Section 5.9), which is used in place
of a supplied MID. If these values or the candidate string are of a supplied MID. If these values or the candidate string are
invalid, an error is generated. invalid, an error is generated.
The purpose of the ufrag is to resolve ambiguities when trickle ICE The purpose of the ufrag is to resolve ambiguities when trickle ICE
is in progress during an ICE restart. If the ufrag is absent, the is in progress during an ICE restart. If the ufrag is absent, the
candidate MUST be assumed to belong to the most recently applied candidate MUST be assumed to belong to the most recently applied
remote description. Connectivity checks will be sent to the new remote description. Connectivity checks will be sent to the new
candidate. candidate.
This method can also be used to provide an end-of-candidates This method can also be used to provide an end-of-candidates
indication to the ICE Agent, as defined in [I-D.ietf-ice-trickle]). indication to the ICE agent, as defined in [I-D.ietf-ice-trickle]).
The MID and ufrag are used as described above to determine the m= The MID and ufrag are used as described above to determine the m=
section and ICE generation for which candidate gathering is complete. section and ICE generation for which candidate gathering is complete.
If the ufrag is not present, then the end-of-candidates indication If the ufrag is not present, then the end-of-candidates indication
MUST be assumed to apply to the relevant m= section in the most MUST be assumed to apply to the relevant m= section in the most
recently applied remote description. If neither the MID nor the m= recently applied remote description. If neither the MID nor the m=
index is present, then the indication MUST be assumed to apply to all index is present, then the indication MUST be assumed to apply to all
m= sections in the most recently applied remote description. m= sections in the most recently applied 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
skipping to change at page 33, line 42 skipping to change at page 34, line 42
excluded by subsequent calls to createOffer and createAnswer, in excluded by subsequent calls to createOffer and createAnswer, in
which case the corresponding media formats in the associated m= which case the corresponding media formats in the associated m=
section will be excluded. The codec preferences cannot add media section will be excluded. The codec preferences cannot add media
formats that would otherwise not be present. This includes codecs formats that would otherwise not be present. This includes codecs
that were not negotiated in a previous offer/answer exchange that that were not negotiated in a previous offer/answer exchange that
included the transceiver. included the transceiver.
The codec preferences of an RtpTransceiver can also determine the The codec preferences of an RtpTransceiver can also determine the
order of codecs in subsequent calls to createOffer and createAnswer, order of codecs in subsequent calls to createOffer and createAnswer,
in which case the order of the media formats in the associated m= in which case the order of the media formats in the associated m=
section will match. However, the codec preferences cannot change the section will follow the specified preferences.
order of the media formats after an answer containing the transceiver
has been applied. At this point, codecs can only be removed, not
reordered.
5. SDP Interaction Procedures 5. SDP Interaction Procedures
This section describes the specific procedures to be followed when This section describes the specific procedures to be followed when
creating and parsing SDP objects. creating and parsing SDP objects.
5.1. Requirements Overview 5.1. Requirements Overview
JSEP implementations must comply with the specifications listed below JSEP implementations must comply with the specifications listed below
that govern the creation and processing of offers and answers. that govern the creation and processing of offers and answers.
The first set of specifications is the "mandatory-to-implement" set. 5.1.1. Usage Requirements
All implementations must support these behaviors, but may not use all
of them if the remote side, which may not be a JSEP endpoint, does
not support them.
The second set of specifications is the "mandatory-to-use" set. The
local JSEP endpoint and any remote endpoint must indicate support for
these specifications in their session descriptions.
5.1.1. Implementation Requirements
Implementations of JSEP MUST conform to [I-D.ietf-rtcweb-rtp-usage].
This list of mandatory-to-implement specifications is derived from
the requirements outlined in that document and from
[I-D.ietf-rtcweb-security-arch].
R-1 [RFC4566] is the base SDP specification and MUST be
implemented.
R-2 [RFC5764] MUST be supported for signaling the UDP/TLS/RTP/SAVPF
[RFC5764], TCP/DTLS/RTP/SAVPF [RFC7850], "UDP/DTLS/SCTP"
[I-D.ietf-mmusic-sctp-sdp], and "TCP/DTLS/SCTP"
[I-D.ietf-mmusic-sctp-sdp] RTP profiles.
R-3 [RFC5245] MUST be implemented for signaling the ICE credentials
and candidate lines corresponding to each media stream. The
ICE implementation MUST be a Full implementation, not a Lite
implementation.
R-4 [RFC5763] MUST be implemented to signal DTLS certificate
fingerprints.
R-5 [RFC5888] MUST be implemented for signaling grouping
information, and MUST be used to identify m= lines via the
a=mid attribute.
R-6 [I-D.ietf-mmusic-msid] MUST be supported, in order to signal
associations between RTP objects and W3C MediaStreams and
MediaStreamTracks in a standard way.
R-7 The bundle mechanism in
[I-D.ietf-mmusic-sdp-bundle-negotiation] MUST be supported to
signal the ability to multiplex RTP streams on a single UDP
port, in order to avoid excessive use of port number resources.
R-8 The SDP attributes of "sendonly", "recvonly", "inactive", and
"sendrecv" from [RFC4566] MUST be implemented to signal
information about media direction.
R-9 [RFC5576] MUST be implemented to signal RTP SSRC values and
grouping semantics.
R-10 [RFC4585] MUST be implemented to signal RTCP based feedback.
R-11 [RFC5761] MUST be implemented to signal multiplexing of RTP and
RTCP.
R-12 [RFC5506] MUST be implemented to signal reduced-size RTCP
messages.
R-13 [RFC4588] MUST be implemented to signal RTX payload type
associations.
R-14 [RFC3556] MUST be supported for control of RTCP bandwidth
limits.
The SDES SRTP keying mechanism from [RFC4568] MUST NOT be
implemented, as discussed in [I-D.ietf-rtcweb-security-arch].
As required by [RFC4566], Section 5.13, JSEP implementations MUST
ignore unknown attribute (a=) lines.
5.1.2. Usage Requirements All session descriptions handled by JSEP implementations, both local
and remote, MUST indicate support for the following specifications.
If any of these are absent, this omission MUST be treated as an
error.
All session descriptions handled by JSEP endpoints, both local and o ICE, as specified in [RFC5245], MUST be used. Note that the
remote, MUST indicate support for the following specifications. If remote endpoint may use a Lite implementation; implementations
any of these are absent, this omission MUST be treated as an error. MUST properly handle remote endpoints which do ICE-Lite.
U-1 ICE, as specified in [RFC5245], MUST be used. Note that the o DTLS [RFC6347] or DTLS-SRTP [RFC5763], MUST be used, as
remote endpoint may use a Lite implementation; implementations appropriate for the media type, as specified in
MUST properly handle remote endpoints which do ICE-Lite. [I-D.ietf-rtcweb-security-arch]
U-2 DTLS [RFC6347] or DTLS-SRTP [RFC5763], MUST be used, as The SDES SRTP keying mechanism from [RFC4568] MUST NOT be used, as
appropriate for the media type, as specified in discussed in [I-D.ietf-rtcweb-security-arch].
[I-D.ietf-rtcweb-security-arch]
5.1.3. Profile Names and Interoperability 5.1.2. Profile Names and Interoperability
For media m= sections, JSEP endpoints MUST support both the "UDP/TLS/ For media m= sections, JSEP implementations MUST support the
RTP/SAVPF" and "TCP/DTLS/RTP/SAVPF" profiles and MUST indicate one of "UDP/TLS/RTP/SAVPF" profile specified in [RFC7850], and MUST indicate
these two profiles for each media m= line they produce in an offer. this profile for each media m= line they produce in an offer. For
For data m= sections, JSEP endpoints must support both the "UDP/DTLS/ data m= sections, implementations MUST support the "UDP/DTLS/SCTP"
SCTP" and "TCP/DTLS/SCTP" profiles and MUST indicate one of these two profile and MUST indicate this profile for each data m= line they
profiles for each data m= line they produce in an offer. Because ICE produce in an offer. Because ICE can select either UDP [RFC5245] or
can select either TCP or UDP transport depending on network TCP [RFC6544] transport depending on network conditions, this
conditions, both advertisements are consistent with ICE eventually advertisement is consistent with ICE eventually selecting either
selecting either either UDP or TCP. either UDP or TCP.
Unfortunately, in an attempt at compatibility, some endpoints Unfortunately, in an attempt at compatibility, some endpoints
generate other profile strings even when they mean to support one of generate other profile strings even when they mean to support one of
these profiles. For instance, an endpoint might generate "RTP/AVP" these profiles. For instance, an endpoint might generate "RTP/AVP"
but supply "a=fingerprint" and "a=rtcp-fb" attributes, indicating its but supply "a=fingerprint" and "a=rtcp-fb" attributes, indicating its
willingness to support "(UDP,TCP)/TLS/RTP/SAVPF". In order to willingness to support "(UDP,TCP)/TLS/RTP/SAVPF". In order to
simplify compatibility with such endpoints, JSEP endpoints MUST simplify compatibility with such endpoints, JSEP implementations MUST
follow the following rules when processing the media m= sections in follow the following rules when processing the media m= sections in
an offer: an offer:
o The profile in any "m=" line in any answer MUST exactly match the o The profile in any "m=" line in any answer MUST exactly match the
profile provided in the offer. profile provided in the offer.
o Any profile matching the following patterns MUST be accepted: o Any profile matching the following patterns MUST be accepted:
"RTP/[S]AVP[F]" and "(UDP/TCP)/TLS/RTP/SAVP[F]" "RTP/[S]AVP[F]" and "(UDP/TCP)/TLS/RTP/SAVP[F]"
o Because DTLS-SRTP is REQUIRED, the choice of SAVP or AVP has no o Because DTLS-SRTP is REQUIRED, the choice of SAVP or AVP has no
effect; support for DTLS-SRTP is determined by the presence of one effect; support for DTLS-SRTP is determined by the presence of one
or more "a=fingerprint" attribute. Note that lack of an or more "a=fingerprint" attribute. Note that lack of an
"a=fingerprint" attribute will lead to negotiation failure. "a=fingerprint" attribute will lead to negotiation failure.
o The use of AVPF or AVP simply controls the timing rules used for o The use of AVPF or AVP simply controls the timing rules used for
RTCP feedback. If AVPF is provided, or an "a=rtcp-fb" attribute RTCP feedback. If AVPF is provided, or an "a=rtcp-fb" attribute
is present, assume AVPF timing, i.e., a default value of "trr- is present, assume AVPF timing, i.e., a default value of "trr-
int=0". Otherwise, assume that AVPF is being used in an AVP int=0". Otherwise, assume that AVPF is being used in an AVP
compatible mode and use a value of "trr-int=4000". compatible mode and use a value of "trr-int=4000".
o For data m= sections, JSEP endpoints MUST support receiving the o For data m= sections, implementations MUST support receiving the
"UDP/DTLS/SCTP", "TCP/DTLS/SCTP", or "DTLS/SCTP" (for backwards "UDP/DTLS/SCTP", "TCP/DTLS/SCTP", or "DTLS/SCTP" (for backwards
compatibility) profiles. compatibility) profiles.
Note that re-offers by JSEP endpoints MUST use the correct profile Note that re-offers by JSEP implementations MUST use the correct
strings even if the initial offer/answer exchange used an (incorrect) profile strings even if the initial offer/answer exchange used an
older profile string. (incorrect) older profile string.
5.2. Constructing an Offer 5.2. Constructing an Offer
When createOffer is called, a new SDP description must be created When createOffer is called, a new SDP description must be created
that includes the functionality specified in that includes the functionality specified in
[I-D.ietf-rtcweb-rtp-usage]. The exact details of this process are [I-D.ietf-rtcweb-rtp-usage]. The exact details of this process are
explained below. explained below.
5.2.1. Initial Offers 5.2.1. Initial Offers
skipping to change at page 38, line 8 skipping to change at page 37, line 30
o Encryption Keys ("k=") lines do not provide sufficient security o Encryption Keys ("k=") lines do not provide sufficient security
and MUST NOT be included. and MUST NOT be included.
o A "t=" line MUST be added, as specified in [RFC4566], Section 5.9; o A "t=" line MUST be added, as specified in [RFC4566], Section 5.9;
both <start-time> and <stop-time> SHOULD be set to zero, e.g. "t=0 both <start-time> and <stop-time> SHOULD be set to zero, e.g. "t=0
0". 0".
o An "a=ice-options" line with the "trickle" option MUST be added, o An "a=ice-options" line with the "trickle" option MUST be added,
as specified in [I-D.ietf-ice-trickle], Section 4. as specified in [I-D.ietf-ice-trickle], Section 4.
o If WebRTC identity is being used, an "a=identity" line as
described in [I-D.ietf-rtcweb-security-arch], Section 5.
The next step is to generate m= sections, as specified in [RFC4566] The next step is to generate m= sections, as specified in [RFC4566]
Section 5.14. An m= section is generated for each RtpTransceiver Section 5.14. An m= section is generated for each RtpTransceiver
that has been added to the PeerConnection, excluding any stopped that has been added to the PeerConnection, excluding any stopped
RtpTransceivers. This is done in the order the RtpTransceivers were RtpTransceivers. This is done in the order the RtpTransceivers were
added to the PeerConnection. added to the PeerConnection.
For each m= section generated for an RtpTransceiver, establish a For each m= section generated for an RtpTransceiver, establish a
mapping between the transceiver and the index of the generated m= mapping between the transceiver and the index of the generated m=
section. section.
skipping to change at page 39, line 36 skipping to change at page 39, line 14
occurs when the description is applied. The generated MID value occurs when the description is applied. The generated MID value
can be considered a "proposed" MID at this point. can be considered a "proposed" MID at this point.
o A direction attribute which is the same as that of the associated o A direction attribute which is the same as that of the associated
transceiver. transceiver.
o For each media format on the m= line, "a=rtpmap" and "a=fmtp" o For each media format on the m= line, "a=rtpmap" and "a=fmtp"
lines, as specified in [RFC4566], Section 6, and [RFC3264], lines, as specified in [RFC4566], Section 6, and [RFC3264],
Section 5.1. Section 5.1.
o If this m= section is for media with configurable durations of
media per packet, e.g., audio, an "a=maxptime" line, indicating
the maximum amount of media, specified in milliseconds, that can
be encapsulated in each packet, as specified in [RFC4566],
Section 6. This value is set to the smallest of the maximum
duration values across all the codecs included in the m= section.
o If this m= section is for video media, and there are known
limitations on the size of images which can be decoded, an
"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],
Section 6, and specific usage for each media type is outlined in Section 6, and specific usage for each media type is outlined in
Sections 4 and 5. Sections 4 and 5.
o If this m= section is for media with configurable durations of
media per packet, e.g., audio, an "a=maxptime" line, indicating
the maximum amount of media, specified in milliseconds, that can
be encapsulated in each packet, as specified in [RFC4566],
Section 6. This value is set to the smallest of the maximum
duration values across all the codecs included in the m= section.
o If this m= section is for video media, and there are known
limitations on the size of images which can be decoded, an
"a=imageattr" line, as specified in Section 3.6.
o For each supported RTP header extension, an "a=extmap" line, as o For each supported RTP header extension, an "a=extmap" line, as
specified in [RFC5285], Section 5. The list of header extensions specified in [RFC5285], Section 5. The list of header extensions
that SHOULD/MUST be supported is specified in that SHOULD/MUST be supported is specified in
[I-D.ietf-rtcweb-rtp-usage], Section 5.2. Any header extensions [I-D.ietf-rtcweb-rtp-usage], Section 5.2. Any header extensions
that require encryption MUST be specified as indicated in that require encryption MUST be specified as indicated in
[RFC6904], Section 4. [RFC6904], Section 4.
o For each supported RTCP feedback mechanism, an "a=rtcp-fb" o For each supported RTCP feedback mechanism, an "a=rtcp-fb"
mechanism, as specified in [RFC4585], Section 4.2. The list of mechanism, as specified in [RFC4585], Section 4.2. The list of
RTCP feedback mechanisms that SHOULD/MUST be supported is RTCP feedback mechanisms that SHOULD/MUST be supported is
specified in [I-D.ietf-rtcweb-rtp-usage], Section 5.1. specified in [I-D.ietf-rtcweb-rtp-usage], Section 5.1.
o If the bundle policy for this PeerConnection is set to "max-
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
for this media type, an "a=bundle-only" line.
o If the RtpTransceiver has a sendrecv or sendonly direction: o If the RtpTransceiver has a sendrecv or sendonly direction:
* An "a=msid" line, as specified in [I-D.ietf-mmusic-msid], * For each MediaStream that was associated with the transceiver
Section 2. when it was created via addTrack or addTransceiver, an "a=msid"
line, as specified in [I-D.ietf-mmusic-msid], Section 2. If a
MediaStreamTrack is attached to the transceiver's RtpSender,
the "a=msid" lines MUST use that track's ID. If no
MediaStreamTrack is attached, a valid ID MUST be generated, in
the same way that the implementation generates IDs for local
tracks.
* If no MediaStream is associated with the transceiver, a single
"a=msid" line with the special value "-" in place of the
MediaStream ID, as specified in [I-D.ietf-mmusic-msid],
Section 3. The track ID MUST be selected as described above.
o If the RtpTransceiver has a sendrecv or sendonly direction, and o If the RtpTransceiver has a sendrecv or sendonly direction, and
the application has specified RID values or has specified more the application has specified RID values or has specified more
than one encoding in the RtpSenders's parameters, an "a=rid" line than one encoding in the RtpSenders's parameters, an "a=rid" line
for each encoding specified. The "a=rid" line is specified in for each encoding specified. The "a=rid" line is specified in
[I-D.ietf-mmusic-rid], and its direction MUST be "send". If the [I-D.ietf-mmusic-rid], and its direction MUST be "send". If the
application has chosen a RID value, it MUST be used as the rid- application has chosen a RID value, it MUST be used as the rid-
identifier; otherwise a RID value MUST be generated by the identifier; otherwise a RID value MUST be generated by the
implementation. RID values MUST be generated in a fashion that implementation. RID values MUST be generated in a fashion that
does not leak user information, e.g., randomly or using a per- does not leak user information, e.g., randomly or using a per-
skipping to change at page 41, line 7 skipping to change at page 40, line 41
specified, or only one encoding is specified but without a RID specified, or only one encoding is specified but without a RID
value, then no "a=rid" lines are generated. value, then no "a=rid" lines are generated.
o If the RtpTransceiver has a sendrecv or sendonly direction and o If the RtpTransceiver has a sendrecv or sendonly direction and
more than one "a=rid" line has been generated, an "a=simulcast" more than one "a=rid" line has been generated, an "a=simulcast"
line, with direction "send", as defined in line, with direction "send", as defined in
[I-D.ietf-mmusic-sdp-simulcast], Section 6.2. The list of RIDs [I-D.ietf-mmusic-sdp-simulcast], Section 6.2. The list of RIDs
MUST include all of the RID identifiers used in the "a=rid" lines MUST include all of the RID identifiers used in the "a=rid" lines
for this m= section. for this m= section.
o If the bundle policy for this PeerConnection is set to "max-
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
for this media type, an "a=bundle-only" line.
The following attributes, which are of category IDENTICAL or The following attributes, which are of category IDENTICAL or
TRANSPORT, MUST appear only in "m=" sections which either have a TRANSPORT, MUST appear only in "m=" sections which either have a
unique address or which are associated with the bundle-tag. (In unique address or which are associated with the bundle-tag. (In
initial offers, this means those "m=" sections which do not contain initial offers, this means those "m=" sections which do not contain
an "a=bundle-only" attribute. an "a=bundle-only" attribute.)
o "a=ice-ufrag" and "a=ice-pwd" lines, as specified in [RFC5245], o "a=ice-ufrag" and "a=ice-pwd" lines, as specified in [RFC5245],
Section 15.4. Section 15.4.
o An "a=fingerprint" line for each of the endpoint's certificates, o An "a=fingerprint" line for each of the endpoint's certificates,
as specified in [RFC4572], Section 5; the digest algorithm used as specified in [RFC4572], Section 5; the digest algorithm used
for the fingerprint MUST match that used in the certificate for the fingerprint MUST match that used in the certificate
signature. signature.
o An "a=setup" line, as specified in [RFC4145], Section 4, and o An "a=setup" line, as specified in [RFC4145], Section 4, and
clarified for use in DTLS-SRTP scenarios in [RFC5763], Section 5. clarified for use in DTLS-SRTP scenarios in [RFC5763], Section 5.
skipping to change at page 41, line 34 skipping to change at page 41, line 25
o An "a=dtls-id" line, as specified in [I-D.ietf-mmusic-dtls-sdp] o An "a=dtls-id" line, as specified in [I-D.ietf-mmusic-dtls-sdp]
Section 5.2. Section 5.2.
o An "a=rtcp" line, as specified in [RFC3605], Section 2.1, o An "a=rtcp" line, as specified in [RFC3605], Section 2.1,
containing the dummy value "9 IN IP4 0.0.0.0", because no containing the dummy value "9 IN IP4 0.0.0.0", because no
candidates have yet been gathered. candidates have yet been gathered.
o An "a=rtcp-mux" line, as specified in [RFC5761], Section 5.1.3. o An "a=rtcp-mux" line, as specified in [RFC5761], Section 5.1.3.
o If the RTP/RTCP multiplexing policy is "require", an "a=rtcp-mux-
only" line, as specified in [I-D.ietf-mmusic-mux-exclusive],
Section 4.
o An "a=rtcp-rsize" line, as specified in [RFC5506], Section 5. o An "a=rtcp-rsize" line, as specified in [RFC5506], Section 5.
Lastly, if a data channel has been created, a m= section MUST be Lastly, if a data channel has been created, a m= section MUST be
generated for data. The <media> field MUST be set to "application" generated for data. The <media> field MUST be set to "application"
and the <proto> field MUST be set to "UDP/DTLS/SCTP" and the <proto> field MUST be set to "UDP/DTLS/SCTP"
[I-D.ietf-mmusic-sctp-sdp]. The "fmt" value MUST be set to "webrtc- [I-D.ietf-mmusic-sctp-sdp]. The "fmt" value MUST be set to "webrtc-
datachannel" as specified in [I-D.ietf-mmusic-sctp-sdp], Section 4.1. datachannel" as specified in [I-D.ietf-mmusic-sctp-sdp], Section 4.1.
Within the data m= section, the "a=mid", "a=ice-ufrag", "a=ice-pwd", Within the data m= section, an "a=mid" line MUST be generated and
"a=fingerprint", "a=dtls-id", and "a=setup" lines MUST be included as included as described above, along with an "a=sctp-port" line
mentioned above, along with an "a=fmtp:webrtc-datachannel" line and referencing the SCTP port number, as defined in
an "a=sctp-port" line referencing the SCTP port number as defined in [I-D.ietf-mmusic-sctp-sdp], Section 5.1, and, if appropriate, an
[I-D.ietf-mmusic-sctp-sdp], Section 4.1. "a=max-message-size" line, as defined in [I-D.ietf-mmusic-sctp-sdp],
Section 6.1.
As discussed above, the following attributes of category IDENTICAL or
TRANSPORT are included only if the data m= section either has a
unique address or is associated with the bundle-tag (e.g., if it is
the only m= section):
o "a=ice-ufrag"
o "a=ice-pwd"
o "a=fingerprint"
o "a=setup"
o "a=dtls-id"
Once all m= sections have been generated, a session-level "a=group" Once all m= sections have been generated, a session-level "a=group"
attribute MUST be added as specified in [RFC5888]. This attribute attribute MUST be added as specified in [RFC5888]. This attribute
MUST have semantics "bundle", and MUST include the mid identifiers of MUST have semantics "BUNDLE", and MUST include the mid identifiers of
each m= section. The effect of this is that the browser offers all each m= section. The effect of this is that the JSEP implementation
m= sections as one bundle group. However, whether the m= sections offers all m= sections as one bundle group. However, whether the m=
are bundle-only or not depends on the bundle policy. sections are bundle-only or not depends on the bundle policy.
The next step is to generate session-level lip sync groups as defined The next step is to generate session-level lip sync groups as defined
in [RFC5888], Section 7. For each MediaStream referenced by more in [RFC5888], Section 7. For each MediaStream referenced by more
than one RtpTransceiver (by passing those MediaStreams as arguments than one RtpTransceiver (by passing those MediaStreams as arguments
to the addTrack and addTransceiver methods), a group of type "LS" to the addTrack and addTransceiver methods), a group of type "LS"
MUST be added that contains the mid values for each RtpTransceiver. MUST be added that contains the mid values for each RtpTransceiver.
Attributes which SDP permits to either be at the session level or the Attributes which SDP permits to either be at the session level or the
media level SHOULD generally be at the media level even if they are media level SHOULD generally be at the media level even if they are
identical. This promotes readability, especially if one of a set of identical. This promotes readability, especially if one of a set of
skipping to change at page 43, line 46 skipping to change at page 44, line 14
o If an RtpTransceiver is stopped and is not associated with an m= o If an RtpTransceiver is stopped and is not associated with an m=
section, an m= section MUST NOT be generated for it. This section, an m= section MUST NOT be generated for it. This
prevents adding back RtpTransceivers whose m= sections were prevents adding back RtpTransceivers whose m= sections were
recycled and used for a new RtpTransceiver in a previous offer/ recycled and used for a new RtpTransceiver in a previous offer/
answer exchange, as described above. answer exchange, as described above.
o If an RtpTransceiver has been stopped and is associated with an m= o If an RtpTransceiver has been stopped and is associated with an m=
section, and the m= section is not being recycled as described section, and the m= section is not being recycled as described
above, an m= section MUST be generated for it with the port set to above, an m= section MUST be generated for it with the port set to
zero and the "a=msid" line removed. zero and all "a=msid" lines removed.
o For RtpTransceivers that are not stopped, the "a=msid" line MUST o For RtpTransceivers that are not stopped, the "a=msid" line(s)
stay the same if they are present in the current description. MUST stay the same if they are present in the current description,
regardless of changes to the transceiver's direction or track. If
no "a=msid" line is present in the current description, "a=msid"
line(s) MUST be generated according to the same rules as for an
initial offer.
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. In each case, if no RTP candidates have when ICE has completed. In each case, if no RTP candidates have
yet been gathered, dummy values MUST be used, as described above. yet been gathered, dummy values MUST be used, as described 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 5.2.3.1 was specified. If the m= "IceRestart" option ( Section 5.2.3.1 was specified. If the m=
section is bundled into another m= section, it still MUST NOT section is bundled into another m= section, it still MUST NOT
contain any ICE credentials. contain any ICE credentials.
o If the m= section is not bundled into another m= section, an o If the m= section is not bundled into another m= section, its
"a=rtcp" attribute line MUST be added with of the default RTCP "a=rtcp" attribute line MUST be filled in with the port and
candidate, as indicated in [RFC5761], Section 5.1.3. address of the default RTCP candidate, as indicated in [RFC5761],
Section 5.1.3. If no RTCP candidates have yet been gathered,
dummy values MUST be used, as described in the initial offer
section above.
o If the m= section is not bundled into another m= section, for each o If the m= section is not bundled into another m= section, for each
candidate that has been gathered during the most recent gathering candidate that has been gathered during the most recent gathering
phase (see Section 3.5.1), an "a=candidate" line MUST be added, as phase (see Section 3.5.1), an "a=candidate" line MUST be added, as
defined in [RFC5245], Section 4.3., paragraph 3. If candidate defined in [RFC5245], Section 4.3., paragraph 3. If candidate
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 RtpTransceivers that are still present, the "a=msid" line MUST
stay the same.
o For RtpTransceivers that are still present, the "a=rid" lines MUST o For RtpTransceivers that are still present, the "a=rid" lines MUST
stay the same. stay the same.
o For RtpTransceivers that are still present, any "a=simulcast" line o For RtpTransceivers that are still present, any "a=simulcast" line
MUST stay the same. MUST stay the same.
o If any RtpTransceiver has been stopped, the port MUST be set to o If any RtpTransceiver has been stopped, the port MUST be set to
zero and the "a=msid" line MUST be removed. zero and all "a=msid" lines MUST be removed.
o If any RtpTransceiver has been added, and there exists a m= o If any RtpTransceiver has been added, and there exists a m=
section with a zero port in the current local description or the section with a zero port in the current local description or the
current remote description, that m= section MUST be recycled by current remote description, that m= section MUST be recycled by
generating a m= section for the added RtpTransceiver as if the m= generating a m= section for the added RtpTransceiver as if the m=
section were being added to session description, except that section were being added to session description, except that
instead of adding it, the generated m= section replaces the m= instead of adding it, the generated m= section replaces the m=
section with a zero port. The new m= section MUST contain a new section with a zero port. The new m= section MUST contain a new
MID. MID.
skipping to change at page 45, line 28 skipping to change at page 45, line 46
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, if of the corresponding m= section in the current remote description, if
any: any:
o The m= line and corresponding "a=rtpmap" and "a=fmtp" lines MUST o The m= line and corresponding "a=rtpmap" and "a=fmtp" lines MUST
only include codecs present in the most recent answer which have only include codecs present in the most recent answer which have
not been excluded by the codec preferences of the associated not been excluded by the codec preferences of the associated
transceiver. Note that non-JSEP endpoints are not subject to transceiver. Note that non-JSEP endpoints are not subject to
these restrictions, and might offer media formats that were not these restrictions, and might offer media formats that were not
present in the most recent answer, as specified in [RFC3264], present in the most recent answer, as specified in [RFC3264],
Section 8. Therefore, JSEP endpoints MUST be prepared to receive Section 8. Therefore, JSEP implementations MUST be prepared to
such offers. receive such offers.
o The media formats on the m= line MUST be generated in the same o Unless codec preferences have been set for the associated
order as in the current local description. transceiver, the media formats on the m= line MUST be generated in
the same order as in the current local description.
o The RTP header extensions MUST only include those that are present o The RTP header extensions MUST only include those that are present
in the most recent answer. in the most recent answer.
o The RTCP feedback extensions MUST only include those that are o The RTCP feedback extensions MUST only include those that are
present in the most recent answer. present in the most recent answer.
o The "a=rtcp" line MUST only be added if the most recent answer did o The "a=rtcp" line MUST only be added if the most recent answer did
not include an "a=rtcp-mux" line. not include an "a=rtcp-mux" line.
o The "a=rtcp-mux" line MUST only be added if present in the most o The "a=rtcp-mux" line MUST only be added if present in the most
recent answer. recent answer.
o The "a=rtcp-mux-only" line MUST only be added if present in the o The "a=rtcp-mux-only" line MUST NOT be added.
most recent answer.
o The "a=rtcp-rsize" line MUST only be added if present in the most o The "a=rtcp-rsize" line MUST only be added if present in the most
recent answer. recent answer.
The "a=group:BUNDLE" attribute MUST include the mid identifiers o An "a=bundle-only" line MUST NOT be added, as indicated in
[I-D.ietf-mmusic-sdp-bundle-negotiation], Section 6. Instead,
JSEP implementations MUST simply omit parameters in the IDENTICAL
and TRANSPORT categories for bundled m= sections, as described in
[I-D.ietf-mmusic-sdp-bundle-negotiation], Section 8.1.
o Note that if media m= sections are bundled into a data m= section,
then certain TRANSPORT and IDENTICAL attributes may appear in the
data m= section even if they would otherwise only be appropriate
for a media m= section (e.g., "a=rtcp-mux"). This cannot happen
in initial offers because in the initial offer JSEP
implementations always list media m= sections (if any) before the
data m= section (if any), and at least one of those media m=
sections will not have the "a=bundle-only" attribute. Therefore,
in initial offers, any "a=bundle-only" m= sections will be bundled
into a preceding non-bundle-only media m= section.
The "a=group:BUNDLE" attribute MUST include the MID identifiers
specified in the bundle group in the most recent answer, minus any m= specified in the bundle group in the most recent answer, minus any m=
sections that have been marked as rejected, plus any newly added or sections that have been marked as rejected, plus any newly added or
re-enabled m= sections. In other words, the bundle attribute must re-enabled m= sections. In other words, the bundle attribute must
contain all m= sections that were previously bundled, as long as they contain all m= sections that were previously bundled, as long as they
are still alive, as well as any new m= sections. are still alive, as well as any new m= sections.
The "LS" groups are generated in the same way as with initial offers. "a=group:LS" attributes are generated in the same way as for initial
offers, with the additional stipulation that any lip sync groups that
were present in the most recent answer MUST continue to exist and
MUST contain any previously existing MID identifiers, as long as the
identified m= sections still exist and are not rejected, and the
group still contains at least two MID identifiers. This ensures that
any synchronized "recvonly" m= sections continue to be synchronized
in the new offer.
5.2.3. Options Handling 5.2.3. Options Handling
The createOffer method takes as a parameter an RTCOfferOptions The createOffer method takes as a parameter an RTCOfferOptions
object. Special processing is performed when generating a SDP object. Special processing is performed when generating a SDP
description if the following options are present. description if the following options are present.
5.2.3.1. IceRestart 5.2.3.1. IceRestart
If the "IceRestart" option is specified, with a value of "true", the If the "IceRestart" option is specified, with a value of "true", the
skipping to change at page 46, line 42 skipping to change at page 47, line 35
5.2.3.2. 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 [RFC6716], the
parameter would be specified in the offer. This option allows the "usedtx=1" parameter, specified in [RFC7587], would be used in the
endpoint to significantly reduce the amount of audio bandwidth it offer. This option allows the endpoint to significantly reduce the
receives, at the cost of some fidelity, depending on the quality of amount of audio bandwidth it receives, at the cost of some fidelity,
the remote VAD algorithm. depending on the quality of the remote VAD algorithm.
If the "VoiceActivityDetection" option is specified, with a value of If the "VoiceActivityDetection" option is specified, with a value of
"false", the browser MUST NOT emit "CN" codecs. For codecs that have "false", the JSEP implementation MUST NOT emit "CN" codecs. For
their own internal silence suppression support, the appropriate fmtp codecs that have their own internal silence suppression support, the
parameters for that codec MUST be specified to indicate that silence appropriate fmtp parameters for that codec MUST be specified to
suppression for received audio is not desired. For example, when indicate that silence suppression for received audio is not desired.
using the Opus codec, the "usedtx=0" parameter would be specified in For example, when using the Opus codec, the "usedtx=0" parameter
the offer. would be specified in the offer.
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.
skipping to change at page 47, line 40 skipping to change at page 48, line 35
Note that the remote description SDP may not have been created by a Note that the remote description SDP may not have been created by a
JSEP endpoint and may not conform to all the requirements listed in JSEP endpoint and may not conform to all the requirements listed in
Section 5.2. For many cases, this is not a problem. However, if any Section 5.2. For many cases, this is not a problem. However, if any
mandatory SDP attributes are missing, or functionality listed as mandatory SDP attributes are missing, or functionality listed as
mandatory-to-use above is not present, this MUST be treated as an mandatory-to-use above is not present, this MUST be treated as an
error, and MUST cause the affected m= sections to be marked as error, and MUST cause the affected m= sections to be marked as
rejected. rejected.
The first step in generating an initial answer is to generate The first step in generating an initial answer is to generate
session-level attributes. The process here is identical to that session-level attributes. The process here is identical to that
indicated in the Initial Offers section above, except that the indicated in 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 session-level lip sync groups as defined The next step is to generate session-level lip sync groups, as
in [RFC5888], Section 7. For each group of type "LS" present in the defined in [RFC5888], Section 7. For each group of type "LS" present
offer, determine which of the local RtpTransceivers identified by in the offer, select the local RtpTransceivers that are referenced by
that group's mid values reference a common local MediaStream (as the MID values in the specified group, and determine which of them
specified in the addTrack and addTransceiver methods). If at least either reference a common local MediaStream (specified in the calls
two such RtpTransceivers exist, a group of type "LS" with the mid to addTrack/addTransceiver used to create them), or have no
values of these RtpTransceivers MUST be added. Otherwise, this MediaStream to reference because they were not created by addTrack/
indicates a difference of opinion between the offerer and answerer addTransceiver. If at least two such RtpTransceivers exist, a group
regarding lip sync status, and as such, the offered group MUST be of type "LS" with the mid values of these RtpTransceivers MUST be
ignored and no corresponding "LS" group generated. added. Otherwise the offered "LS" group MUST be ignored and no
corresponding group generated in the answer.
As a simple example, consider the following offer of a single audio
and single video track contained in the same MediaStream. SDP lines
not relevant to this example have been removed for clarity. As
explained in Section 5.2, a group of type "LS" has been added that
references each track's RtpTransceiver.
a=group:LS a1 v1
m=audio 10000 UDP/TLS/RTP/SAVPF 0
a=mid:a1
a=msid:ms1 mst1a
m=video 10001 UDP/TLS/RTP/SAVPF 96
a=mid:v1
a=msid:ms1 mst1v
If the answerer uses a single MediaStream when it adds its tracks,
both of its transceivers will reference this stream, and so the
subsequent answer will contain a "LS" group identical to that in the
offer, as shown below:
a=group:LS a1 v1
m=audio 20000 UDP/TLS/RTP/SAVPF 0
a=mid:a1
a=msid:ms2 mst2a
m=video 20001 UDP/TLS/RTP/SAVPF 96
a=mid:v1
a=msid:ms2 mst2v
However, if the answerer groups its tracks into separate
MediaStreams, its transceivers will reference different streams, and
so the subsequent answer will not contain a "LS" group.
m=audio 20000 UDP/TLS/RTP/SAVPF 0
a=mid:a1
a=msid:ms2a mst2a
m=video 20001 UDP/TLS/RTP/SAVPF 96
a=mid:v1
a=msid:ms2b mst2v
Finally, if the answerer does not add any tracks, its transceivers
will not reference any MediaStreams, causing the preferences of the
offerer to be maintained, and so the subsequent answer will contain
an identical "LS" group.
a=group:LS a1 v1
m=audio 20000 UDP/TLS/RTP/SAVPF 0
a=mid:a1
a=recvonly
m=video 20001 UDP/TLS/RTP/SAVPF 96
a=mid:v1
a=recvonly
The Section 7.2 example later in this document shows a more involved
case of "LS" group generation.
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 are
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. Each offered The next step is to go through each offered m= section. Each offered
m= section will have an associated RtpTransceiver, as described in m= section will have an associated RtpTransceiver, as described in
Section 5.9. If there are more RtpTransceivers than there are m= Section 5.9. If there are more RtpTransceivers than there are m=
sections, the unmatched RtpTransceivers will need to be associated in sections, the unmatched RtpTransceivers will need to be associated in
a subsequent offer. a subsequent offer.
For each offered m= section, if any of the following conditions are For each offered m= section, if any of the following conditions are
true, the corresponding m= section in the answer MUST be marked as true, the corresponding m= section in the answer MUST be marked as
skipping to change at page 49, line 7 skipping to change at page 51, line 16
o The <proto> field MUST be set to exactly match the <proto> field o The <proto> field MUST be set to exactly match the <proto> field
for the corresponding m= line in the offer. for the corresponding m= line in the offer.
o If codec preferences have been set for the associated transceiver, o If codec preferences have been set for the associated transceiver,
media formats MUST be generated in the corresponding order, and media formats MUST be generated in the corresponding order, and
MUST exclude any codecs not present in the codec preferences or MUST exclude any codecs not present in the codec preferences or
not present in the offer. Note that non-JSEP endpoints are not not present in the offer. Note that non-JSEP endpoints are not
subject to this restriction, and might add media formats in the subject to this restriction, and might add media formats in the
answer that are not present in the offer, as specified in answer that are not present in the offer, as specified in
[RFC3264], Section 6.1. Therefore, JSEP endpoints MUST be [RFC3264], Section 6.1. Therefore, JSEP implementations MUST be
prepared to receive such answers. prepared to receive such answers.
o Unless excluded by the above restrictions, the media formats MUST o Unless excluded by the above restrictions, the media formats MUST
include the mandatory audio/video codecs as specified in include the mandatory audio/video codecs as specified in
[I-D.ietf-rtcweb-audio](see Section 3) and [I-D.ietf-rtcweb-audio](see Section 3) and
[I-D.ietf-rtcweb-video](see Section 5). [I-D.ietf-rtcweb-video](see Section 5).
The m= line MUST be followed immediately by a "c=" line, as specified The m= line MUST be followed immediately by a "c=" line, as specified
in [RFC4566], Section 5.7. Again, as no candidates are available in [RFC4566], Section 5.7. Again, as no candidates are available
yet, the "c=" line must contain the "dummy" value "IN IP4 0.0.0.0", yet, the "c=" line must contain the "dummy" value "IN IP4 0.0.0.0",
skipping to change at page 49, line 41 skipping to change at page 51, line 50
the offered direction specified in [RFC3264], Section 6.1, and the offered direction specified in [RFC3264], Section 6.1, and
then intersecting with the direction of the associated then intersecting with the direction of the associated
RtpTransceiver. For example, in the case where an m= section is RtpTransceiver. For example, in the case where an m= section is
offered as "sendonly", and the local transceiver is set to offered as "sendonly", and the local transceiver is set to
"sendrecv", the result in the answer is a "recvonly" direction. "sendrecv", the result in the answer is a "recvonly" direction.
o For each media format on the m= line, "a=rtpmap" and "a=fmtp" o For each media format on the m= line, "a=rtpmap" and "a=fmtp"
lines, as specified in [RFC4566], Section 6, and [RFC3264], lines, as specified in [RFC4566], Section 6, and [RFC3264],
Section 6.1. Section 6.1.
o If this m= section is for media with configurable durations of
media per packet, e.g., audio, an "a=maxptime" line, as described
in Section 5.2.
o If this m= section is for video media, and there are known
limitations on the size of images which can be decoded, an
"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],
Section 6, and specific usage for each media type is outlined in Section 6, and specific usage for each media type is outlined in
Sections 4 and 5. Sections 4 and 5.
o If this m= section is for media with configurable durations of
media per packet, e.g., audio, an "a=maxptime" line, as described
in Section 5.2.
o If this m= section is for video media, and there are known
limitations on the size of images which can be decoded, an
"a=imageattr" line, as specified in Section 3.6.
o For each supported RTP header extension that is present in the o For each supported RTP header extension that is present in the
offer, an "a=extmap" line, as specified in [RFC5285], Section 5. offer, an "a=extmap" line, as specified in [RFC5285], Section 5.
The list of header extensions that SHOULD/MUST be supported is The list of header extensions that SHOULD/MUST be supported is
specified in [I-D.ietf-rtcweb-rtp-usage], Section 5.2. Any header specified in [I-D.ietf-rtcweb-rtp-usage], Section 5.2. Any header
extensions that require encryption MUST be specified as indicated extensions that require encryption MUST be specified as indicated
in [RFC6904], Section 4. in [RFC6904], Section 4.
o For each supported RTCP feedback mechanism that is present in the o For each supported RTCP feedback mechanism that is present in the
offer, an "a=rtcp-fb" mechanism, as specified in [RFC4585], offer, an "a=rtcp-fb" 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 the RtpTransceiver has a sendrecv or sendonly direction: o If the RtpTransceiver has a sendrecv or sendonly direction:
* An "a=msid" line, as specified in [I-D.ietf-mmusic-msid], * For each MediaStream that was associated with the transceiver
Section 2. when it was created via addTrack or addTransceiver, an "a=msid"
line, as specified in [I-D.ietf-mmusic-msid], Section 2. If a
MediaStreamTrack is attached to the transceiver's RtpSender,
the "a=msid" lines MUST use that track's ID. If no
MediaStreamTrack is attached, a valid ID MUST be generated, in
the same way that the implementation generates IDs for local
tracks.
* If no MediaStream is associated with the transceiver, a single
"a=msid" line with the special value "-" in place of the
MediaStream ID, as specified in [I-D.ietf-mmusic-msid],
Section 3. The track ID MUST be selected as described above.
Each m= section which is not bundled into another m= section, MUST Each m= section which is not bundled into another m= section, MUST
contain the following attributes (which are of category IDENTICAL or contain the following attributes (which are of category IDENTICAL or
TRANSPORT): TRANSPORT):
o "a=ice-ufrag" and "a=ice-pwd" lines, as specified in [RFC5245], o "a=ice-ufrag" and "a=ice-pwd" lines, as specified in [RFC5245],
Section 15.4. Section 15.4.
o An "a=fingerprint" line for each of the endpoint's certificates, o An "a=fingerprint" line for each of the endpoint's certificates,
as specified in [RFC4572], Section 5; the digest algorithm used as specified in [RFC4572], Section 5; the digest algorithm used
skipping to change at page 51, line 15 skipping to change at page 53, line 35
o If present in the offer, an "a=rtcp-mux" line, as specified in o If present in the offer, an "a=rtcp-mux" line, as specified in
[RFC5761], Section 5.1.3. Otherwise, an "a=rtcp" line, as [RFC5761], Section 5.1.3. Otherwise, an "a=rtcp" line, as
specified in [RFC3605], Section 2.1, containing the dummy value "9 specified in [RFC3605], Section 2.1, containing the dummy value "9
IN IP4 0.0.0.0" (because no candidates have yet been gathered). IN IP4 0.0.0.0" (because no candidates have yet been gathered).
o If present in the offer, an "a=rtcp-rsize" line, as specified in o If present in the offer, an "a=rtcp-rsize" line, as specified in
[RFC5506], Section 5. [RFC5506], Section 5.
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, an "a=mid" line MUST be generated and
"a=candidate", "a=fingerprint", "a=dtls-id", and "a=setup" lines MUST included as described above, along with an "a=sctp-port" line
be included under the conditions described above, along with an referencing the SCTP port number, as defined in
"a=fmtp:webrtc-datachannel" line and an "a=sctp-port" line [I-D.ietf-mmusic-sctp-sdp], Section 5.1, and, if appropriate, an
referencing the SCTP port number as defined in "a=max-message-size" line, as defined in [I-D.ietf-mmusic-sctp-sdp],
[I-D.ietf-mmusic-sctp-sdp], Section 4.1. Section 6.1.
As discussed above, the following attributes of category IDENTICAL or
TRANSPORT are included only if the data m= section is not bundled
into another m= section:
o "a=ice-ufrag"
o "a=ice-pwd"
o "a=fingerprint"
o "a=setup"
o "a=dtls-id"
Note that if media m= sections are bundled into a data m= section,
then certain TRANSPORT and IDENTICAL attributes may also appear in
the data m= section even if they would otherwise only be appropriate
for a media m= section (e.g., "a=rtcp-mux").
If "a=group" attributes with semantics of "BUNDLE" are offered, If "a=group" attributes with semantics of "BUNDLE" are offered,
corresponding session-level "a=group" attributes MUST be added as corresponding session-level "a=group" attributes MUST be added as
specified in [RFC5888]. These attributes MUST have semantics specified in [RFC5888]. These attributes MUST have semantics
"BUNDLE", and MUST include the all mid identifiers from the offered "BUNDLE", and MUST include the all mid identifiers from the offered
bundle groups that have not been rejected. Note that regardless of bundle groups that have not been rejected. Note that regardless of
the presence of "a=bundle-only" in the offer, no m= sections in the the presence of "a=bundle-only" in the offer, no m= sections in the
answer should have an "a=bundle-only" line. answer should have an "a=bundle-only" line.
Attributes that are common between all m= sections MAY be moved to Attributes that are common between all m= sections MAY be moved to
skipping to change at page 52, line 20 skipping to change at page 55, line 12
o The "s=" and "t=" lines MUST stay the same. o The "s=" and "t=" lines MUST stay the same.
o Each "m=" and c=" line MUST be filled in with the port and address o Each "m=" and c=" line MUST be filled in with the port and address
of the default candidate for the m= section, as described in of the default candidate for the m= section, as described in
[RFC5245], Section 4.3. Note, however, that the m= line protocol [RFC5245], Section 4.3. Note, however, that the m= line protocol
need not match the default candidate, because this protocol value need not match the default candidate, because this protocol value
must instead match what was supplied in the offer, as described must instead match what was supplied in the offer, as described
above. above.
o The media formats on the m= line MUST be generated in the same o Unless codec preferences have been set for the associated
order as in the current local description. transceiver, the media formats on the m= line MUST be generated in
the same order as in the current local description.
o Each "a=ice-ufrag" and "a=ice-pwd" line MUST stay the same, unless o Each "a=ice-ufrag" and "a=ice-pwd" line MUST stay the same, unless
the m= section is restarting, in which case new ICE credentials the m= section is restarting, in which case new ICE credentials
must be created as specified in [RFC5245], Section 9.2.1.1. If must be created as specified in [RFC5245], Section 9.2.1.1. If
the m= section is bundled into another m= section, it still MUST the m= section is bundled into another m= section, it still MUST
NOT contain any ICE credentials. NOT contain any ICE credentials.
o Each "a=setup" line MUST use an "active" or "passive" role value o Each "a=setup" line MUST use an "active" or "passive" role value
consistent with the existing DTLS association, if the association consistent with the existing DTLS association, if the association
is being continued by the offerer. is being continued by the offerer.
skipping to change at page 52, line 49 skipping to change at page 55, line 42
o If the m= section is not bundled into another m= section, for each o If the m= section is not bundled into another m= section, for each
candidate that has been gathered during the most recent gathering candidate that has been gathered during the most recent gathering
phase (see Section 3.5.1), an "a=candidate" line MUST be added, as phase (see Section 3.5.1), an "a=candidate" line MUST be added, as
defined in [RFC5245], Section 4.3., paragraph 3. If candidate defined in [RFC5245], Section 4.3., paragraph 3. If candidate
gathering for the section has completed, an "a=end-of-candidates" gathering for the section has completed, an "a=end-of-candidates"
attribute MUST be added, as described in [I-D.ietf-ice-trickle], attribute MUST be added, as described in [I-D.ietf-ice-trickle],
Section 9.3. If the m= section is bundled into another m= Section 9.3. If the m= section is bundled into another m=
section, both "a=candidate" and "a=end-of-candidates" MUST be section, both "a=candidate" and "a=end-of-candidates" MUST be
omitted. omitted.
o For RtpTransceivers that are not stopped, the "a=msid" line MUST o For RtpTransceivers that are not stopped, the "a=msid" line(s)
stay the same. MUST stay the same, regardless of changes to the transceiver's
direction or track. If no "a=msid" line is present in the current
description, "a=msid" line(s) MUST be generated according to the
same rules as for an initial answer.
5.3.3. Options Handling 5.3.3. Options Handling
The createAnswer method takes as a parameter an RTCAnswerOptions The createAnswer method takes as a parameter an RTCAnswerOptions
object. The set of parameters for RTCAnswerOptions is different than object. The set of parameters for RTCAnswerOptions is different than
those supported in RTCOfferOptions; the IceRestart option is those supported in RTCOfferOptions; the IceRestart option is
unnecessary, as ICE credentials will automatically be changed for all unnecessary, as ICE credentials will automatically be changed for all
m= sections where the offerer chose to perform ICE restart. m= sections where the offerer chose to perform ICE restart.
The following options are supported in RTCAnswerOptions. The following options are supported in RTCAnswerOptions.
skipping to change at page 53, line 48 skipping to change at page 56, line 42
and specifies a subset of what was in the original offer. This is and specifies a subset of what was in the original offer. This is
safe because the answer is not permitted to expand capabilities, and safe because the answer is not permitted to expand capabilities, and
therefore will just respond to what is present in the offer. therefore will just respond to what is present in the offer.
The application SHOULD NOT modify the SDP in the answer it transmits, The application SHOULD NOT modify the SDP in the answer it transmits,
as the answer contains the negotiated capabilities, and this can as the answer contains the negotiated capabilities, and this can
cause the two sides to have different ideas about what exactly was cause the two sides to have different ideas about what exactly was
negotiated. negotiated.
As always, the application is solely responsible for what it sends to As always, the application is solely responsible for what it sends to
the other party, and all incoming SDP will be processed by the the other party, and all incoming SDP will be processed by the JSEP
browser to the extent of its capabilities. It is an error to assume implementation to the extent of its capabilities. It is an error to
that all SDP is well-formed; however, one should be able to assume assume that all SDP is well-formed; however, one should be able to
that any implementation of this specification will be able to assume that any implementation of this specification will be able to
process, as a remote offer or answer, unmodified SDP coming from any process, as a remote offer or answer, unmodified SDP coming from any
other implementation of this specification. other implementation of this specification.
5.5. Processing a Local Description 5.5. Processing a Local Description
When a SessionDescription is supplied to setLocalDescription, the When a SessionDescription is supplied to setLocalDescription, the
following steps MUST be performed: following steps MUST be performed:
o First, the type of the SessionDescription is checked against the o 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".
* If the type is "pranswer" or "answer", the PeerConnection state * If the type is "pranswer" or "answer", the PeerConnection state
MUST be either "have-remote-offer" or "have-local-pranswer". MUST be either "have-remote-offer" or "have-local-pranswer".
o If the type is not correct for the current state, processing MUST o If the type is not correct for the current state, processing MUST
stop and an error MUST be returned. stop and an error MUST be returned.
o The SessionDescription is then checked to ensure that its contents
are identical to those generated in the last call to createOffer/
createAnswer, and thus have not been altered, as discussed in
Section 5.4; otherwise, processing MUST stop and an error MUST be
returned.
o Next, the SessionDescription is parsed into a data structure, as o Next, the SessionDescription is parsed into a data structure, as
described in the Section 5.7 section below. If parsing fails for described in the Section 5.7 section below. If parsing fails for
any reason, processing MUST stop and an error MUST be returned. any reason, processing MUST stop and an error MUST be returned.
o Finally, the parsed SessionDescription is applied as described in o Finally, the parsed SessionDescription is applied as described in
the Section 5.8 section below. the Section 5.8 section below.
5.6. Processing a Remote Description 5.6. Processing a Remote Description
When a SessionDescription is supplied to setRemoteDescription, the When a SessionDescription is supplied to setRemoteDescription, the
skipping to change at page 56, line 48 skipping to change at page 60, line 5
Section 5, and the set of fingerprint and algorithm values is Section 5, and the set of fingerprint and algorithm values is
stored. stored.
o If present, a single "a=setup" line is parsed as specified in o If present, a single "a=setup" line is parsed as specified in
[RFC4145], Section 4, and the setup value is stored. [RFC4145], Section 4, and the setup value is stored.
o If present, a single "a=dtls-id" line is parsed as specified in o If present, a single "a=dtls-id" line is parsed as specified in
[I-D.ietf-mmusic-dtls-sdp] Section 5, and the dtls-id value is [I-D.ietf-mmusic-dtls-sdp] Section 5, and the dtls-id value is
stored. stored.
o Any "a=identity" lines are parsed and the identity values stored
for subsequent verification, as specified
[I-D.ietf-rtcweb-security-arch], Section 5.
o Any "a=extmap" lines are parsed as specified in [RFC5285], o Any "a=extmap" lines are parsed as specified in [RFC5285],
Section 5, and their values are stored. Section 5, and their values are stored.
As required by [RFC4566], Section 5.13, unknown attribute lines MUST
be ignored.
Once all the session-level lines have been parsed, processing Once all the session-level lines have been parsed, processing
continues with the lines in media sections. continues with the lines in m= sections.
5.7.2. Media Section Parsing 5.7.2. Media Section Parsing
Like the session-level lines, the media session lines MUST occur in Like the session-level lines, the media section lines MUST occur in
the specific order and with the specific syntax defined in [RFC4566], the specific order and with the specific syntax defined in [RFC4566],
Section 5. Section 5.
The "m=" line itself MUST be parsed as described in [RFC4566], The "m=" line itself MUST be parsed as described in [RFC4566],
Section 5.14, and the media, port, proto, and fmt values stored. Section 5.14, and the media, port, proto, and fmt values stored.
Following the "m=" line, specific processing MUST be applied for the Following the "m=" line, specific processing MUST be applied for the
following non-attribute lines: following non-attribute lines:
o As with the "c=" line at the session level, the "c=" line MUST be o As with the "c=" line at the session level, the "c=" line MUST be
skipping to change at page 58, line 6 skipping to change at page 61, line 17
o If present, a single "a=end-of-candidates" attribute MUST be o If present, a single "a=end-of-candidates" attribute MUST be
parsed as specified in [I-D.ietf-ice-trickle], Section 8.2, and parsed as specified in [I-D.ietf-ice-trickle], Section 8.2, and
its presence or absence flagged and stored. its presence or absence flagged and stored.
o Any "a=fingerprint" lines are parsed as specified in [RFC4572], o Any "a=fingerprint" lines are parsed as specified in [RFC4572],
Section 5, and the set of fingerprint and algorithm values is Section 5, and the set of fingerprint and algorithm values is
stored. stored.
If the "m=" proto value indicates use of RTP, as described in the If the "m=" proto value indicates use of RTP, as described in the
Section 5.1.3 section above, the following attribute lines MUST be Section 5.1.2 section above, the following attribute lines MUST be
processed: processed:
o The "m=" fmt value MUST be parsed as specified in [RFC4566], o The "m=" fmt value MUST be parsed as specified in [RFC4566],
Section 5.14, and the individual values stored. Section 5.14, and the individual values stored.
o Any "a=rtpmap" or "a=fmtp" lines MUST be parsed as specified in o Any "a=rtpmap" or "a=fmtp" lines MUST be parsed as specified in
[RFC4566], Section 6, and their values stored. [RFC4566], Section 6, and their values stored.
o If present, a single "a=ptime" line MUST be parsed as described in o If present, a single "a=ptime" line MUST be parsed as described in
[RFC4566], Section 6, and its value stored. [RFC4566], Section 6, and its value stored.
skipping to change at page 58, line 50 skipping to change at page 62, line 13
presence or absence flagged and stored. presence or absence flagged and stored.
o If present, a single "a=rtcp-rsize" attribute MUST be parsed as o If present, a single "a=rtcp-rsize" attribute MUST be parsed as
specified in [RFC5506], Section 5, and its presence or absence specified in [RFC5506], Section 5, and its presence or absence
flagged and stored. flagged and stored.
o If present, a single "a=rtcp" attribute MUST be parsed as o If present, a single "a=rtcp" attribute MUST be parsed as
specified in [RFC3605], Section 2.1, but its value is ignored, as specified in [RFC3605], Section 2.1, but its value is ignored, as
this information is superfluous when using ICE. this information is superfluous when using ICE.
o If present, a single "a=msid" attribute MUST be parsed as o If present, "a=msid" attributes MUST be parsed as specified in
specified in [I-D.ietf-mmusic-msid], Section 3.2, and its value [I-D.ietf-mmusic-msid], Section 3.2, and their values stored.
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 o Any "a=rid" lines MUST be parsed as specified in
[I-D.ietf-mmusic-rid], Section 10, and their values stored. [I-D.ietf-mmusic-rid], Section 10, and their values stored.
o If present, a single "a=simulcast" line MUST be parsed as o If present, a single "a=simulcast" line MUST be parsed as
specified in [I-D.ietf-mmusic-sdp-simulcast], and its values specified in [I-D.ietf-mmusic-sdp-simulcast], and its values
stored. stored.
skipping to change at page 59, line 30 skipping to change at page 62, line 41
protocol value stored. protocol value stored.
o An "a=sctp-port" attribute MUST be present, and it MUST be parsed o An "a=sctp-port" attribute MUST be present, and it MUST be parsed
as specified in [I-D.ietf-mmusic-sctp-sdp], Section 5.2, and the as specified in [I-D.ietf-mmusic-sctp-sdp], Section 5.2, and the
value stored. value stored.
o If present, a single "a=max-message-size" attribute MUST be parsed o If present, a single "a=max-message-size" attribute MUST be parsed
as specified in [I-D.ietf-mmusic-sctp-sdp], Section 6, and the as specified in [I-D.ietf-mmusic-sctp-sdp], Section 6, and the
value stored. Otherwise, use the specified default. value stored. Otherwise, use the specified default.
As required by [RFC4566], Section 5.13, unknown attribute lines MUST
be ignored.
5.7.3. Semantics Verification 5.7.3. Semantics Verification
Assuming parsing completes successfully, the parsed description is Assuming parsing completes successfully, the parsed description is
then evaluated to ensure internal consistency as well as proper then evaluated to ensure internal consistency as well as proper
support for mandatory features. Specifically, the following checks support for mandatory features. Specifically, the following checks
are performed: are performed:
o For each m= section, valid values for each of the mandatory-to-use o For each m= section, valid values for each of the mandatory-to-use
features enumerated in Section 5.1.2 MUST be present. These features enumerated in Section 5.1.1 MUST be present. These
values MAY either be present at the media level, or inherited from values MAY either be present at the media level, or inherited from
the session level. the session level.
* ICE ufrag and password values, which MUST comply with the size * ICE ufrag and password values, which MUST comply with the size
limits specified in [RFC5245], Section 15.4. limits specified in [RFC5245], Section 15.4.
* dtls-id value, which MUST be set according to * dtls-id value, which MUST be set according to
[I-D.ietf-mmusic-dtls-sdp] Section 5. If this is a re-offer [I-D.ietf-mmusic-dtls-sdp] Section 5. If this is a re-offer
and the dtls-id value is different from that presently in use, and the dtls-id value is different from that presently in use,
the DTLS connection is not being continued and the remote the DTLS connection is not being continued and the remote
skipping to change at page 60, line 21 skipping to change at page 63, line 34
present. present.
o All RID values referenced in an "a=simulcast" line MUST exist as o All RID values referenced in an "a=simulcast" line MUST exist as
"a=rid" lines. "a=rid" lines.
o Each m= section is also checked to ensure prohibited features are o Each m= section is also checked to ensure prohibited features are
not used. If this is a local description, the "ice-lite" not used. If this is a local description, the "ice-lite"
attribute MUST NOT be specified. attribute MUST NOT be specified.
o If the RTP/RTCP multiplexing policy is "require", each m= section o If the RTP/RTCP multiplexing policy is "require", each m= section
MUST contain an "a=rtcp-mux" attribute. MUST contain an "a=rtcp-mux" attribute. If an "m=" section
contains an "a=rtcp-mux-only" attribute then that section MUST
also contain an "a=rtcp-mux" attribute.
If this session description is of type "pranswer" or "answer", the If this session description is of type "pranswer" or "answer", the
following additional checks are applied: following additional checks are applied:
o The session description must follow the rules defined in o The session description must follow the rules defined in
[RFC3264], Section 6, including the requirement that the number of [RFC3264], Section 6, including the requirement that the number of
m= sections MUST exactly match the number of m= sections in the m= sections MUST exactly match the number of m= sections in the
associated offer. associated offer.
o For each m= section, the media type and protocol values MUST o For each m= section, the media type and protocol values MUST
exactly match the media type and protocol values in the exactly match the media type and protocol values in the
corresponding m= section in the associated offer. corresponding m= section in the associated offer.
If any of the preceding checks failed, processing MUST stop and an If any of the preceding checks failed, processing MUST stop and an
error MUST be returned. error MUST be returned.
5.8. Applying a Local Description 5.8. Applying a Local Description
The following steps are performed at the media engine level to apply The following steps are performed at the media engine level to apply
a local description. a local description. If an error is returned, the session MUST be
restored to the state it was in before performing these steps.
First, the parsed parameters are checked to ensure that they are Next, m= sections are processed. For each m= section, the following
identical to those generated in the last call to createOffer/ steps MUST be performed; if any parameters are out of bounds, or
createAnswer, and thus have not been altered, as discussed in cannot be applied, processing MUST stop and an error MUST be
Section 5.4; otherwise, processing MUST stop and an error MUST be
returned. returned.
Next, media sections are processed. For each media section, the o If this m= section is new, begin gathering candidates for it, as
following steps MUST be performed; if any parameters are out of defined in [RFC5245], Section 4.1.1, unless it has been marked as
bounds, or cannot be applied, processing MUST stop and an error MUST bundle-only.
be returned.
o If this media section is new, begin gathering candidates for it,
as defined in [RFC5245], Section 4.1.1, unless it has been marked
as bundle-only.
o Or, if the ICE ufrag and password values have changed, and it has o Or, if the ICE ufrag and password values have changed, and it has
not been marked as bundle-only, trigger the ICE Agent to start an not been marked as bundle-only, trigger the ICE agent to start an
ICE restart, and begin gathering new candidates for the media ICE restart, and begin gathering new candidates for the m= section
section as described in [RFC5245], Section 9.1.1.1. If this as described in [RFC5245], Section 9.1.1.1. If this description
description is an answer, also start checks on that media section is an answer, also start checks on that media section as defined
as defined in [RFC5245], Section 9.3.1.1. in [RFC5245], Section 9.3.1.1.
o If the media section proto value indicates use of RTP: o If the m= section proto value indicates use of RTP:
* If there is no RtpTransceiver associated with this m= section * If there is no RtpTransceiver associated with this m= section
(which will only happen when applying an offer), find one and (which will only happen when applying an offer), find one and
associate it with this m= section according to the following associate it with this m= section according to the following
steps: steps:
+ Find the RtpTransceiver that corresponds to this m= section, + Find the RtpTransceiver that corresponds to this m= section,
using the mapping between transceivers and m= section using the mapping between transceivers and m= section
indices established when creating the offer. indices established when creating the offer.
skipping to change at page 61, line 42 skipping to change at page 65, line 6
Section 5.1.3. If RTCP mux is not indicated, but was Section 5.1.3. If RTCP mux is not indicated, but was
previously negotiated, i.e., the RTCP ICE component no longer previously negotiated, i.e., the RTCP ICE component no longer
exists, this MUST result in an error. exists, this MUST result in an error.
* For each specified RTP header extension, establish a mapping * For each specified RTP header extension, establish a mapping
between the extension ID and URI, as described in section 6 of between the extension ID and URI, as described in section 6 of
[RFC5285]. If any indicated RTP header extension is not [RFC5285]. If any indicated RTP header extension is not
supported, this MUST result in an error. supported, this MUST result in an error.
* If the MID header extension is supported, prepare to demux RTP * If the MID header extension is supported, prepare to demux RTP
streams intended for this media section based on the MID header streams intended for this m= section based on the MID header
extension, as described in extension, as described in
[I-D.ietf-mmusic-sdp-bundle-negotiation], Section 14. [I-D.ietf-mmusic-sdp-bundle-negotiation], Section 14.
* For each specified media format, establish a mapping between * For each specified media format, establish a mapping between
the payload type and the actual media format, as described in the payload type and the actual media format, as described in
[RFC3264], Section 6.1. If any indicated media format is not [RFC3264], Section 6.1. If any indicated media format is not
supported, this MUST result in an error. supported, this MUST result in an error.
* For each specified "rtx" media format, establish a mapping * For each specified "rtx" media format, establish a mapping
between the RTX payload type and its associated primary payload between the RTX payload type and its associated primary payload
skipping to change at page 62, line 16 skipping to change at page 65, line 29
result in an error. result in an error.
* If the directional attribute is of type "sendrecv" or * If the directional attribute is of type "sendrecv" or
"recvonly", enable receipt and decoding of media. "recvonly", enable receipt and decoding of media.
Finally, if this description is of type "pranswer" or "answer", Finally, if this description is of type "pranswer" or "answer",
follow the processing defined in the Section 5.10 section below. follow the processing defined in the Section 5.10 section below.
5.9. Applying a Remote Description 5.9. Applying a Remote Description
The following steps are performed to apply a remote description. If
an error is returned, the session MUST be restored to the state it
was in before performing these steps.
If the answer contains any "a=ice-options" attributes where "trickle" If the answer contains any "a=ice-options" attributes where "trickle"
is listed as an attribute, update the PeerConnection canTrickle is listed as an attribute, update the PeerConnection canTrickle
property to be true. Otherwise, set this property to false. property to be true. Otherwise, set this property to false.
The following steps are performed at the media engine level to apply
a remote description.
The following steps MUST be performed for attributes at the session The following steps MUST be performed for attributes at the session
level; if any parameters are out of bounds, or cannot be applied, level; if any parameters are out of bounds, or cannot be applied,
processing MUST stop and an error MUST be returned. processing MUST stop and an error MUST be returned.
o For any specified "CT" bandwidth value, set this as the limit for o For any specified "CT" bandwidth value, set this as the limit for
the maximum total bitrate for all m= sections, as specified in the maximum total bitrate for all m= sections, as specified in
Section 5.8 of [RFC4566]. Within this overall limit, the Section 5.8 of [RFC4566]. Within this overall limit, the
implementation can dynamically decide how to best allocate the implementation can dynamically decide how to best allocate the
available bandwidth between m= sections, respecting any specific available bandwidth between m= sections, respecting any specific
limits that have been specified for individual m= sections. limits that have been specified for individual m= sections.
o For any specified "RR" or "RS" bandwidth values, handle as o For any specified "RR" or "RS" bandwidth values, handle as
specified in [RFC3556], Section 2. specified in [RFC3556], Section 2.
o Any "AS" bandwidth value MUST be ignored, as the meaning of this o Any "AS" bandwidth value MUST be ignored, as the meaning of this
construct at the session level is not well defined. construct at the session level is not well defined.
For each media section, the following steps MUST be performed; if any For each m= section, the following steps MUST be performed; if any
parameters are out of bounds, or cannot be applied, processing MUST parameters are out of bounds, or cannot be applied, processing MUST
stop and an error MUST be returned. stop and an error MUST be returned.
o If the ICE ufrag or password changed from the previous remote o If the PeerConnection state is "have-local-offer", and the ICE
description, then an ICE restart is needed, as described in ufrag or password changed from the previous remote description,
Section 9.1.1.1 of [RFC5245] If the description is of type then an ICE restart is needed, as described in Section 9.1.1.1 of
"offer", mark that an ICE restart is needed. If the description [RFC5245]. If the description is of type "offer", note that an
is of type "answer" and the current local description is also an ICE restart is needed. If the description is of type "answer" or
ICE restart, then signal the ICE agent to begin checks as "pranswer" and the current local description is also an ICE
described in Section 9.3.1.1 of [RFC5245]. An answer MUST change restart, then signal the ICE agent to begin checks as described in
the ufrag and password in an answer if and only if ICE is Section 9.3.1.1 of [RFC5245]. An answerer MUST change the ufrag
restarting, as described in Section 9.2.1.1 of [RFC5245]. and password in an answer if and only if ICE is restarting, as
described in Section 9.2.1.1 of [RFC5245].
o If the PeerConnection state is "have-remote-pranswer", and the ICE
ufrag or password changed from the previous provisional answer,
then signal the ICE agent to discard any previous ICE check list
state for the m= section and begin checks as if this were the
first answer. However, such an answer MAY only change the ICE
ufrag or password if the local offer is starting or restarting ICE
for the m= section.
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- o If an "a=end-of-candidates" attribute is present, process the end-
of-candidates indication as described in [I-D.ietf-ice-trickle] of-candidates indication as described in [I-D.ietf-ice-trickle]
Section 11. Section 11.
o If the media section proto value indicates use of RTP: o If the m= section proto value indicates use of RTP:
* If the m= section is being recycled (see Section 5.2.2), * If the m= section is being recycled (see Section 5.2.2),
dissociate the currently associated RtpTransceiver by setting dissociate the currently associated RtpTransceiver by setting
its mid property to null, and discard the mapping between the its mid property to null, and discard the mapping between the
transceiver and its m= section index. transceiver and its m= section index.
* If the m= section is not associated with any RtpTransceiver * If the m= section is not associated with any RtpTransceiver
(possibly because it was dissociated in the previous step), (possibly because it was dissociated in the previous step),
either find an RtpTransceiver or create one according to the either find an RtpTransceiver or create one according to the
following steps: following steps:
skipping to change at page 65, line 12 skipping to change at page 68, line 30
5% to RTCP. "TIAS" is used in preference to "AS" because it 5% to RTCP. "TIAS" is used in preference to "AS" because it
provides more accurate control of bandwidth. provides more accurate control of bandwidth.
* For any "RR" or "RS" bandwidth values, handle as specified in * For any "RR" or "RS" bandwidth values, handle as specified in
[RFC3556], Section 2. [RFC3556], Section 2.
* Any specified "CT" bandwidth value MUST be ignored, as the * Any specified "CT" bandwidth value MUST be ignored, as the
meaning of this construct at the media level is not well meaning of this construct at the media level is not well
defined. defined.
* If the media section is of type audio: * If the m= section is of type audio:
+ For each specified "CN" media format, enable DTX for all + For each specified "CN" media format, enable DTX for all
supported media formats with the same clockrate, as supported media formats with the same clockrate, as
described in [RFC3389], Section 5, except for formats that described in [RFC3389], Section 5, except for formats that
have their own internal DTX mechanisms. DTX for such have their own internal DTX mechanisms. DTX for such
formats (e.g., Opus) is controlled via fmtp parameters, as formats (e.g., Opus) is controlled via fmtp parameters, as
discussed in Section 5.2.3.2. discussed in Section 5.2.3.2.
+ For each specified "telephone-event" media format, enable + For each specified "telephone-event" media format, enable
DTMF transmission for all supported media formats with the DTMF transmission for all supported media formats with the
skipping to change at page 65, line 42 skipping to change at page 69, line 11
Finally, if this description is of type "pranswer" or "answer", Finally, if this description is of type "pranswer" or "answer",
follow the processing defined in the Section 5.10 section below. follow the processing defined in the Section 5.10 section below.
5.10. Applying an Answer 5.10. Applying an Answer
In addition to the steps mentioned above for processing a local or In addition to the steps mentioned above for processing a local or
remote description, the following steps are performed when processing remote description, the following steps are performed when processing
a description of type "pranswer" or "answer". a description of type "pranswer" or "answer".
For each media section, the following steps MUST be performed: For each m= section, the following steps MUST be performed:
o If the media section has been rejected (i.e. port is set to zero o If the m= section has been rejected (i.e. port is set to zero in
in the answer), stop any reception or transmission of media for the answer), stop any reception or transmission of media for this
this section, and, unless a non-rejected media section is bundled section, and, unless a non-rejected m= section is bundled with
with this media section, discard any associated ICE components, as this m= section, discard any associated ICE components, as
described in Section 9.2.1.3 of [RFC5245]. described in Section 9.2.1.3 of [RFC5245].
o If the remote DTLS fingerprint has been changed or the dtls-id has o If the remote DTLS fingerprint has been changed or the dtls-id has
changed, tear down the DTLS connection. If a DTLS connection changed, tear down the DTLS connection. This includes the case
needs to be torn down but the answer does not indicate an ICE when the PeerConnection state is "have-remote-pranswer". If a
restart, an error MUST be generated. If an ICE restart is DTLS connection needs to be torn down but the answer does not
performed without a change in dtls-id or fingerprint, then the indicate an ICE restart or, in the case of "have-remote-pranswer",
same DTLS connection is continued over the new ICE channel. new ICE credentials, an error MUST be generated. If an ICE
restart is performed without a change in dtls-id or fingerprint,
then the same DTLS connection is continued over the new ICE
channel.
o If no valid DTLS connection exists, prepare to start a DTLS o If no valid DTLS connection exists, prepare to start a DTLS
connection, using the specified roles and fingerprints, on any connection, using the specified roles and fingerprints, on any
underlying ICE components, once they are active. underlying ICE components, once they are active.
o If the media section proto value indicates use of RTP: o If the m= section proto value indicates use of RTP:
* If the media section references any media formats, RTP header * If the m= section references any media formats, RTP header
extensions, or RTCP feedback mechanisms that were not present extensions, or RTCP feedback mechanisms that were not present
in the corresponding media section in the offer, this indicates in the corresponding m= section in the offer, this indicates a
a negotiation problem and MUST result in an error. negotiation problem and MUST result in an error.
* If the media section has RTCP mux enabled, discard the RTCP ICE * If the m= section has RTCP mux enabled, discard the RTCP ICE
component, if one exists, and begin or continue muxing RTCP component, if one exists, and begin or continue muxing RTCP
over the RTP ICE component, as specified in [RFC5761], over the RTP ICE component, as specified in [RFC5761],
Section 5.1.3. Otherwise, prepare to transmit RTCP over the Section 5.1.3. Otherwise, prepare to transmit RTCP over the
RTCP ICE component; if no RTCP ICE component exists, because RTCP ICE component; if no RTCP ICE component exists, because
RTCP mux was previously enabled, this MUST result in an error. RTCP mux was previously enabled, this MUST result in an error.
* If the media section has reduced-size RTCP enabled, configure * If the m= section has reduced-size RTCP enabled, configure the
the RTCP transmission for this media section to use reduced- RTCP transmission for this m= section to use reduced-size RTCP,
size RTCP, as specified in [RFC5506]. as specified in [RFC5506].
* If the directional attribute in the answer is of type * If the directional attribute in the answer is of type
"sendrecv" or "sendonly", choose the media format to send as "sendrecv" or "sendonly", choose the media format to send as
the most preferred media format from the remote description the most preferred media format from the remote description
that is also present in the answer, as described in [RFC3264], that is also present in the answer, as described in [RFC3264],
Sections 6.1 and 7, and start transmitting RTP media once the Sections 6.1 and 7, and start transmitting RTP media once the
underlying transport layers have been established. If a SSRC underlying transport layers have been established. If an SSRC
has not already been chosen for this outgoing RTP stream, has not already been chosen for this outgoing RTP stream,
choose a random one. choose a random one. If media is already being transmitted,
the same SSRC SHOULD be used unless the clockrate of the new
codec is different, in which case a new SSRC MUST be chosen, as
specified in [RFC7160], Section 3.1.
* The payload type mapping from the remote description is used to * The payload type mapping from the remote description is used to
determine payload types for the outgoing RTP streams, including determine payload types for the outgoing RTP streams, including
the payload type for the send media format chosen above. Any the payload type for the send media format chosen above. Any
RTP header extensions that were negotiated should be included RTP header extensions that were negotiated should be included
in the outgoing RTP streams, using the extension mapping from in the outgoing RTP streams, using the extension mapping from
the remote description; if the RID header extension has been the remote description; if the RID header extension has been
negotiated, and RID values are specified, include the RID negotiated, and RID values are specified, include the RID
header extension in the outgoing RTP streams, as indicated in header extension in the outgoing RTP streams, as indicated in
[I-D.ietf-mmusic-rid], Section 4. [I-D.ietf-mmusic-rid], Section 4.
skipping to change at page 67, line 32 skipping to change at page 71, line 9
feedback types and reacting to received feedback, as specified feedback types and reacting to received feedback, as specified
in [RFC4585], Section 4.2. When sending RTCP feedback, follow in [RFC4585], Section 4.2. When sending RTCP feedback, follow
the rules and recommendations from the rules and recommendations from
[I-D.ietf-avtcore-rtp-multi-stream], Section 5.4.1 to select [I-D.ietf-avtcore-rtp-multi-stream], Section 5.4.1 to select
which SSRC to use. which SSRC to use.
* If the directional attribute is of type "recvonly" or * If the directional attribute is of type "recvonly" or
"inactive", stop transmitting all RTP media, but continue "inactive", stop transmitting all RTP media, but continue
sending RTCP, as described in [RFC3264], Section 5.1. sending RTCP, as described in [RFC3264], Section 5.1.
o If the media section proto value indicates use of SCTP: o If the m= section proto value indicates use of SCTP:
* If no SCTP association yet exists, prepare to initiate a SCTP * If an SCTP association exists, and the remote SCTP port has
changed, discard the existing SCTP association. This includes
the case when the PeerConnection state is "have-remote-
pranswer".
* If no valid SCTP association exists, prepare to initiate a SCTP
association over the associated ICE component and DTLS association over the associated ICE component and DTLS
connection, using the local SCTP port value from the local connection, using the local SCTP port value from the local
description, and the remote SCTP port value from the remote description, and the remote SCTP port value from the remote
description, as described in [I-D.ietf-mmusic-sctp-sdp], description, as described in [I-D.ietf-mmusic-sctp-sdp],
Section 10.2. Section 10.2.
If the answer contains valid bundle groups, discard any ICE If the answer contains valid bundle groups, discard any ICE
components for the m= sections that will be bundled onto the primary components for the m= sections that will be bundled onto the primary
ICE components in each bundle, and begin muxing these m= sections ICE components in each bundle, and begin muxing these m= sections
accordingly, as described in accordingly, as described in
skipping to change at page 68, line 35 skipping to change at page 72, line 13
multiple lines, where leading whitespace indicates that a line is a multiple lines, where leading whitespace indicates that a line is a
continuation of the previous line. In addition, some blank lines continuation of the previous line. In addition, some blank lines
have been added to improve readability but are not valid in SDP. have been added to improve readability but are not valid in SDP.
More examples of SDP for WebRTC call flows can be found in More examples of SDP for WebRTC call flows can be found in
[I-D.nandakumar-rtcweb-sdp]. [I-D.nandakumar-rtcweb-sdp].
7.1. Simple Example 7.1. Simple Example
This section shows a very simple example that sets up a minimal audio This section shows a very simple example that sets up a minimal audio
/ video call between two browsers and does not use trickle ICE. The / video call between two JSEP endpoints without using trickle ICE.
example in the following section provides a more realistic example of The example in the following section provides a more detailed example
what would happen in a normal browser to browser connection. of what could happen in a JSEP session.
The flow shows Alice's browser initiating the session to Bob's The code flow below shows Alice's endpoint initiating the session to
browser. The messages from Alice's JS to Bob's JS are assumed to Bob's endpoint. The messages from Alice's JS to Bob's JS are assumed
flow over some signaling protocol via a web server. The JS on both to flow over some signaling protocol via a web server. The JS on
Alice's side and Bob's side waits for all candidates before sending both Alice's side and Bob's side waits for all candidates before
the offer or answer, so the offers and answers are complete. Trickle sending the offer or answer, so the offers and answers are complete;
ICE is not used. Both Alice and Bob are using the default policy of trickle ICE is not used. Both Alice and Bob are using the default
balanced. bundle policy of "balanced", and the default RTCP mux policy of
"require".
// set up local media state // set up local media state
AliceJS->AliceUA: create new PeerConnection AliceJS->AliceUA: create new PeerConnection
AliceJS->AliceUA: addTrack with two tracks: audio and video AliceJS->AliceUA: addTrack with two tracks: audio and video
AliceJS->AliceUA: createOffer to get offer AliceJS->AliceUA: createOffer to get offer
AliceJS->AliceUA: setLocalDescription with offer AliceJS->AliceUA: setLocalDescription with offer
AliceUA->AliceJS: multiple onicecandidate events with candidates AliceUA->AliceJS: multiple onicecandidate events with candidates
// wait for ICE gathering to complete // wait for ICE gathering to complete
AliceUA->AliceJS: onicecandidate event with null candidate AliceUA->AliceJS: onicecandidate event with null candidate
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: ontrack events for audio and video tracks
// Bob accepts call // Bob accepts call
BobJS->BobUA: addTrack with local tracks BobJS->BobUA: addTrack with local tracks
BobJS->BobUA: createAnswer BobJS->BobUA: createAnswer
BobJS->BobUA: setLocalDescription with answer BobJS->BobUA: setLocalDescription with answer
BobUA->BobJS: multiple onicecandidate events with candidates BobUA->BobJS: multiple onicecandidate events with candidates
// wait for ICE gathering to complete // wait for ICE gathering to complete
BobUA->BobJS: onicecandidate event with null candidate BobUA->BobJS: onicecandidate event with null candidate
BobJS->BobUA: get |answer-A1| from currentLocalDescription BobJS->BobUA: get |answer-A1| from currentLocalDescription
// |answer-A1| is sent over signaling protocol to Alice // |answer-A1| is sent over signaling protocol to Alice
BobJS->WebServer: signaling with |answer-A1| BobJS->WebServer: signaling with |answer-A1|
WebServer->AliceJS: signaling with |answer-A1| WebServer->AliceJS: signaling with |answer-A1|
// |answer-A1| arrives at Alice // |answer-A1| arrives at Alice
AliceJS->AliceUA: setRemoteDescription with |answer-A1| AliceJS->AliceUA: setRemoteDescription with |answer-A1|
AliceUA->AliceJS: onaddstream event with remoteStream AliceUA->AliceJS: ontrack events for audio and video tracks
// media flows // media flows
BobUA->AliceUA: media sent from Bob to Alice BobUA->AliceUA: media sent from Bob to Alice
AliceUA->BobUA: media sent from Alice to Bob AliceUA->BobUA: media sent from Alice to Bob
The SDP for |offer-A1| looks like: The SDP for |offer-A1| looks like:
v=0 v=0
o=- 4962303333179871722 1 IN IP4 0.0.0.0 o=- 4962303333179871722 1 IN IP4 0.0.0.0
s=- s=-
t=0 0 t=0 0
a=group:BUNDLE a1 v1
a=ice-options:trickle a=ice-options:trickle
m=audio 56500 UDP/TLS/RTP/SAVPF 96 0 8 97 98 a=group:BUNDLE a1 v1
c=IN IP4 192.0.2.1 a=group:LS a1 v1
m=audio 10100 UDP/TLS/RTP/SAVPF 96 0 8 97 98
c=IN IP4 203.0.113.100
a=mid:a1 a=mid:a1
a=rtcp:56501 IN IP4 192.0.2.1
a=msid:47017fee-b6c1-4162-929c-a25110252400
f83006c5-a0ff-4e0a-9ed9-d3e6747be7d9
a=sendrecv a=sendrecv
a=rtpmap:96 opus/48000/2 a=rtpmap:96 opus/48000/2
a=rtpmap:0 PCMU/8000 a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000 a=rtpmap:8 PCMA/8000
a=rtpmap:97 telephone-event/8000 a=rtpmap:97 telephone-event/8000
a=rtpmap:98 telephone-event/48000 a=rtpmap:98 telephone-event/48000
a=maxptime:120 a=maxptime:120
a=ice-ufrag:ETEn1v9DoTMB9J4r a=extmap:1 urn:ietf:params:rtp-hdrext:sdes:mid
a=extmap:2 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=msid:47017fee-b6c1-4162-929c-a25110252400
f83006c5-a0ff-4e0a-9ed9-d3e6747be7d9
a=ice-ufrag:ETEn
a=ice-pwd:OtSK0WpNtpUjkY4+86js7ZQl a=ice-pwd:OtSK0WpNtpUjkY4+86js7ZQl
a=fingerprint:sha-256 a=fingerprint:sha-256
19:E2:1C:3B:4B:9F:81:E6:B8:5C:F4:A5:A8:D8:73:04 19:E2:1C:3B:4B:9F:81:E6:B8:5C:F4:A5:A8:D8:73:04:
:BB:05:2F:70:9F:04:A9:0E:05:E9:26:33:E8:70:88:A2 BB:05:2F:70:9F:04:A9:0E:05:E9:26:33:E8:70:88:A2
a=setup:actpass a=setup:actpass
a=dtls-id:1
a=rtcp:10101 IN IP4 203.0.113.100
a=rtcp-mux a=rtcp-mux
a=rtcp-rsize a=rtcp-rsize
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level a=candidate:1 1 udp 2113929471 203.0.113.100 10100 typ host
a=extmap:2 urn:ietf:params:rtp-hdrext:sdes:mid a=candidate:1 2 udp 2113929470 203.0.113.100 10101 typ host
a=candidate:3348148302 1 udp 2113937151 192.0.2.1 56500
typ host
a=candidate:3348148302 2 udp 2113937151 192.0.2.1 56501
typ host
a=end-of-candidates a=end-of-candidates
m=video 56502 UDP/TLS/RTP/SAVPF 100 101 m=video 10102 UDP/TLS/RTP/SAVPF 100 101
c=IN IP4 192.0.2.1 c=IN IP4 203.0.113.100
a=rtcp:56503 IN IP4 192.0.2.1
a=mid:v1 a=mid:v1
a=msid:61317484-2ed4-49d7-9eb7-1414322a7aae
f30bdb4a-5db8-49b5-bcdc-e0c9a23172e0
a=sendrecv a=sendrecv
a=rtpmap:100 VP8/90000 a=rtpmap:100 VP8/90000
a=rtpmap:101 rtx/90000 a=rtpmap:101 rtx/90000
a=fmtp:101 apt=100 a=fmtp:101 apt=100
a=ice-ufrag:BGKkWnG5GmiUpdIV a=extmap:1 urn:ietf:params:rtp-hdrext:sdes:mid
a=rtcp-fb:100 ccm fir
a=rtcp-fb:100 nack
a=rtcp-fb:100 nack pli
a=msid:47017fee-b6c1-4162-929c-a25110252400
f30bdb4a-5db8-49b5-bcdc-e0c9a23172e0
a=ice-ufrag:BGKk
a=ice-pwd:mqyWsAjvtKwTGnvhPztQ9mIf a=ice-pwd:mqyWsAjvtKwTGnvhPztQ9mIf
a=fingerprint:sha-256 a=fingerprint:sha-256
19:E2:1C:3B:4B:9F:81:E6:B8:5C:F4:A5:A8:D8:73:04 19:E2:1C:3B:4B:9F:81:E6:B8:5C:F4:A5:A8:D8:73:04:
:BB:05:2F:70:9F:04:A9:0E:05:E9:26:33:E8:70:88:A2 BB:05:2F:70:9F:04:A9:0E:05:E9:26:33:E8:70:88:A2
a=setup:actpass a=setup:actpass
a=dtls-id:1
a=rtcp:10103 IN IP4 203.0.113.100
a=rtcp-mux a=rtcp-mux
a=rtcp-rsize a=rtcp-rsize
a=extmap:3 urn:ietf:params:rtp-hdrext:sdes:mid a=candidate:1 1 udp 2113929471 203.0.113.100 10102 typ host
a=rtcp-fb:100 ccm fir a=candidate:1 2 udp 2113929470 203.0.113.100 10103 typ host
a=rtcp-fb:100 nack
a=rtcp-fb:100 nack pli
a=candidate:3348148302 1 udp 2113937151 192.0.2.1 56502
typ host
a=candidate:3348148302 2 udp 2113937151 192.0.2.1 56503
typ host
a=end-of-candidates a=end-of-candidates
The SDP for |answer-A1| looks like: The SDP for |answer-A1| looks like:
v=0 v=0
o=- 6729291447651054566 1 IN IP4 0.0.0.0 o=- 6729291447651054566 1 IN IP4 0.0.0.0
s=- s=-
t=0 0 t=0 0
a=ice-options:trickle
a=group:BUNDLE a1 v1 a=group:BUNDLE a1 v1
m=audio 20000 UDP/TLS/RTP/SAVPF 96 0 8 97 98 a=group:LS a1 v1
c=IN IP4 192.0.2.2
m=audio 10200 UDP/TLS/RTP/SAVPF 96 0 8 97 98
c=IN IP4 203.0.113.200
a=mid:a1 a=mid:a1
a=rtcp:20000 IN IP4 192.0.2.2
a=msid:PI39StLS8W7ZbQl1sJsWUXkr3Zf12fJUvzQ1
PI39StLS8W7ZbQl1sJsWUXkr3Zf12fJUvzQ1a0
a=sendrecv a=sendrecv
a=rtpmap:96 opus/48000/2 a=rtpmap:96 opus/48000/2
a=rtpmap:0 PCMU/8000 a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000 a=rtpmap:8 PCMA/8000
a=rtpmap:97 telephone-event/8000 a=rtpmap:97 telephone-event/8000
a=rtpmap:98 telephone-event/48000 a=rtpmap:98 telephone-event/48000
a=maxptime:120 a=maxptime:120
a=ice-ufrag:6sFvz2gdLkEwjZEr a=extmap:1 urn:ietf:params:rtp-hdrext:sdes:mid
a=extmap:2 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=msid:61317484-2ed4-49d7-9eb7-1414322a7aae
5a7b57b8-f043-4bd1-a45d-09d4dfa31226
a=ice-ufrag:6sFv
a=ice-pwd:cOTZKZNVlO9RSGsEGM63JXT2 a=ice-pwd:cOTZKZNVlO9RSGsEGM63JXT2
a=fingerprint:sha-256 6B:8B:F0:65:5F:78:E2:51:3B:AC:6F:F3:3F:46:1B:35 a=fingerprint:sha-256
:DC:B8:5F:64:1A:24:C2:43:F0:A1:58:D0:A1:2C:19:08 6B:8B:F0:65:5F:78:E2:51:3B:AC:6F:F3:3F:46:1B:35:
DC:B8:5F:64:1A:24:C2:43:F0:A1:58:D0:A1:2C:19:08
a=setup:active a=setup:active
a=dtls-id:1
a=rtcp-mux a=rtcp-mux
a=rtcp-rsize a=rtcp-rsize
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level a=candidate:1 1 udp 2113929471 203.0.113.200 10200 typ host
a=candidate:2299743422 1 udp 2113937151 192.0.2.2 20000
typ host
a=end-of-candidates a=end-of-candidates
m=video 20000 UDP/TLS/RTP/SAVPF 100 101 m=video 10200 UDP/TLS/RTP/SAVPF 100 101
c=IN IP4 192.0.2.2 c=IN IP4 203.0.113.200
a=rtcp 20001 IN IP4 192.0.2.2
a=mid:v1 a=mid:v1
a=msid:PI39StLS8W7ZbQl1sJsWUXkr3Zf12fJUvzQ1
PI39StLS8W7ZbQl1sJsWUXkr3Zf12fJUvzQ1v0
a=sendrecv a=sendrecv
a=rtpmap:100 VP8/90000 a=rtpmap:100 VP8/90000
a=rtpmap:101 rtx/90000 a=rtpmap:101 rtx/90000
a=fmtp:101 apt=100 a=fmtp:101 apt=100
a=fingerprint:sha-256 6B:8B:F0:65:5F:78:E2:51:3B:AC:6F:F3:3F:46:1B:35 a=extmap:1 urn:ietf:params:rtp-hdrext:sdes:mid
:DC:B8:5F:64:1A:24:C2:43:F0:A1:58:D0:A1:2C:19:08
a=setup:active
a=rtcp-mux
a=rtcp-rsize
a=rtcp-fb:100 ccm fir a=rtcp-fb:100 ccm fir
a=rtcp-fb:100 nack a=rtcp-fb:100 nack
a=rtcp-fb:100 nack pli a=rtcp-fb:100 nack pli
a=msid:61317484-2ed4-49d7-9eb7-1414322a7aae
4ea4d4a1-2fda-4511-a9cc-1b32c2e59552
7.2. Normal Examples 7.2. Detailed Example
This section shows a typical example of a session between two This section shows a more involved example of a session between two
browsers setting up an audio channel and a data channel. Trickle ICE JSEP endpoints. Trickle ICE is used in full trickle mode, with a
is used in full trickle mode with a bundle policy of max-bundle, an bundle policy of "max-bundle", an RTCP mux policy of "require", and a
RTCP mux policy of require, and a single TURN server. Later, two single TURN server. Initially, both Alice and Bob establish an audio
video flows, one for the presenter and one for screen sharing, are channel and a data channel. Later, Bob adds two video flows, one for
added to the session. This example shows Alice's browser initiating his video feed, and one for screensharing, both supporting FEC, and
the session to Bob's browser. The messages from Alice's JS to Bob's with the video feed configured for simulcast. Alice accepts these
JS are assumed to flow over some signaling protocol via a web server. video flows, but does not add video flows of her own, so they are
handled as recvonly. Alice also specifies a maximum video decoder
resolution.
// set up local media state // set up local media state
AliceJS->AliceUA: create new PeerConnection AliceJS->AliceUA: create new PeerConnection
AliceJS->AliceUA: addTrack with an audio track AliceJS->AliceUA: addTrack with an audio track
AliceJS->AliceUA: createDataChannel to get data channel AliceJS->AliceUA: createDataChannel to get data channel
AliceJS->AliceUA: createOffer to get |offer-B1| AliceJS->AliceUA: createOffer to get |offer-B1|
AliceJS->AliceUA: setLocalDescription with |offer-B1| AliceJS->AliceUA: setLocalDescription with |offer-B1|
// |offer-B1| is sent over signaling protocol to Bob // |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
BobJS->BobUA: setRemoteDescription with |offer-B1| BobJS->BobUA: setRemoteDescription with |offer-B1|
BobUA->BobJS: onaddstream with audio track from Alice BobUA->BobJS: ontrack with audio track from Alice
// candidates are sent to Bob // candidates are sent to Bob
AliceUA->AliceJS: onicecandidate event with |candidate-B1| (host) AliceUA->AliceJS: onicecandidate (host) |offer-B1-candidate-1|
AliceJS->WebServer: signaling with |candidate-B1| AliceJS->WebServer: signaling with |offer-B1-candidate-1|
AliceUA->AliceJS: onicecandidate event with |candidate-B2| (srflx) AliceUA->AliceJS: onicecandidate (srflx) |offer-B1-candidate-2|
AliceJS->WebServer: signaling with |candidate-B2| AliceJS->WebServer: signaling with |offer-B1-candidate-2|
AliceUA->AliceJS: onicecandidate (relay) |offer-B1-candidate-3|
AliceJS->WebServer: signaling with |offer-B1-candidate-3|
WebServer->BobJS: signaling with |candidate-B1| WebServer->BobJS: signaling with |offer-B1-candidate-1|
BobJS->BobUA: addIceCandidate with |candidate-B1| BobJS->BobUA: addIceCandidate with |offer-B1-candidate-1|
WebServer->BobJS: signaling with |candidate-B2| WebServer->BobJS: signaling with |offer-B1-candidate-2|
BobJS->BobUA: addIceCandidate with |candidate-B2| BobJS->BobUA: addIceCandidate with |offer-B1-candidate-2|
WebServer->BobJS: signaling with |offer-B1-candidate-3|
BobJS->BobUA: addIceCandidate with |offer-B1-candidate-3|
// Bob accepts call // Bob accepts call
BobJS->BobUA: addTrack with local audio 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: ontrack event with audio track from Bob
// candidates are sent to Alice // candidates are sent to Alice
BobUA->BobJS: onicecandidate event with |candidate-B3| (host) BobUA->BobJS: onicecandidate (host) |answer-B1-candidate-1|
BobJS->WebServer: signaling with |candidate-B3| BobJS->WebServer: signaling with |answer-B1-candidate-1|
BobUA->BobJS: onicecandidate event with |candidate-B4| (srflx) BobUA->BobJS: onicecandidate (srflx) |answer-B1-candidate-2|
BobJS->WebServer: signaling with |candidate-B4| BobJS->WebServer: signaling with |answer-B1-candidate-2|
BobUA->BobJS: onicecandidate (relay) |answer-B1-candidate-3|
BobJS->WebServer: signaling with |answer-B1-candidate-3|
WebServer->AliceJS: signaling with |candidate-B3| WebServer->AliceJS: signaling with |answer-B1-candidate-1|
AliceJS->AliceUA: addIceCandidate with |candidate-B3| AliceJS->AliceUA: addIceCandidate with |answer-B1-candidate-1|
WebServer->AliceJS: signaling with |candidate-B4| WebServer->AliceJS: signaling with |answer-B1-candidate-2|
AliceJS->AliceUA: addIceCandidate with |candidate-B4| AliceJS->AliceUA: addIceCandidate with |answer-B1-candidate-2|
WebServer->AliceJS: signaling with |answer-B1-candidate-3|
AliceJS->AliceUA: addIceCandidate with |answer-B1-candidate-3|
// data channel opens // data channel opens
BobUA->BobJS: ondatachannel event BobUA->BobJS: ondatachannel event
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 endpoints
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: addTrack with first video stream BobJS->BobUA: addTrack with first video stream
BobJS->BobUA: addTrack 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: ontrack event with first video track
AliceUA->AliceJS: onaddstream event with second video stream AliceUA->AliceJS: ontrack event with second video track
AliceJS->AliceUA: createAnswer to get |answer-B2| AliceJS->AliceUA: createAnswer to get |answer-B2|
AliceJS->AliceUA: setLocalDescription with |answer-B2| AliceJS->AliceUA: setLocalDescription with |answer-B2|
// |answer-B2| is sent over signaling protocol to Bob // |answer-B2| is sent over signaling protocol to Bob
AliceJS->WebServer: signaling with |answer-B2| AliceJS->WebServer: signaling with |answer-B2|
WebServer->BobJS: signaling with |answer-B2| WebServer->BobJS: signaling with |answer-B2|
BobJS->BobUA: setRemoteDescription with |answer-B2| BobJS->BobUA: setRemoteDescription with |answer-B2|
// media is flowing between browsers // media is flowing between endpoints
BobUA->AliceUA: audio+video+data sent from Bob to Alice BobUA->AliceUA: audio+video+data sent from Bob to Alice
AliceUA->BobUA: audio+video+data sent from Alice to Bob AliceUA->BobUA: audio+video+data sent from Alice to Bob
The SDP for |offer-B1| looks like: The SDP for |offer-B1| looks like:
v=0 v=0
o=- 4962303333179871723 1 IN IP4 0.0.0.0 o=- 4962303333179871723 1 IN IP4 0.0.0.0
s=- s=-
t=0 0 t=0 0
a=group:BUNDLE a1 d1
a=ice-options:trickle a=ice-options:trickle
a=group:BUNDLE a1 d1
m=audio 9 UDP/TLS/RTP/SAVPF 96 0 8 97 98 m=audio 9 UDP/TLS/RTP/SAVPF 96 0 8 97 98
c=IN IP4 0.0.0.0 c=IN IP4 0.0.0.0
a=rtcp:9 IN IP4 0.0.0.0
a=mid:a1 a=mid:a1
a=msid:57017fee-b6c1-4162-929c-a25110252400
e83006c5-a0ff-4e0a-9ed9-d3e6747be7d9
a=sendrecv a=sendrecv
a=rtpmap:96 opus/48000/2 a=rtpmap:96 opus/48000/2
a=rtpmap:0 PCMU/8000 a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000 a=rtpmap:8 PCMA/8000
a=rtpmap:97 telephone-event/8000 a=rtpmap:97 telephone-event/8000
a=rtpmap:98 telephone-event/48000 a=rtpmap:98 telephone-event/48000
a=maxptime:120 a=maxptime:120
a=ice-ufrag:ATEn1v9DoTMB9J4r a=extmap:1 urn:ietf:params:rtp-hdrext:sdes:mid
a=extmap:2 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=msid:57017fee-b6c1-4162-929c-a25110252400
e83006c5-a0ff-4e0a-9ed9-d3e6747be7d9
a=ice-ufrag:ATEn
a=ice-pwd:AtSK0WpNtpUjkY4+86js7ZQl a=ice-pwd:AtSK0WpNtpUjkY4+86js7ZQl
a=fingerprint:sha-256 a=fingerprint:sha-256
19:E2:1C:3B:4B:9F:81:E6:B8:5C:F4:A5:A8:D8:73:04 29:E2:1C:3B:4B:9F:81:E6:B8:5C:F4:A5:A8:D8:73:04:
:BB:05:2F:70:9F:04:A9:0E:05:E9:26:33:E8:70:88:A2 BB:05:2F:70:9F:04:A9:0E:05:E9:26:33:E8:70:88:A2
a=setup:actpass a=setup:actpass
a=dtls-id:1
a=rtcp-mux a=rtcp-mux
a=rtcp-mux-only
a=rtcp-rsize a=rtcp-rsize
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=extmap:2 urn:ietf:params:rtp-hdrext:sdes:mid
m=application 0 UDP/DTLS/SCTP webrtc-datachannel m=application 0 UDP/DTLS/SCTP webrtc-datachannel
c=IN IP4 0.0.0.0 c=IN IP4 0.0.0.0
a=bundle-only
a=mid:d1 a=mid:d1
a=fmtp:webrtc-datachannel max-message-size=65536 a=sctp-port:5000
a=sctp-port 5000 a=max-message-size:65536
a=fingerprint:sha-256 19:E2:1C:3B:4B:9F:81:E6:B8:5C:F4:A5:A8:D8:73:04 a=bundle-only
:BB:05:2F:70:9F:04:A9:0E:05:E9:26:33:E8:70:88:A2
a=setup:actpass
The SDP for |candidate-B1| looks like: |offer-B1-candidate-1| looks like:
candidate:109270923 1 udp 2122194687 192.168.1.2 51556 typ host ufrag ATEn
index 0
mid a1
attr candidate:1 1 udp 2113929471 203.0.113.100 10100 typ host
|offer-B1-candidate-2| looks like:
The SDP for |candidate-B2| looks like: ufrag ATEn
index 0
mid a1
attr candidate:1 1 udp 1845494015 198.51.100.100 11100 typ srflx
raddr 203.0.113.100 rport 10100
candidate:4036177503 1 udp 1685987071 11.22.33.44 52546 typ srflx |offer-B1-candidate-3| looks like:
raddr 192.168.1.2 rport 51556
ufrag ATEn
index 0
mid a1
attr candidate:1 1 udp 255 192.0.2.100 12100 typ relay
raddr 198.51.100.100 rport 11100
The SDP for |answer-B1| looks like: The SDP for |answer-B1| looks like:
v=0 v=0
o=- 7729291447651054566 1 IN IP4 0.0.0.0 o=- 7729291447651054566 1 IN IP4 0.0.0.0
s=- s=-
t=0 0 t=0 0
a=group:BUNDLE a1 d1
a=ice-options:trickle a=ice-options:trickle
a=group:BUNDLE a1 d1
m=audio 9 UDP/TLS/RTP/SAVPF 96 0 8 97 98 m=audio 9 UDP/TLS/RTP/SAVPF 96 0 8 97 98
c=IN IP4 0.0.0.0 c=IN IP4 0.0.0.0
a=rtcp:9 IN IP4 0.0.0.0
a=mid:a1 a=mid:a1
a=msid:QI39StLS8W7ZbQl1sJsWUXkr3Zf12fJUvzQ1
QI39StLS8W7ZbQl1sJsWUXkr3Zf12fJUvzQ1a0
a=sendrecv a=sendrecv
a=rtpmap:96 opus/48000/2 a=rtpmap:96 opus/48000/2
a=rtpmap:0 PCMU/8000 a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000 a=rtpmap:8 PCMA/8000
a=rtpmap:97 telephone-event/8000 a=rtpmap:97 telephone-event/8000
a=rtpmap:98 telephone-event/48000 a=rtpmap:98 telephone-event/48000
a=maxptime:120 a=maxptime:120
a=ice-ufrag:7sFvz2gdLkEwjZEr a=extmap:1 urn:ietf:params:rtp-hdrext:sdes:mid
a=extmap:2 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=msid:71317484-2ed4-49d7-9eb7-1414322a7aae
6a7b57b8-f043-4bd1-a45d-09d4dfa31226
a=ice-ufrag:7sFv
a=ice-pwd:dOTZKZNVlO9RSGsEGM63JXT2 a=ice-pwd:dOTZKZNVlO9RSGsEGM63JXT2
a=fingerprint:sha-256 6B:8B:F0:65:5F:78:E2:51:3B:AC:6F:F3:3F:46:1B:35 a=fingerprint:sha-256
:DC:B8:5F:64:1A:24:C2:43:F0:A1:58:D0:A1:2C:19:08 7B:8B:F0:65:5F:78:E2:51:3B:AC:6F:F3:3F:46:1B:35:
DC:B8:5F:64:1A:24:C2:43:F0:A1:58:D0:A1:2C:19:08
a=setup:active a=setup:active
a=dtls-id:1
a=rtcp-mux a=rtcp-mux
a=rtcp-mux-only
a=rtcp-rsize a=rtcp-rsize
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=extmap:2 urn:ietf:params:rtp-hdrext:sdes:mid
m=application 9 UDP/DTLS/SCTP webrtc-datachannel m=application 9 UDP/DTLS/SCTP webrtc-datachannel
c=IN IP4 0.0.0.0 c=IN IP4 0.0.0.0
a=mid:d1 a=mid:d1
a=fmtp:webrtc-datachannel max-message-size=65536 a=sctp-port:5000
a=sctp-port 5000 a=max-message-size:65536
a=fingerprint:sha-256 6B:8B:F0:65:5F:78:E2:51:3B:AC:6F:F3:3F:46:1B:35
:DC:B8:5F:64:1A:24:C2:43:F0:A1:58:D0:A1:2C:19:08
a=setup:active
The SDP for |candidate-B3| looks like: |answer-B1-candidate-1| looks like:
candidate:109270924 1 udp 2122194687 192.168.2.3 61665 typ host ufrag 7sFv
index 0
mid a1
attr candidate:1 1 udp 2113929471 203.0.113.200 10200 typ host
|answer-B1-candidate-2| looks like:
The SDP for |candidate-B4| looks like: ufrag 7sFv
index 0
mid a1
attr candidate:1 1 udp 1845494015 198.51.100.200 11200 typ srflx
raddr 203.0.113.200 rport 10200
candidate:4036177504 1 udp 1685987071 55.66.77.88 64532 typ srflx |answer-B1-candidate-3| looks like:
raddr 192.168.2.3 rport 61665
The SDP for |offer-B2| looks like: (note the increment of the version ufrag 7sFv
number in the o= line, and the c= and a=rtcp lines, which indicate index 0
the local candidate that was selected) mid a1
attr candidate:1 1 udp 255 192.0.2.200 12200 typ relay
raddr 198.51.100.200 rport 11200
The SDP for |offer-B2| is shown below. In addition to the new m=
sections for video, both of which are offering FEC, and one of which
is offering simulcast, note the increment of the version number in
the o= line, changes to the c= line, indicating the local candidate
that was selected, and the inclusion of gathered candidates as
a=candidate lines.
v=0 v=0
o=- 7729291447651054566 2 IN IP4 0.0.0.0 o=- 7729291447651054566 2 IN IP4 0.0.0.0
s=- s=-
t=0 0 t=0 0
a=group:BUNDLE a1 d1 v1 v2
a=ice-options:trickle a=ice-options:trickle
m=audio 64532 UDP/TLS/RTP/SAVPF 96 0 8 97 98 a=group:BUNDLE a1 d1 v1 v2
c=IN IP4 55.66.77.88 a=group:LS a1 v1
a=rtcp:64532 IN IP4 55.66.77.88
m=audio 12200 UDP/TLS/RTP/SAVPF 96 0 8 97 98
c=IN IP4 192.0.2.200
a=mid:a1 a=mid:a1
a=msid:QI39StLS8W7ZbQl1sJsWUXkr3Zf12fJUvzQ1
QI39StLS8W7ZbQl1sJsWUXkr3Zf12fJUvzQ1a0
a=sendrecv a=sendrecv
a=rtpmap:96 opus/48000/2 a=rtpmap:96 opus/48000/2
a=rtpmap:0 PCMU/8000 a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000 a=rtpmap:8 PCMA/8000
a=rtpmap:97 telephone-event/8000 a=rtpmap:97 telephone-event/8000
a=rtpmap:98 telephone-event/48000 a=rtpmap:98 telephone-event/48000
a=maxptime:120 a=maxptime:120
a=ice-ufrag:7sFvz2gdLkEwjZEr a=extmap:1 urn:ietf:params:rtp-hdrext:sdes:mid
a=extmap:2 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=msid:71317484-2ed4-49d7-9eb7-1414322a7aae
6a7b57b8-f043-4bd1-a45d-09d4dfa31226
a=ice-ufrag:7sFv
a=ice-pwd:dOTZKZNVlO9RSGsEGM63JXT2 a=ice-pwd:dOTZKZNVlO9RSGsEGM63JXT2
a=fingerprint:sha-256 6B:8B:F0:65:5F:78:E2:51:3B:AC:6F:F3:3F:46:1B:35 a=fingerprint:sha-256
:DC:B8:5F:64:1A:24:C2:43:F0:A1:58:D0:A1:2C:19:08 7B:8B:F0:65:5F:78:E2:51:3B:AC:6F:F3:3F:46:1B:35:
DC:B8:5F:64:1A:24:C2:43:F0:A1:58:D0:A1:2C:19:08
a=setup:actpass a=setup:actpass
a=dtls-id:1
a=rtcp-mux a=rtcp-mux
a=rtcp-mux-only
a=rtcp-rsize a=rtcp-rsize
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level a=candidate:1 1 udp 2113929471 203.0.113.200 10200 typ host
a=extmap:2 urn:ietf:params:rtp-hdrext:sdes:mid a=candidate:1 1 udp 1845494015 198.51.100.200 11200 typ srflx
a=candidate:109270924 1 udp 2122194687 192.168.2.3 61665 typ host raddr 203.0.113.200 rport 10200
a=candidate:4036177504 1 udp 1685987071 55.66.77.88 64532 typ srflx a=candidate:1 1 udp 255 192.0.2.200 12200 typ relay
raddr 192.168.2.3 rport 61665 raddr 198.51.100.200 rport 11200
a=candidate:3671762467 1 udp 41819903 66.77.88.99 50416 typ relay
raddr 55.66.77.88 rport 64532
a=end-of-candidates a=end-of-candidates
m=application 64532 UDP/DTLS/SCTP webrtc-datachannel m=application 12200 UDP/DTLS/SCTP webrtc-datachannel
c=IN IP4 55.66.77.88 c=IN IP4 192.0.2.200
a=mid:d1 a=mid:d1
a=fmtp:webrtc-datachannel max-message-size=65536 a=sctp-port:5000
a=sctp-port 5000 a=max-message-size:65536
a=ice-ufrag:7sFvz2gdLkEwjZEr
a=ice-pwd:dOTZKZNVlO9RSGsEGM63JXT2
a=fingerprint:sha-256 6B:8B:F0:65:5F:78:E2:51:3B:AC:6F:F3:3F:46:1B:35
:DC:B8:5F:64:1A:24:C2:43:F0:A1:58:D0:A1:2C:19:08
a=setup:actpass
a=candidate:109270924 1 udp 2122194687 192.168.2.3 61665 typ host
a=candidate:4036177504 1 udp 1685987071 55.66.77.88 64532 typ srflx
raddr 192.168.2.3 rport 61665
a=candidate:3671762467 1 udp 41819903 66.77.88.99 50416 typ relay
raddr 55.66.77.88 rport 64532
a=end-of-candidates
m=video 0 UDP/TLS/RTP/SAVPF 100 101 m=video 12200 UDP/TLS/RTP/SAVPF 100 101 102
c=IN IP4 55.66.77.88 c=IN IP4 192.0.2.200
a=bundle-only
a=rtcp:64532 IN IP4 55.66.77.88
a=mid:v1 a=mid:v1
a=msid:61317484-2ed4-49d7-9eb7-1414322a7aae
f30bdb4a-5db8-49b5-bcdc-e0c9a23172e0
a=sendrecv a=sendrecv
a=rtpmap:100 VP8/90000 a=rtpmap:100 VP8/90000
a=rtpmap:101 rtx/90000 a=rtpmap:101 rtx/90000
a=fmtp:101 apt=100 a=fmtp:101 apt=100
a=fingerprint:sha-256 a=rtpmap:102 flexfec/90000
19:E2:1C:3B:4B:9F:81:E6:B8:5C:F4:A5:A8:D8:73:04 a=extmap:1 urn:ietf:params:rtp-hdrext:sdes:mid
:BB:05:2F:70:9F:04:A9:0E:05:E9:26:33:E8:70:88:A2
a=setup:actpass
a=rtcp-mux
a=rtcp-rsize
a=extmap:2 urn:ietf:params:rtp-hdrext:sdes:mid
a=rtcp-fb:100 ccm fir a=rtcp-fb:100 ccm fir
a=rtcp-fb:100 nack a=rtcp-fb:100 nack
a=rtcp-fb:100 nack pli a=rtcp-fb:100 nack pli
m=video 0 UDP/TLS/RTP/SAVPF 100 101
c=IN IP4 55.66.77.88
a=bundle-only
a=rtcp:64532 IN IP4 55.66.77.88
a=mid:v1
a=msid:71317484-2ed4-49d7-9eb7-1414322a7aae a=msid:71317484-2ed4-49d7-9eb7-1414322a7aae
f30bdb4a-5db8-49b5-bcdc-e0c9a23172e0 5ea4d4a1-2fda-4511-a9cc-1b32c2e59552
a=rid:1 send
a=rid:2 send
a=rid:3 send
a=simulcast:send 1;2;3
m=video 12200 UDP/TLS/RTP/SAVPF 100 101 102
c=IN IP4 192.0.2.200
a=mid:v2
a=sendrecv a=sendrecv
a=rtpmap:100 VP8/90000 a=rtpmap:100 VP8/90000
a=rtpmap:101 rtx/90000 a=rtpmap:101 rtx/90000
a=fmtp:101 apt=100 a=fmtp:101 apt=100
a=fingerprint:sha-256 a=rtpmap:102 flexfec/90000
19:E2:1C:3B:4B:9F:81:E6:B8:5C:F4:A5:A8:D8:73:04 a=extmap:1 urn:ietf:params:rtp-hdrext:sdes:mid
:BB:05:2F:70:9F:04:A9:0E:05:E9:26:33:E8:70:88:A2
a=setup:actpass
a=rtcp-mux
a=rtcp-rsize
a=extmap:2 urn:ietf:params:rtp-hdrext:sdes:mid
a=rtcp-fb:100 ccm fir a=rtcp-fb:100 ccm fir
a=rtcp-fb:100 nack a=rtcp-fb:100 nack
a=rtcp-fb:100 nack pli a=rtcp-fb:100 nack pli
a=msid:81317484-2ed4-49d7-9eb7-1414322a7aae
6ea4d4a1-2fda-4511-a9cc-1b32c2e59552
The SDP for |answer-B2| looks like: (note the use of setup:passive to The SDP for |answer-B2| is shown below. In addition to the
maintain the existing DTLS roles, and the use of a=recvonly to acceptance of the video m= sections, the use of a=recvonly to
indicate that the video streams are one-way) indicate one-way video, and the use of a=imageattr to limit the
received resolution, note the use of setup:passive to maintain the
existing DTLS roles.
v=0 v=0
o=- 4962303333179871723 2 IN IP4 0.0.0.0 o=- 4962303333179871723 2 IN IP4 0.0.0.0
s=- s=-
t=0 0 t=0 0
a=group:BUNDLE a1 d1 v1 v2
a=ice-options:trickle a=ice-options:trickle
m=audio 52546 UDP/TLS/RTP/SAVPF 96 0 8 97 98 a=group:BUNDLE a1 d1 v1 v2
c=IN IP4 11.22.33.44 a=group:LS a1 v1
a=rtcp:52546 IN IP4 11.22.33.44
m=audio 12100 UDP/TLS/RTP/SAVPF 96 0 8 97 98
c=IN IP4 192.0.2.100
a=mid:a1 a=mid:a1
a=msid:57017fee-b6c1-4162-929c-a25110252400
e83006c5-a0ff-4e0a-9ed9-d3e6747be7d9
a=sendrecv a=sendrecv
a=rtpmap:96 opus/48000/2 a=rtpmap:96 opus/48000/2
a=rtpmap:0 PCMU/8000 a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000 a=rtpmap:8 PCMA/8000
a=rtpmap:97 telephone-event/8000 a=rtpmap:97 telephone-event/8000
a=rtpmap:98 telephone-event/48000 a=rtpmap:98 telephone-event/48000
a=maxptime:120 a=maxptime:120
a=ice-ufrag:ATEn1v9DoTMB9J4r a=extmap:1 urn:ietf:params:rtp-hdrext:sdes:mid
a=extmap:2 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=msid:57017fee-b6c1-4162-929c-a25110252400
e83006c5-a0ff-4e0a-9ed9-d3e6747be7d9
a=ice-ufrag:ATEn
a=ice-pwd:AtSK0WpNtpUjkY4+86js7ZQl a=ice-pwd:AtSK0WpNtpUjkY4+86js7ZQl
a=fingerprint:sha-256 a=fingerprint:sha-256
19:E2:1C:3B:4B:9F:81:E6:B8:5C:F4:A5:A8:D8:73:04 29:E2:1C:3B:4B:9F:81:E6:B8:5C:F4:A5:A8:D8:73:04:
:BB:05:2F:70:9F:04:A9:0E:05:E9:26:33:E8:70:88:A2 BB:05:2F:70:9F:04:A9:0E:05:E9:26:33:E8:70:88:A2
a=setup:passive a=setup:passive
a=dtls-id:1
a=rtcp-mux a=rtcp-mux
a=rtcp-mux-only
a=rtcp-rsize a=rtcp-rsize
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level a=candidate:1 1 udp 2113929471 203.0.113.100 10100 typ host
a=extmap:2 urn:ietf:params:rtp-hdrext:sdes:mid a=candidate:1 1 udp 1845494015 198.51.100.100 11100 typ srflx
a=candidate:109270923 1 udp 2122194687 192.168.1.2 51556 typ host raddr 203.0.113.100 rport 10100
a=candidate:4036177503 1 udp 1685987071 11.22.33.44 52546 typ srflx a=candidate:1 1 udp 255 192.0.2.100 12100 typ relay
raddr 192.168.1.2 rport 51556 raddr 198.51.100.100 rport 11100
a=candidate:3671762466 1 udp 41819903 22.33.44.55 61405 typ relay
raddr 11.22.33.44 rport 52546
a=end-of-candidates a=end-of-candidates
m=application 52546 UDP/DTLS/SCTP webrtc-datachannel m=application 12100 UDP/DTLS/SCTP webrtc-datachannel
c=IN IP4 11.22.33.44 c=IN IP4 192.0.2.100
a=mid:d1 a=mid:d1
a=fmtp:webrtc-datachannel max-message-size=65536 a=sctp-port:5000
a=sctp-port 5000 a=max-message-size:65536
a=fingerprint:sha-256 19:E2:1C:3B:4B:9F:81:E6:B8:5C:F4:A5:A8:D8:73:04
:BB:05:2F:70:9F:04:A9:0E:05:E9:26:33:E8:70:88:A2
a=setup:passive
m=video 52546 UDP/TLS/RTP/SAVPF 100 101 m=video 12100 UDP/TLS/RTP/SAVPF 100 101
c=IN IP4 11.22.33.44 c=IN IP4 192.0.2.100
a=rtcp:52546 IN IP4 11.22.33.44
a=mid:v1 a=mid:v1
a=recvonly a=recvonly
a=rtpmap:100 VP8/90000 a=rtpmap:100 VP8/90000
a=rtpmap:101 rtx/90000 a=rtpmap:101 rtx/90000
a=fmtp:101 apt=100 a=fmtp:101 apt=100
a=imageattr:100 recv [x=[48:1920],y=[48:1080],q=1.0]
a=extmap:1 urn:ietf:params:rtp-hdrext:sdes:mid
a=rtcp-fb:100 ccm fir
a=rtcp-fb:100 nack
a=rtcp-fb:100 nack pli
m=video 12100 UDP/TLS/RTP/SAVPF 100 101
c=IN IP4 192.0.2.100
a=mid:v2
a=recvonly
a=rtpmap:100 VP8/90000
a=rtpmap:101 rtx/90000
a=fmtp:101 apt=100
a=imageattr:100 recv [x=[48:1920],y=[48:1080],q=1.0]
a=extmap:1 urn:ietf:params:rtp-hdrext:sdes:mid
a=rtcp-fb:100 ccm fir
a=rtcp-fb:100 nack
a=rtcp-fb:100 nack pli
7.3. Early Transport Warmup Example
This example demonstrates the early warmup technique described in
Section 4.1.8.1. Here, Alice's endpoint sends an offer to Bob's
endpoint to start an audio/video call. Bob immediately responds with
an answer that accepts the audio/video m= sections, but marks them as
sendonly (from his perspective), meaning that Alice will not yet send
media. This allows the JSEP implementation to start negotiating ICE
and DTLS immediately. Bob's endpoint then prompts him to answer the
call, and when he does, his endpoint sends a second offer which
enables the audio and video m= sections, and thereby bidirectional
media transmission. The advantage of such a flow is that as soon as
the first answer is received, the implementation can proceed with ICE
and DTLS negotiation and establish the session transport. If the
transport setup completes before the second offer is sent, then media
can be transmitted immediately by the callee immediately upon
answering the call, minimizing perceived post-dial-delay. The second
offer/answer exchange can also change the preferred codecs or other
session parameters.
This example also makes use of the "relay" ICE candidate policy
described in Section 3.5.3 to minimize the ICE gathering and checking
needed.
// set up local media state
AliceJS->AliceUA: create new PeerConnection with "relay" ICE policy
AliceJS->AliceUA: addTrack with two tracks: audio and video
AliceJS->AliceUA: createOffer to get |offer-C1|
AliceJS->AliceUA: setLocalDescription with |offer-C1|
// |offer-C1| is sent over signaling protocol to Bob
AliceJS->WebServer: signaling with |offer-C1|
WebServer->BobJS: signaling with |offer-C1|
// |offer-C1| arrives at Bob
BobJS->BobUA: create new PeerConnection with "relay" ICE policy
BobJS->BobUA: setRemoteDescription with |offer-C1|
BobUA->BobJS: ontrack events for audio and video
// a relay candidate is sent to Bob
AliceUA->AliceJS: onicecandidate (relay) |offer-C1-candidate-1|
AliceJS->WebServer: signaling with |offer-C1-candidate-1|
WebServer->BobJS: signaling with |offer-C1-candidate-1|
BobJS->BobUA: addIceCandidate with |offer-C1-candidate-1|
// Bob prepares an early answer to warm up the transport
BobJS->BobUA: addTransceiver with null audio and video tracks
BobJS->BobUA: transceiver.setDirection(sendonly) for both
BobJS->BobUA: createAnswer
BobJS->BobUA: setLocalDescription with answer
// |answer-C1| is sent over signaling protocol to Alice
BobJS->WebServer: signaling with |answer-C1|
WebServer->AliceJS: signaling with |answer-C1|
// |answer-C1| (sendonly) arrives at Alice
AliceJS->AliceUA: setRemoteDescription with |answer-C1|
AliceUA->AliceJS: ontrack events for audio and video
// a relay candidate is sent to Alice
BobUA->BobJS: onicecandidate (relay) |answer-B1-candidate-1|
BobJS->WebServer: signaling with |answer-B1-candidate-1|
WebServer->AliceJS: signaling with |answer-B1-candidate-1|
AliceJS->AliceUA: addIceCandidate with |answer-B1-candidate-1|
// ICE and DTLS establish while call is ringing
// Bob accepts call, starts media, and sends a new offer
BobJS->BobUA: transceiver.setTrack with audio and video tracks
BobUA->AliceUA: media sent from Bob to Alice
BobJS->BobUA: transceiver.setDirection(sendrecv) for both
transceivers
BobJS->BobUA: createOffer
BobJS->BobUA: setLocalDescription with offer
// |offer-C2| is sent over signaling protocol to Alice
BobJS->WebServer: signaling with |offer-C2|
WebServer->AliceJS: signaling with |offer-C2|
// |offer-C2| (sendrecv) arrives at Alice
AliceJS->AliceUA: setRemoteDescription with |offer-C2|
AliceJS->AliceUA: createAnswer
AliceJS->AliceUA: setLocalDescription with |answer-C2|
AliceUA->BobUA: media sent from Alice to Bob
// |answer-C2| is sent over signaling protocol to Bob
AliceJS->WebServer: signaling with |answer-C2|
WebServer->BobJS: signaling with |answer-C2|
BobJS->BobUA: setRemoteDescription with |answer-C2|
The SDP for |offer-C1| looks like:
v=0
o=- 1070771854436052752 1 IN IP4 0.0.0.0
s=-
t=0 0
a=ice-options:trickle
a=group:BUNDLE a1 v1
a=group:LS a1 v1
m=audio 9 UDP/TLS/RTP/SAVPF 96 0 8 97 98
c=IN IP4 0.0.0.0
a=mid:a1
a=sendrecv
a=rtpmap:96 opus/48000/2
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:97 telephone-event/8000
a=rtpmap:98 telephone-event/48000
a=maxptime:120
a=extmap:1 urn:ietf:params:rtp-hdrext:sdes:mid
a=extmap:2 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=msid:bbce3ba6-abfc-ac63-d00a-e15b286f8fce
e80098db-7159-3c06-229a-00df2a9b3dbc
a=ice-ufrag:4ZcD
a=ice-pwd:ZaaG6OG7tCn4J/lehAGz+HHD
a=fingerprint:sha-256 a=fingerprint:sha-256
19:E2:1C:3B:4B:9F:81:E6:B8:5C:F4:A5:A8:D8:73:04 C4:68:F8:77:6A:44:F1:98:6D:7C:9F:47:EB:E3:34:A4:
:BB:05:2F:70:9F:04:A9:0E:05:E9:26:33:E8:70:88:A2 0A:AA:2D:49:08:28:70:2E:1F:AE:18:7D:4E:3E:66:BF
a=setup:passive a=setup:actpass
a=dtls-id:1
a=rtcp-mux a=rtcp-mux
a=rtcp-mux-only
a=rtcp-rsize a=rtcp-rsize
a=extmap:2 urn:ietf:params:rtp-hdrext:sdes:mid
m=video 0 UDP/TLS/RTP/SAVPF 100 101
c=IN IP4 0.0.0.0
a=mid:v1
a=sendrecv
a=rtpmap:100 VP8/90000
a=rtpmap:101 rtx/90000
a=fmtp:101 apt=100
a=extmap:1 urn:ietf:params:rtp-hdrext:sdes:mid
a=rtcp-fb:100 ccm fir a=rtcp-fb:100 ccm fir
a=rtcp-fb:100 nack a=rtcp-fb:100 nack
a=rtcp-fb:100 nack pli a=rtcp-fb:100 nack pli
a=msid:bbce3ba6-abfc-ac63-d00a-e15b286f8fce
ac701365-eb06-42df-cc93-7f22bc308789
a=bundle-only
|offer-C1-candidate-1| looks like:
m=video 52546 UDP/TLS/RTP/SAVPF 100 101 ufrag 4ZcD
c=IN IP4 11.22.33.44 index 0
a=rtcp:52546 IN IP4 11.22.33.44 mid a1
a=mid:v2 attr candidate:1 1 udp 255 192.0.2.100 12100 typ relay
a=recvonly raddr 0.0.0.0 rport 0
The SDP for |answer-C1| looks like:
v=0
o=- 6386516489780559513 1 IN IP4 0.0.0.0
s=-
t=0 0
a=ice-options:trickle
a=group:BUNDLE a1 v1
a=group:LS a1 v1
m=audio 9 UDP/TLS/RTP/SAVPF 96 0 8 97 98
c=IN IP4 0.0.0.0
a=mid:a1
a=sendonly
a=rtpmap:96 opus/48000/2
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:97 telephone-event/8000
a=rtpmap:98 telephone-event/48000
a=maxptime:120
a=extmap:1 urn:ietf:params:rtp-hdrext:sdes:mid
a=extmap:2 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=msid:751f239e-4ae0-c549-aa3d-890de772998b
04b5a445-82cc-c9e8-9ffe-c24d0ef4b0ff
a=ice-ufrag:TpaA
a=ice-pwd:t2Ouhc67y8JcCaYZxUUTgKw/
a=fingerprint:sha-256
A2:F3:A5:6D:4C:8C:1E:B2:62:10:4A:F6:70:61:C4:FC:
3C:E0:01:D6:F3:24:80:74:DA:7C:3E:50:18:7B:CE:4D
a=setup:active
a=dtls-id:1
a=rtcp-mux
a=rtcp-mux-only
a=rtcp-rsize
m=video 9 UDP/TLS/RTP/SAVPF 100 101
c=IN IP4 0.0.0.0
a=mid:v1
a=sendonly
a=rtpmap:100 VP8/90000 a=rtpmap:100 VP8/90000
a=rtpmap:101 rtx/90000 a=rtpmap:101 rtx/90000
a=fmtp:101 apt=100 a=fmtp:101 apt=100
a=extmap:1 urn:ietf:params:rtp-hdrext:sdes:mid
a=rtcp-fb:100 ccm fir
a=rtcp-fb:100 nack
a=rtcp-fb:100 nack pli
a=msid:751f239e-4ae0-c549-aa3d-890de772998b
39292672-c102-d075-f580-5826f31ca958
|answer-C1-candidate-1| looks like:
ufrag TpaA
index 0
mid a1
attr candidate:1 1 udp 255 192.0.2.200 12200 typ relay
raddr 0.0.0.0 rport 0
The SDP for |offer-C2| looks like:
v=0
o=- 6386516489780559513 2 IN IP4 0.0.0.0
s=-
t=0 0
a=ice-options:trickle
a=group:BUNDLE a1 v1
a=group:LS a1 v1
m=audio 12200 UDP/TLS/RTP/SAVPF 96 0 8 97 98
c=IN IP4 192.0.2.200
a=mid:a1
a=sendrecv
a=rtpmap:96 opus/48000/2
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:97 telephone-event/8000
a=rtpmap:98 telephone-event/48000
a=maxptime:120
a=extmap:1 urn:ietf:params:rtp-hdrext:sdes:mid
a=extmap:2 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=msid:751f239e-4ae0-c549-aa3d-890de772998b
04b5a445-82cc-c9e8-9ffe-c24d0ef4b0ff
a=ice-ufrag:TpaA
a=ice-pwd:t2Ouhc67y8JcCaYZxUUTgKw/
a=fingerprint:sha-256 a=fingerprint:sha-256
19:E2:1C:3B:4B:9F:81:E6:B8:5C:F4:A5:A8:D8:73:04 A2:F3:A5:6D:4C:8C:1E:B2:62:10:4A:F6:70:61:C4:FC:
:BB:05:2F:70:9F:04:A9:0E:05:E9:26:33:E8:70:88:A2 3C:E0:01:D6:F3:24:80:74:DA:7C:3E:50:18:7B:CE:4D
a=setup:actpass
a=dtls-id:1
a=rtcp-mux
a=rtcp-mux-only
a=rtcp-rsize
a=candidate:1 1 udp 255 192.0.2.200 12200 typ relay
raddr 0.0.0.0 rport 0
a=end-of-candidates
m=video 12200 UDP/TLS/RTP/SAVPF 100 101
c=IN IP4 192.0.2.200
a=mid:v1
a=sendrecv
a=rtpmap:100 VP8/90000
a=rtpmap:101 rtx/90000
a=fmtp:101 apt=100
a=extmap:1 urn:ietf:params:rtp-hdrext:sdes:mid
a=rtcp-fb:100 ccm fir
a=rtcp-fb:100 nack
a=rtcp-fb:100 nack pli
a=msid:751f239e-4ae0-c549-aa3d-890de772998b
39292672-c102-d075-f580-5826f31ca958
The SDP for |answer-C2| looks like:
v=0
o=- 1070771854436052752 2 IN IP4 0.0.0.0
s=-
t=0 0
a=ice-options:trickle
a=group:BUNDLE a1 v1
a=group:LS a1 v1
m=audio 12100 UDP/TLS/RTP/SAVPF 96 0 8 97 98
c=IN IP4 192.0.2.100
a=mid:a1
a=sendrecv
a=rtpmap:96 opus/48000/2
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:97 telephone-event/8000
a=rtpmap:98 telephone-event/48000
a=maxptime:120
a=extmap:1 urn:ietf:params:rtp-hdrext:sdes:mid
a=extmap:2 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=msid:bbce3ba6-abfc-ac63-d00a-e15b286f8fce
e80098db-7159-3c06-229a-00df2a9b3dbc
a=ice-ufrag:4ZcD
a=ice-pwd:ZaaG6OG7tCn4J/lehAGz+HHD
a=fingerprint:sha-256
C4:68:F8:77:6A:44:F1:98:6D:7C:9F:47:EB:E3:34:A4:
0A:AA:2D:49:08:28:70:2E:1F:AE:18:7D:4E:3E:66:BF
a=setup:passive a=setup:passive
a=dtls-id:1
a=rtcp-mux a=rtcp-mux
a=rtcp-mux-only
a=rtcp-rsize a=rtcp-rsize
a=extmap:2 urn:ietf:params:rtp-hdrext:sdes:mid a=candidate:1 1 udp 255 192.0.2.100 12100 typ relay
raddr 0.0.0.0 rport 0
a=end-of-candidates
m=video 12100 UDP/TLS/RTP/SAVPF 100 101
c=IN IP4 192.0.2.100
a=mid:v1
a=sendrecv
a=rtpmap:100 VP8/90000
a=rtpmap:101 rtx/90000
a=fmtp:101 apt=100
a=extmap:1 urn:ietf:params:rtp-hdrext:sdes:mid
a=rtcp-fb:100 ccm fir a=rtcp-fb:100 ccm fir
a=rtcp-fb:100 nack a=rtcp-fb:100 nack
a=rtcp-fb:100 nack pli a=rtcp-fb:100 nack pli
a=msid:bbce3ba6-abfc-ac63-d00a-e15b286f8fce
ac701365-eb06-42df-cc93-7f22bc308789
8. Security Considerations 8. Security Considerations
The IETF has published separate documents The IETF has published separate documents
[I-D.ietf-rtcweb-security-arch] [I-D.ietf-rtcweb-security] describing [I-D.ietf-rtcweb-security-arch] [I-D.ietf-rtcweb-security] describing
the security architecture for WebRTC as a whole. The remainder of the security architecture for WebRTC as a whole. The remainder of
this section describes security considerations for this document. this section describes security considerations for this document.
While formally the JSEP interface is an API, it is better to think of While formally the JSEP interface is an API, it is better to think of
it is an Internet protocol, with the JS being untrustworthy from the it is an Internet protocol, with the JS being untrustworthy from the
perspective of the browser. Thus, the threat model of [RFC3552] perspective of the endpoint. Thus, the threat model of [RFC3552]
applies. In particular, JS can call the API in any order and with applies. In particular, JS can call the API in any order and with
any inputs, including malicious ones. This is particularly relevant any inputs, including malicious ones. This is particularly relevant
when we consider the SDP which is passed to setLocalDescription(). when we consider the SDP which is passed to setLocalDescription().
While correct API usage requires that the application pass in SDP While correct API usage requires that the application pass in SDP
which was derived from createOffer() or createAnswer(), there is no which was derived from createOffer() or createAnswer(), there is no
guarantee that applications do so. The browser MUST be prepared for guarantee that applications do so. The JSEP implementation MUST be
the JS to pass in bogus data instead. prepared for the JS to pass in bogus data instead.
Conversely, the application programmer MUST recognize that the JS Conversely, the application programmer MUST recognize that the JS
does not have complete control of browser behavior. One case that does not have complete control of endpoint behavior. One case that
bears particular mention is that editing ICE candidates out of the bears particular mention is that editing ICE candidates out of the
SDP or suppressing trickled candidates does not have the expected SDP or suppressing trickled candidates does not have the expected
behavior: implementations will still perform checks from those behavior: implementations will still perform checks from those
candidates even if they are not sent to the other side. Thus, for candidates even if they are not sent to the other side. Thus, for
instance, it is not possible to prevent the remote peer from learning instance, it is not possible to prevent the remote peer from learning
your public IP address by removing server reflexive candidates. your public IP address by removing server reflexive candidates.
Applications which wish to conceal their public IP address should Applications which wish to conceal their public IP address should
instead configure the ICE agent to use only relay candidates. instead configure the ICE agent to use only relay candidates.
9. IANA Considerations 9. IANA Considerations
This document requires no actions from IANA. This document requires no actions from IANA.
10. Acknowledgements 10. Acknowledgements
Significant text incorporated in the draft as well and review was Harald Alvestrand, Taylor Brandstetter, Suhas Nandakumar, and Peter
provided by Peter Thatcher, Taylor Brandstetter, Harald Alvestrand Thatcher provided significant text for this draft. Bernard Aboba,
and Suhas Nandakumar. Dan Burnett, Neil Stratford, Anant Narayanan, Adam Bergkvist, Dan Burnett, Ben Campbell, Alissa Cooper, Richard
Andrew Hutton, Richard Ejzak, Adam Bergkvist and Matthew Kaufman all Ejzak, Stefan Hakansson, Ted Hardie, Christer Holmberg Andrew Hutton,
Randell Jesup, Matthew Kaufman, Anant Narayanan, Adam Roach, Neil
Stratford, Martin Thomson, Sean Turner, and Magnus Westerlund all
provided valuable feedback on this proposal. provided valuable feedback on this proposal.
11. References 11. References
11.1. Normative References 11.1. Normative References
[I-D.ietf-avtcore-rtp-multi-stream] [I-D.ietf-avtcore-rtp-multi-stream]
Lennox, J., Westerlund, M., Wu, Q., and C. Perkins, Lennox, J., Westerlund, M., Wu, Q., and C. Perkins,
"Sending Multiple RTP Streams in a Single RTP Session", "Sending Multiple RTP Streams in a Single RTP Session",
draft-ietf-avtcore-rtp-multi-stream-11 (work in progress), draft-ietf-avtcore-rtp-multi-stream-11 (work in progress),
skipping to change at page 85, line 26 skipping to change at page 98, line 38
[RFC5888] Camarillo, G. and H. Schulzrinne, "The Session Description [RFC5888] Camarillo, G. and H. Schulzrinne, "The Session Description
Protocol (SDP) Grouping Framework", RFC 5888, June 2010. Protocol (SDP) Grouping Framework", RFC 5888, June 2010.
[RFC6236] Johansson, I. and K. Jung, "Negotiation of Generic Image [RFC6236] Johansson, I. and K. Jung, "Negotiation of Generic Image
Attributes in the Session Description Protocol (SDP)", Attributes in the Session Description Protocol (SDP)",
RFC 6236, May 2011. RFC 6236, May 2011.
[RFC6347] Rescorla, E. and N. Modadugu, "Datagram Transport Layer [RFC6347] Rescorla, E. and N. Modadugu, "Datagram Transport Layer
Security Version 1.2", RFC 6347, January 2012. Security Version 1.2", RFC 6347, January 2012.
[RFC6716] Valin, JM., Vos, K., and T. Terriberry, "Definition of the
Opus Audio Codec", RFC 6716, DOI 10.17487/RFC6716,
September 2012, <http://www.rfc-editor.org/info/rfc6716>.
[RFC6904] Lennox, J., "Encryption of Header Extensions in the Secure [RFC6904] Lennox, J., "Encryption of Header Extensions in the Secure
Real-time Transport Protocol (SRTP)", RFC 6904, April Real-time Transport Protocol (SRTP)", RFC 6904, April
2013. 2013.
[RFC7160] Petit-Huguenin, M. and G. Zorn, Ed., "Support for Multiple
Clock Rates in an RTP Session", RFC 7160,
DOI 10.17487/RFC7160, April 2014,
<http://www.rfc-editor.org/info/rfc7160>.
[RFC7587] Spittka, J., Vos, K., and JM. Valin, "RTP Payload Format
for the Opus Speech and Audio Codec", RFC 7587,
DOI 10.17487/RFC7587, June 2015,
<http://www.rfc-editor.org/info/rfc7587>.
[RFC7850] Nandakumar, S., "Registering Values of the SDP 'proto' [RFC7850] Nandakumar, S., "Registering Values of the SDP 'proto'
Field for Transporting RTP Media over TCP under Various Field for Transporting RTP Media over TCP under Various
RTP Profiles", RFC 7850, DOI 10.17487/RFC7850, April 2016, RTP Profiles", RFC 7850, DOI 10.17487/RFC7850, April 2016,
<http://www.rfc-editor.org/info/rfc7850>. <http://www.rfc-editor.org/info/rfc7850>.
[RFC7941] Westerlund, M., Burman, B., Even, R., and M. Zanaty, "RTP
Header Extension for the RTP Control Protocol (RTCP)
Source Description Items", RFC 7941, DOI 10.17487/RFC7941,
August 2016, <http://www.rfc-editor.org/info/rfc7941>.
11.2. Informative References 11.2. Informative References
[I-D.ietf-avtext-lrr]
Lennox, J., Hong, D., Uberti, J., Homer, S., and M.
Flodman, "The Layer Refresh Request (LRR) RTCP Feedback
Message", draft-ietf-avtext-lrr-03 (work in progress),
July 2016.
[I-D.ietf-rtcweb-ip-handling] [I-D.ietf-rtcweb-ip-handling]
Uberti, J. and G. Shieh, "WebRTC IP Address Handling Uberti, J. and G. Shieh, "WebRTC IP Address Handling
Recommendations", draft-ietf-rtcweb-ip-handling-01 (work Recommendations", draft-ietf-rtcweb-ip-handling-01 (work
in progress), March 2016. in progress), March 2016.
[I-D.nandakumar-rtcweb-sdp] [I-D.nandakumar-rtcweb-sdp]
Nandakumar, S. and C. Jennings, "SDP for the WebRTC", Nandakumar, S. and C. Jennings, "SDP for the WebRTC",
draft-nandakumar-rtcweb-sdp-02 (work in progress), July draft-nandakumar-rtcweb-sdp-02 (work in progress), July
2013. 2013.
skipping to change at page 86, line 14 skipping to change at page 100, line 5
[RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V. [RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V.
Jacobson, "RTP: A Transport Protocol for Real-Time Jacobson, "RTP: A Transport Protocol for Real-Time
Applications", STD 64, RFC 3550, DOI 10.17487/RFC3550, Applications", STD 64, RFC 3550, DOI 10.17487/RFC3550,
July 2003, <http://www.rfc-editor.org/info/rfc3550>. July 2003, <http://www.rfc-editor.org/info/rfc3550>.
[RFC3556] Casner, S., "Session Description Protocol (SDP) Bandwidth [RFC3556] Casner, S., "Session Description Protocol (SDP) Bandwidth
Modifiers for RTP Control Protocol (RTCP) Bandwidth", Modifiers for RTP Control Protocol (RTCP) Bandwidth",
RFC 3556, July 2003. RFC 3556, July 2003.
[RFC3611] Friedman, T., Caceres, R., and A. Clark, "RTP Control
Protocol Extended Reports (RTCP XR)", RFC 3611,
DOI 10.17487/RFC3611, November 2003,
<http://www.rfc-editor.org/info/rfc3611>.
[RFC3960] Camarillo, G. and H. Schulzrinne, "Early Media and Ringing [RFC3960] Camarillo, G. and H. Schulzrinne, "Early Media and Ringing
Tone Generation in the Session Initiation Protocol (SIP)", Tone Generation in the Session Initiation Protocol (SIP)",
RFC 3960, December 2004. RFC 3960, December 2004.
[RFC4568] Andreasen, F., Baugher, M., and D. Wing, "Session [RFC4568] Andreasen, F., Baugher, M., and D. Wing, "Session
Description Protocol (SDP) Security Descriptions for Media Description Protocol (SDP) Security Descriptions for Media
Streams", RFC 4568, July 2006. Streams", RFC 4568, July 2006.
[RFC4588] Rey, J., Leon, D., Miyazaki, A., Varsa, V., and R. [RFC4588] Rey, J., Leon, D., Miyazaki, A., Varsa, V., and R.
Hakenberg, "RTP Retransmission Payload Format", RFC 4588, Hakenberg, "RTP Retransmission Payload Format", RFC 4588,
July 2006. July 2006.
[RFC4733] Schulzrinne, H. and T. Taylor, "RTP Payload for DTMF [RFC4733] Schulzrinne, H. and T. Taylor, "RTP Payload for DTMF
Digits, Telephony Tones, and Telephony Signals", RFC 4733, Digits, Telephony Tones, and Telephony Signals", RFC 4733,
DOI 10.17487/RFC4733, December 2006, DOI 10.17487/RFC4733, December 2006,
<http://www.rfc-editor.org/info/rfc4733>. <http://www.rfc-editor.org/info/rfc4733>.
[RFC5104] Wenger, S., Chandra, U., Westerlund, M., and B. Burman,
"Codec Control Messages in the RTP Audio-Visual Profile
with Feedback (AVPF)", RFC 5104, DOI 10.17487/RFC5104,
February 2008, <http://www.rfc-editor.org/info/rfc5104>.
[RFC5506] Johansson, I. and M. Westerlund, "Support for Reduced-Size [RFC5506] Johansson, I. and M. Westerlund, "Support for Reduced-Size
Real-Time Transport Control Protocol (RTCP): Opportunities Real-Time Transport Control Protocol (RTCP): Opportunities
and Consequences", RFC 5506, April 2009. and Consequences", RFC 5506, April 2009.
[RFC5576] Lennox, J., Ott, J., and T. Schierl, "Source-Specific [RFC5576] Lennox, J., Ott, J., and T. Schierl, "Source-Specific
Media Attributes in the Session Description Protocol Media Attributes in the Session Description Protocol
(SDP)", RFC 5576, June 2009. (SDP)", RFC 5576, June 2009.
[RFC5763] Fischl, J., Tschofenig, H., and E. Rescorla, "Framework [RFC5763] Fischl, J., Tschofenig, H., and E. Rescorla, "Framework
for Establishing a Secure Real-time Transport Protocol for Establishing a Secure Real-time Transport Protocol
skipping to change at page 87, line 5 skipping to change at page 101, line 11
[RFC5764] McGrew, D. and E. Rescorla, "Datagram Transport Layer [RFC5764] McGrew, D. and E. Rescorla, "Datagram Transport Layer
Security (DTLS) Extension to Establish Keys for the Secure Security (DTLS) Extension to Establish Keys for the Secure
Real-time Transport Protocol (SRTP)", RFC 5764, May 2010. Real-time Transport Protocol (SRTP)", RFC 5764, May 2010.
[RFC6464] Lennox, J., Ed., Ivov, E., and E. Marocco, "A Real-time [RFC6464] Lennox, J., Ed., Ivov, E., and E. Marocco, "A Real-time
Transport Protocol (RTP) Header Extension for Client-to- Transport Protocol (RTP) Header Extension for Client-to-
Mixer Audio Level Indication", RFC 6464, Mixer Audio Level Indication", RFC 6464,
DOI 10.17487/RFC6464, December 2011, DOI 10.17487/RFC6464, December 2011,
<http://www.rfc-editor.org/info/rfc6464>. <http://www.rfc-editor.org/info/rfc6464>.
[RFC6544] Rosenberg, J., Keranen, A., Lowekamp, B., and A. Roach,
"TCP Candidates with Interactive Connectivity
Establishment (ICE)", RFC 6544, DOI 10.17487/RFC6544,
March 2012, <http://www.rfc-editor.org/info/rfc6544>.
[RFC7656] Lennox, J., Gross, K., Nandakumar, S., Salgueiro, G., and [RFC7656] Lennox, J., Gross, K., Nandakumar, S., Salgueiro, G., and
B. Burman, Ed., "A Taxonomy of Semantics and Mechanisms B. Burman, Ed., "A Taxonomy of Semantics and Mechanisms
for Real-Time Transport Protocol (RTP) Sources", RFC 7656, for Real-Time Transport Protocol (RTP) Sources", RFC 7656,
DOI 10.17487/RFC7656, November 2015, DOI 10.17487/RFC7656, November 2015,
<http://www.rfc-editor.org/info/rfc7656>. <http://www.rfc-editor.org/info/rfc7656>.
[TS26.114]
3GPP TS 26.114 V12.8.0, "3rd Generation Partnership
Project; Technical Specification Group Services and System
Aspects; IP Multimedia Subsystem (IMS); Multimedia
Telephony; Media handling and interaction (Release 12)",
December 2014, <http://www.3gpp.org/DynaReport/26114.htm>.
[W3C.WD-webrtc-20140617] [W3C.WD-webrtc-20140617]
Bergkvist, A., Burnett, D., Narayanan, A., and C. Bergkvist, A., Burnett, D., Narayanan, A., and C.
Jennings, "WebRTC 1.0: Real-time Communication Between Jennings, "WebRTC 1.0: Real-time Communication Between
Browsers", World Wide Web Consortium WD WD-webrtc- Browsers", World Wide Web Consortium WD WD-webrtc-
20140617, June 2014, 20140617, June 2014,
<http://www.w3.org/TR/2011/WD-webrtc-20140617>. <http://www.w3.org/TR/2011/WD-webrtc-20140617>.
Appendix A. Appendix A Appendix A. Appendix A
For the syntax validation performed in Section 5.7, the following For the syntax validation performed in Section 5.7, the following
skipping to change at page 88, line 50 skipping to change at page 102, line 50
+-----------------------+-------------------------------------------+ +-----------------------+-------------------------------------------+
Table 1: SDP ABNF References Table 1: SDP ABNF References
Appendix B. Appendix B Appendix B. Appendix B
The following text is meant to completely replace section The following text is meant to completely replace section
"Associating RTP/RTCP Streams With Correct SDP Media Description" of "Associating RTP/RTCP Streams With Correct SDP Media Description" of
[I-D.ietf-mmusic-sdp-bundle-negotiation]. [I-D.ietf-mmusic-sdp-bundle-negotiation].
As described in [RFC3550], RTP/RTCP packets are associated with RTP As described in [RFC3550], RTP packets are associated with RTP
streams as defined in [RFC7656]. Each RTP stream is identified by an streams [RFC7656]. Each RTP stream is identified by an SSRC value,
SSRC value, and each RTP/RTCP packet carries an SSRC value that is and each RTP packet includes an SSRC field that is used to associate
used to associate the packet with the correct RTP stream. An RTCP the packet with the correct RTP stream. RTCP packets also use SSRCs
packet can carry multiple SSRC values, and might therefore be to identify which RTP streams the packet relates to. However, a RTCP
associated with multiple RTP streams. packet can contain multiple SSRC fields, in the course of providing
feedback or reports on different RTP streams, and therefore can be
associated with multiple such streams.
In order to be able to process received RTP/RTCP packets correctly it In order to be able to process received RTP/RTCP packets correctly,
must be possible to associate an RTP stream with the correct "m=" it must be possible to associate an RTP stream with the correct "m="
line, as the "m=" line and SDP attributes associated with the "m=" line, as the "m=" line and SDP attributes associated with the "m="
line contain information needed to process the packets. line contain information needed to process the packets.
As all RTP streams associated with a BUNDLE group use the same As all RTP streams associated with a BUNDLE group use the same
address:port combination for sending and receiving RTP/RTCP packets, address:port combination for sending and receiving RTP/RTCP packets,
the local address:port combination cannot be used to associate an RTP the local address:port combination cannot be used to associate an RTP
stream with the correct "m=" line. In addition, multiple RTP streams stream with the correct "m=" line. In addition, multiple RTP streams
might be associated with the same "m=" line. might be associated with the same "m=" line.
An offerer and answerer can inform each other which SSRC values they An offerer and answerer can inform each other which SSRC values they
skipping to change at page 89, line 33 skipping to change at page 103, line 35
that information. Due to this, before the offerer has received the that information. Due to this, before the offerer has received the
answer, the offerer will not be able to associate an RTP stream with answer, the offerer will not be able to associate an RTP stream with
the correct "m=" line using the SSRC value associated with the RTP the correct "m=" line using the SSRC value associated with the RTP
stream. In addition, the offerer and answerer may start using new stream. In addition, the offerer and answerer may start using new
SSRC values mid-session, without informing each other using the SDP SSRC values mid-session, without informing each other using the SDP
'ssrc' attribute. 'ssrc' attribute.
In order for an offerer and answerer to always be able to associate In order for an offerer and answerer to always be able to associate
an RTP stream with the correct "m=" line, the offerer and answerer an RTP stream with the correct "m=" line, the offerer and answerer
using the BUNDLE extension MUST support the mechanism defined in using the BUNDLE extension MUST support the mechanism defined in
[I-D.ietf-mmusic-sdp-bundle-negotiation] section 14. where the [I-D.ietf-mmusic-sdp-bundle-negotiation], Section 14, where the
offerer and answerer insert the identification-tag associated with an offerer and answerer insert the identification-tag associated with an
"m=" line (provided by the remote peer) into RTP and RTCP packets "m=" line (provided by the remote peer) into RTP and RTCP packets
associated with a BUNDLE group. associated with a BUNDLE group.
The mapping from an SSRC to an identification-tag is carried in RTCP When using this mechanism, the mapping from an SSRC to an
SDES packets or in RTP header extensions identification-tag is carried in RTP header extensions or RTCP SDES
([I-D.ietf-mmusic-sdp-bundle-negotiation] section 14). Since a packets, as specified in [I-D.ietf-mmusic-sdp-bundle-negotiation],
compound RTCP packet can contain multiple RTCP SDES packets, and each Section 14). Since a compound RTCP packet can contain multiple RTCP
RTCP SDES packet can contain multiple chunks, an RTCP packet can SDES packets, and each RTCP SDES packet can contain multiple chunks,
contain several SSRC to identification-tag mappings. The offerer and a single RTCP packet can contain several SSRC to identification-tag
answerer maintain tables used for routing that are updated each time mappings. The offerer and answerer maintain tables used for routing
an RTP/RTCP packet contains new information that affects how packets that are updated each time an RTP/RTCP packet contains new
should be routed. information that affects how packets should be routed.
To prepare for demultiplexing RTP packets to the correct "m=" line, However, some implementations of
the following steps MUST be followed for each BUNDLE group. [I-D.ietf-mmusic-sdp-bundle-negotiation] may not include this
identification-tag in their RTP and RTCP traffic when using BUNDLE,
and instead use a payload type based mechanism for demuxing. In this
situation, each "m=" line MUST use unique payload type values, in
order for the payload type to be a reliable indicator of the relevant
"m=" line for the RTP stream.
Applications can implement RTP stacks in many different ways. The
algorithm below details one way that demultiplexing can be
accomplished, but is not meant to be prescriptive about exactly how
an RTP stack needs to be implemented. Applications MAY use any
algorithm that achieves equivalent results to those described in the
algorithm below.
To prepare for demultiplexing RTP/RTCP packets to the correct "m="
line, the following steps MUST be followed for each BUNDLE group.
Construct a table mapping MID to "m=" line for each "m=" line in Construct a table mapping MID to "m=" line for each "m=" line in
this BUNDLE group. Note that an "m=" line may only have one MID. this BUNDLE group. Note that an "m=" line may only have one MID.
Construct a table mapping incoming SSRC to "m=" line for each "m=" Construct a table mapping incoming SSRC to "m=" line for each "m="
line in this BUNDLE group and for each SSRC configured for line in this BUNDLE group and for each SSRC configured for
receiving in that "m=" line. receiving in that "m=" line.
Construct a table mapping outgoing SSRC to "m=line" for each "m=" Construct a table mapping outgoing SSRC to "m=line" for each "m="
line in this BUNDLE group and for each SSRC configured for sending line in this BUNDLE group and for each SSRC configured for sending
in that "m=" line. in that "m=" line.
Construct a table mapping payload type to "m=" line for each "m=" Construct a table mapping payload type to "m=" line for each "m="
line in the BUNDLE group and for each payload type configured for line in the BUNDLE group and for each payload type configured for
receiving in that "m=" line. If any payload type is configured receiving in that "m=" line. If any payload type is configured
for receiving in more than one "m=" line in the BUNDLE group, do for receiving in more than one "m=" line in the BUNDLE group, do
not it include it in the table. not it include it in the table, as it cannot be used to uniquely
identify a "m=" line.
Note that for each of these tables, there can only be one mapping Note that for each of these tables, there can only be one mapping
for any given key (MID, SSRC, or PT). In other words, the tables for any given key (MID, SSRC, or PT). In other words, the tables
are not multimaps. are not multimaps.
As "m=" lines are added or removed from the BUNDLE groups, or their As "m=" lines are added or removed from the BUNDLE groups, or their
configurations are changed, the tables above MUST also be updated. configurations are changed, the tables above MUST also be updated.
For each RTP packet received, the following steps MUST be followed to For each RTP packet received, the following steps MUST be followed to
route the packet to the correct "m=" section within a BUNDLE group. route the packet to the correct "m=" section within a BUNDLE group.
Note that the phrase 'deliver a packet to the "m=" line' means to Note that the phrase 'deliver a packet to the "m=" line' means to
further process the packet as would normally happen with RTP/RTCP, if further process the packet as would normally happen with RTP/RTCP, if
it were received on a transport associated with that "m=" line it were received on a transport associated with that "m=" line
outside of a BUNDLE group (i.e., if the "m=" line were not BUNDLEd), outside of a BUNDLE group (i.e., if the "m=" line were not BUNDLEd),
including dropping an RTP packet if the packet's PT does not match including dropping an RTP packet if the packet's PT does not match
any PT in the "m=" line. any PT in the "m=" line.
If the packet has a MID and that MID is not in the table mapping If the packet has a MID, and that MID is not in the table mapping
MID to "m=" line, drop the packet and stop. MID to "m=" line, drop the packet and stop.
If the packet has a MID and that MID is in the table mapping MID If the packet has a MID, and the packet's extended sequence number
to "m=" line, update the incoming SSRC mapping table to include an is greater than that of the last MID update, as discussed in
entry that maps the packet's SSRC to the "m=" line for that MID. [RFC7941], Section 4.2.6, update the incoming SSRC mapping table
to include an entry that maps the packet's SSRC to the "m=" line
for that MID.
If the packet's SSRC is in the incoming SSRC mapping table, route If the packet's SSRC is in the incoming SSRC mapping table, check
the packet to the associated "m=" line and stop. that the packet's PT matches a PT included on the associated "m="
line. If so, route the packet to that associated "m=" line and
stop; otherwise drop the packet and stop.
If the packet's payload type is in the payload type table, update If the packet's payload type is in the payload type table, update
the the incoming SSRC mapping table to include an entry that maps the the incoming SSRC mapping table to include an entry that maps
the packet's SSRC to the "m=" line for that payload type. In the packet's SSRC to the "m=" line for that payload type. In
addition, route the packet to the associated "m=" line and stop. addition, route the packet to the associated "m=" line and stop.
Otherwise, drop the packet. Otherwise, drop the packet.
For each RTCP packet received (including each RTCP packet that is For each RTCP packet received (including each RTCP packet that is
part of a compound RTCP packet), the packet MUST be routed to the part of a compound RTCP packet), the packet MUST be routed to the
appropriate handler for the SSRCs it contains information about. "m=" line for the RTP streams it contains information about. This
Some examples of such handling are given below. routing is type-dependent, as each kind of RTCP packet has its own
mechanism for associating it with the relevant RTP streams.
If the packet is of type SR, and the sender SSRC for the packet is Packets for which no appropriate "m=" line can be identified (i.e.,
for unknown RTP streams) are not relevant in the context of this
algorithm and MAY be dropped. This situation may occur with certain
multiparty RTP topologies.
Rules for handling the various types of RTCP packets are explained
below.
If the packet is of type SDES, for each chunk in the packet whose
SSRC is found in the incoming SSRC table, deliver a copy of the
packet to the "m=" line associated with that SSRC. In addition,
for any SDES MID items contained in these chunks, if the MID is
found in the table mapping MID to "m=" line, update the incoming
SSRC table to include an entry that maps the chunk's SSRC to the
"m=" line associated with that MID, unless the packet is older
than the packet that most recently updated the mapping for this
SSRC, as discussed in [RFC7941], Section 4.2.6.
Note that if an SDES packet is received as part of a compound RTCP
packet, the SSRC to "m=" line mapping may not exist until the SDES
packet is handled (e.g., in the case where RTCP for a source is
received before any RTP packets). Therefore, when processing a
compound packet, any contained SDES packet MUST be handled first.
If the packet is of type BYE, it indicates that the RTP streams
referenced in the packet are ending. Therefore, for each SSRC
indicated in the packet that is found in the incoming SSRC table,
first deliver a copy of the packet to the "m=" line associated
with that SSRC, but then remove the entry for that SSRC from the
incoming SSRC table.
If the packet is of type SR or RR, for each report block in the
report whose "SSRC of source" is found in the outgoing SSRC table,
deliver a copy of the RTCP packet to the "m=" line associated with
that SSRC. In addition, if the packet is of type SR, and the
sender SSRC for the packet is found in the incoming SSRC table,
deliver a copy of the packet to the "m=" line associated with that
SSRC.
If the implementation supports RTCP XR and the packet is of type
XR, as defined in [RFC3611], for each report block in the report
whose "SSRC of source" is is found in the outgoing SSRC table,
deliver a copy of the RTCP packet to the "m=" line associated with
that SSRC. In addition, if the sender SSRC for the packet is
found in the incoming SSRC table, deliver a copy of the packet to found in the incoming SSRC table, deliver a copy of the packet to
the "m=" line associated with that SSRC. In addition, for each the "m=" line associated with that SSRC.
report block in the report whose SSRC is found in the outgoing
SSRC table, deliver a copy of the RTCP packet to the "m=" line
associated with that SSRC.
If the packet is of type RR, for each report block in the packet If the packet is a feedback message of type RTPFB or PSFB, as
whose SSRC is found in the outgoing SSRC table, deliver a copy of defined in [RFC4585], it will contain a media source SSRC, and
the RTCP packet to the "m=" line associated with that SSRC. this SSRC is used for routing certain subtypes of feedback
messages. However, several subtypes of PSFB messages include
target SSRC(s) in a section called Feedback Control Information
(FCI). For these messages, the target SSRC(s) are used for
routing.
If the packet is of type SDES, and the sender SSRC for the packet If the packet is a feedback message that does not include target
is found in the incoming SSRC table, deliver the packet to the SSRCs in its FCI section, and the media source SSRC is found in
"m=" line associated with that SSRC. In addition, for each chunk the outgoing SSRC table, deliver the packet to the "m=" line
in the packet that contains a MID that is in the table mapping MID associated with that SSRC. RTPFB and PSFB types that are handled
to "m=" line, update the incoming SSRC mapping table to include an in this way include:
entry that maps the SSRC for that chunk to the "m=" line
associated with that MID. (This case can occur when RTCP for a
source is received before any RTP packets.)
If the packet is of type BYE, for each SSRC indicated in the Generic NACK: [RFC4585] (PT=RTPFB, FMT=1).
packet that is found in the incoming SSRC table, deliver a copy of
the packet to the "m=" line associated with that SSRC.
If the packet is of type RTPFB or PSFB, as defined in [RFC4585], Picture Loss Indication (PLI): [RFC4585] (PT=PSFB, FMT=1).
and the media source SSRC for the packet is found in the outgoing
SSRC table, deliver the packet to the "m=" line associated with Slice Loss Indication (SLI): [RFC4585] (PT=PSFB, FMT=2).
that SSRC.
Reference Picture Selection Indication (RPSI): [RFC4585]
(PT=PSFB, FMT=3).
If the packet is a feedback message that does include target
SSRC(s) in its FCI section, it can either be a request or a
notification. Requests reference a RTP stream that is being sent
by the message recipient, whereas notifications are responses to
an earlier request, and therefore reference a RTP stream that is
being received by the message recipient.
If the packet is a feedback request that includes target SSRC(s),
for each target SSRC that is found in the outgoing SSRC table,
deliver a copy of the RTCP packet to the "m=" line associated with
that SSRC. PSFB types that are handled in this way include:
Full Intra Request (FIR): [RFC5104] (PT=PSFB, FMT=4).
Temporal-Spatial Trade-off Request (TSTR): [RFC5104] (PT=PSFB,
FMT=5).
H.271 Video Back Channel Message (VBCM): [RFC5104]
(PT=PSFB, FMT=7).
Layer Refresh Request (LRR): [I-D.ietf-avtext-lrr] (PT=PSFB,
FMT=TBD).
If the packet is a feedback notification that include target
SSRC(s), for each target SSRC that is found in the incoming SSRC
table, deliver a copy of the RTCP packet to the "m=" line
associated with that SSRC. PSFB types that are handled in this
way include:
Temporal-Spatial Trade-off Notification (TSTN): [RF
C5104] (PT=PSFB, FMT=6). This message is a notification in
response to a prior TSTR.
If the packet is of type APP, the only routing information
included is the source of the packet, and therefore the packet
could be related to any existing "m=" line. Accordingly, deliver
a copy of the packet to each "m=" line.
Appendix C. Change log Appendix C. Change log
Note: This section will be removed by RFC Editor before publication. Note: This section will be removed by RFC Editor before publication.
Changes in draft-19:
o Examples are now machine-generated for correctness, and use IETF-
approved example IP addresses.
o Add early transport warmup example, and add missing attributes to
existing examples.
o Only send "a=rtcp-mux-only" and "a=bundle-only" on new m=
sections.
o Update references.
o Add coverage of a=identity.
o Explain the lipsync group algorithm more thoroughly.
o Remove unnecessary list of MTI specs.
o Allow codecs which weren't offered to appear in answers and which
weren't selected to appear in subsequent offers.
o Codec preferences now are applied on both initial and subsequent
offers and answers.
o Clarify a=msid handling for recvonly m= sections.
o Clarify behavior of attributes for bundle-only data channels.
o Allow media attributes to appear in data m= sections when all the
media m= sections are bundle-only.
o Use consistent terminology for JSEP implementations.
o Describe how to handle failed API calls.
o Some cleanup on routing rules.
Changes in draft-18: Changes in draft-18:
o Update demux algorithm and move it to an appendix in preparation o Update demux algorithm and move it to an appendix in preparation
for merging it into BUNDLE. for merging it into BUNDLE.
o Clarify why we can't handle an incoming offer to send simulcast. o Clarify why we can't handle an incoming offer to send simulcast.
o Expand IceCandidate object text. o Expand IceCandidate object text.
o Further document use of ICE candidate pool. o Further document use of ICE candidate pool.
 End of changes. 279 change blocks. 
838 lines changed or deleted 1456 lines changed or added

This html diff was produced by rfcdiff 1.45. The latest version is available from http://tools.ietf.org/tools/rfcdiff/