Javascript Session Establishment Protocol
draft-ietf-rtcweb-jsep-20
The information below is for an old version of the document.
| Document | Type |
This is an older version of an Internet-Draft that was ultimately published as RFC 8829.
|
|
|---|---|---|---|
| Authors | Justin Uberti , Cullen Fluffy Jennings , Eric Rescorla | ||
| Last updated | 2017-05-01 (Latest revision 2017-03-29) | ||
| Replaces | draft-uberti-rtcweb-jsep | ||
| RFC stream | Internet Engineering Task Force (IETF) | ||
| Formats | |||
| Reviews |
GENART Last Call review
(of
-21)
by Robert Sparks
Almost ready
|
||
| Additional resources | Mailing list discussion | ||
| Stream | WG state | Submitted to IESG for Publication | |
| Document shepherd | Ted Hardie | ||
| Shepherd write-up | Show Last changed 2017-04-04 | ||
| IESG | IESG state | Became RFC 8829 (Proposed Standard) | |
| Consensus boilerplate | Yes | ||
| Telechat date | (None) | ||
| Responsible AD | Adam Roach | ||
| Send notices to | (None) |
draft-ietf-rtcweb-jsep-20
Network Working Group J. Uberti
Internet-Draft Google
Intended status: Standards Track C. Jennings
Expires: September 30, 2017 Cisco
E. Rescorla, Ed.
Mozilla
March 29, 2017
Javascript Session Establishment Protocol
draft-ietf-rtcweb-jsep-20
Abstract
This document describes the mechanisms for allowing a Javascript
application to control the signaling plane of a multimedia session
via the interface specified in the W3C RTCPeerConnection API, and
discusses how this relates to existing signaling protocols.
Status of This Memo
This Internet-Draft is submitted in full conformance with the
provisions of BCP 78 and BCP 79.
Internet-Drafts are working documents of the Internet Engineering
Task Force (IETF). Note that other groups may also distribute
working documents as Internet-Drafts. The list of current Internet-
Drafts is at http://datatracker.ietf.org/drafts/current/.
Internet-Drafts are draft documents valid for a maximum of six months
and may be updated, replaced, or obsoleted by other documents at any
time. It is inappropriate to use Internet-Drafts as reference
material or to cite them other than as "work in progress."
This Internet-Draft will expire on September 30, 2017.
Copyright Notice
Copyright (c) 2017 IETF Trust and the persons identified as the
document authors. All rights reserved.
This document is subject to BCP 78 and the IETF Trust's Legal
Provisions Relating to IETF Documents
(http://trustee.ietf.org/license-info) in effect on the date of
publication of this document. Please review these documents
carefully, as they describe your rights and restrictions with respect
to this document. Code Components extracted from this document must
include Simplified BSD License text as described in Section 4.e of
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the Trust Legal Provisions and are provided without warranty as
described in the Simplified BSD License.
Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 4
1.1. General Design of JSEP . . . . . . . . . . . . . . . . . 4
1.2. Other Approaches Considered . . . . . . . . . . . . . . . 5
2. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 6
3. Semantics and Syntax . . . . . . . . . . . . . . . . . . . . 6
3.1. Signaling Model . . . . . . . . . . . . . . . . . . . . . 6
3.2. Session Descriptions and State Machine . . . . . . . . . 7
3.3. Session Description Format . . . . . . . . . . . . . . . 10
3.4. Session Description Control . . . . . . . . . . . . . . . 10
3.4.1. RtpTransceivers . . . . . . . . . . . . . . . . . . . 10
3.4.2. RtpSenders . . . . . . . . . . . . . . . . . . . . . 11
3.4.3. RtpReceivers . . . . . . . . . . . . . . . . . . . . 11
3.5. ICE . . . . . . . . . . . . . . . . . . . . . . . . . . . 11
3.5.1. ICE Gathering Overview . . . . . . . . . . . . . . . 11
3.5.2. ICE Candidate Trickling . . . . . . . . . . . . . . . 12
3.5.2.1. ICE Candidate Format . . . . . . . . . . . . . . 13
3.5.3. ICE Candidate Policy . . . . . . . . . . . . . . . . 13
3.5.4. ICE Candidate Pool . . . . . . . . . . . . . . . . . 14
3.6. Video Size Negotiation . . . . . . . . . . . . . . . . . 15
3.6.1. Creating an imageattr Attribute . . . . . . . . . . . 15
3.6.2. Interpreting an imageattr Attribute . . . . . . . . . 16
3.7. Simulcast . . . . . . . . . . . . . . . . . . . . . . . . 17
3.8. Interactions With Forking . . . . . . . . . . . . . . . . 18
3.8.1. Sequential Forking . . . . . . . . . . . . . . . . . 19
3.8.2. Parallel Forking . . . . . . . . . . . . . . . . . . 19
4. Interface . . . . . . . . . . . . . . . . . . . . . . . . . . 20
4.1. PeerConnection . . . . . . . . . . . . . . . . . . . . . 20
4.1.1. Constructor . . . . . . . . . . . . . . . . . . . . . 20
4.1.2. addTrack . . . . . . . . . . . . . . . . . . . . . . 22
4.1.3. removeTrack . . . . . . . . . . . . . . . . . . . . . 23
4.1.4. addTransceiver . . . . . . . . . . . . . . . . . . . 23
4.1.5. createDataChannel . . . . . . . . . . . . . . . . . . 23
4.1.6. createOffer . . . . . . . . . . . . . . . . . . . . . 23
4.1.7. createAnswer . . . . . . . . . . . . . . . . . . . . 24
4.1.8. SessionDescriptionType . . . . . . . . . . . . . . . 25
4.1.8.1. Use of Provisional Answers . . . . . . . . . . . 26
4.1.8.2. Rollback . . . . . . . . . . . . . . . . . . . . 27
4.1.9. setLocalDescription . . . . . . . . . . . . . . . . . 28
4.1.10. setRemoteDescription . . . . . . . . . . . . . . . . 28
4.1.11. currentLocalDescription . . . . . . . . . . . . . . . 29
4.1.12. pendingLocalDescription . . . . . . . . . . . . . . . 29
4.1.13. currentRemoteDescription . . . . . . . . . . . . . . 29
4.1.14. pendingRemoteDescription . . . . . . . . . . . . . . 29
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4.1.15. canTrickleIceCandidates . . . . . . . . . . . . . . . 30
4.1.16. setConfiguration . . . . . . . . . . . . . . . . . . 30
4.1.17. addIceCandidate . . . . . . . . . . . . . . . . . . . 31
4.2. RtpTransceiver . . . . . . . . . . . . . . . . . . . . . 32
4.2.1. stop . . . . . . . . . . . . . . . . . . . . . . . . 32
4.2.2. stopped . . . . . . . . . . . . . . . . . . . . . . . 32
4.2.3. setDirection . . . . . . . . . . . . . . . . . . . . 32
4.2.4. direction . . . . . . . . . . . . . . . . . . . . . . 32
4.2.5. currentDirection . . . . . . . . . . . . . . . . . . 33
4.2.6. setCodecPreferences . . . . . . . . . . . . . . . . . 33
5. SDP Interaction Procedures . . . . . . . . . . . . . . . . . 33
5.1. Requirements Overview . . . . . . . . . . . . . . . . . . 34
5.1.1. Usage Requirements . . . . . . . . . . . . . . . . . 34
5.1.2. Profile Names and Interoperability . . . . . . . . . 34
5.2. Constructing an Offer . . . . . . . . . . . . . . . . . . 35
5.2.1. Initial Offers . . . . . . . . . . . . . . . . . . . 35
5.2.2. Subsequent Offers . . . . . . . . . . . . . . . . . . 42
5.2.3. Options Handling . . . . . . . . . . . . . . . . . . 46
5.2.3.1. IceRestart . . . . . . . . . . . . . . . . . . . 46
5.2.3.2. VoiceActivityDetection . . . . . . . . . . . . . 46
5.3. Generating an Answer . . . . . . . . . . . . . . . . . . 47
5.3.1. Initial Answers . . . . . . . . . . . . . . . . . . . 47
5.3.2. Subsequent Answers . . . . . . . . . . . . . . . . . 53
5.3.3. Options Handling . . . . . . . . . . . . . . . . . . 54
5.3.3.1. VoiceActivityDetection . . . . . . . . . . . . . 55
5.4. Modifying an Offer or Answer . . . . . . . . . . . . . . 55
5.5. Processing a Local Description . . . . . . . . . . . . . 56
5.6. Processing a Remote Description . . . . . . . . . . . . . 56
5.7. Parsing a Session Description . . . . . . . . . . . . . . 57
5.7.1. Session-Level Parsing . . . . . . . . . . . . . . . . 57
5.7.2. Media Section Parsing . . . . . . . . . . . . . . . . 59
5.7.3. Semantics Verification . . . . . . . . . . . . . . . 61
5.8. Applying a Local Description . . . . . . . . . . . . . . 63
5.9. Applying a Remote Description . . . . . . . . . . . . . . 64
5.10. Applying an Answer . . . . . . . . . . . . . . . . . . . 68
6. Processing RTP/RTCP . . . . . . . . . . . . . . . . . . . . . 70
7. Examples . . . . . . . . . . . . . . . . . . . . . . . . . . 70
7.1. Simple Example . . . . . . . . . . . . . . . . . . . . . 71
7.2. Detailed Example . . . . . . . . . . . . . . . . . . . . 76
7.3. Early Transport Warmup Example . . . . . . . . . . . . . 85
8. Security Considerations . . . . . . . . . . . . . . . . . . . 93
9. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 94
10. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 94
11. References . . . . . . . . . . . . . . . . . . . . . . . . . 94
11.1. Normative References . . . . . . . . . . . . . . . . . . 94
11.2. Informative References . . . . . . . . . . . . . . . . . 98
Appendix A. Appendix A . . . . . . . . . . . . . . . . . . . . . 100
Appendix B. Change log . . . . . . . . . . . . . . . . . . . . . 101
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Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 110
1. Introduction
This document describes how the W3C WEBRTC RTCPeerConnection
interface [W3C.WD-webrtc-20140617] is used to control the setup,
management and teardown of a multimedia session.
1.1. General Design of JSEP
The thinking behind WebRTC call setup has been to fully specify and
control the media plane, but to leave the signaling plane up to the
application as much as possible. The rationale is that different
applications may prefer to use different protocols, such as the
existing SIP or Jingle call signaling protocols, or something custom
to the particular application, perhaps for a novel use case. In this
approach, the key information that needs to be exchanged is the
multimedia session description, which specifies the necessary
transport and media configuration information necessary to establish
the media plane.
With these considerations in mind, this document describes the
Javascript Session Establishment Protocol (JSEP) that allows for full
control of the signaling state machine from Javascript. As described
above, JSEP assumes a model in which a Javascript application
executes inside a runtime containing WebRTC APIs (the "JSEP
implementation"). The JSEP implementation is almost entirely
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
between two JSEP endpoints. Note though in many cases it will
actually be between a JSEP endpoint and some kind of server, such as
a gateway or MCU. This distinction is invisible to the JSEP
endpoint; it just follows the instructions it is given via the API.
JSEP's handling of session descriptions is simple and
straightforward. Whenever an offer/answer exchange is needed, 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
setLocalDescription() API. The offer is finally sent off to the
remote side over its preferred signaling mechanism (e.g.,
WebSockets); upon receipt of that offer, the remote party installs it
using the setRemoteDescription() API.
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To complete the offer/answer exchange, the remote party uses the
createAnswer() API to generate an appropriate answer, applies it
using the setLocalDescription() API, and sends the answer back to the
initiator over the signaling channel. When the initiator gets that
answer, it installs it using the setRemoteDescription() API, and
initial setup is complete. This process can be repeated for
additional offer/answer exchanges.
Regarding ICE [RFC5245], JSEP decouples the ICE state machine from
the overall signaling state machine, as the ICE state machine must
remain in the JSEP implementation, because only the implementation
has the necessary knowledge of candidates and other transport info.
Performing this separation also provides additional flexibility; in
protocols that decouple session descriptions from transport, such as
Jingle, the session description can be sent immediately and the
transport information can be sent when available. In protocols that
don't, such as SIP, the information can be used in the aggregated
form. Sending transport information separately can allow for faster
ICE and DTLS startup, since ICE checks can start as soon as any
transport information is available rather than waiting for all of it.
Through its abstraction of signaling, the JSEP approach does require
the application to be aware of the signaling process. While the
application does not need to understand the contents of session
descriptions to set up a call, the application must call the right
APIs at the right times, convert the session descriptions and ICE
information into the defined messages of its chosen signaling
protocol, and perform the reverse conversion on the messages it
receives from the other side.
One way to mitigate this is to provide a Javascript library that
hides this complexity from the developer; said library would
implement a given signaling protocol along with its state machine and
serialization code, presenting a higher level call-oriented interface
to the application developer. For example, libraries exist to adapt
the JSEP API into an API suitable for a SIP or XMPP. Thus, JSEP
provides greater control for the experienced developer without
forcing any additional complexity on the novice developer.
1.2. Other Approaches Considered
One approach that was considered instead of JSEP was to include a
lightweight signaling protocol. Instead of providing session
descriptions to the API, the API would produce and consume messages
from this protocol. While providing a more high-level API, this put
more control of signaling within the JSEP implementation, forcing it
to have to understand and handle concepts like signaling glare. In
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addition, it prevented the application from driving the state machine
to a desired state, as is needed in the page reload case.
A second approach that was considered but not chosen was to decouple
the management of the media control objects from session
descriptions, instead offering APIs that would control each component
directly. This was rejected based on a feeling that requiring
exposure of this level of complexity to the application programmer
would not be beneficial; it would result in an API where even a
simple example would require a significant amount of code to
orchestrate all the needed interactions, as well as creating a large
API surface that needed to be agreed upon and documented. 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
JSEP approach, which specifies how session descriptions are to be
evaluated and applied.
One variation on JSEP that was considered was to keep the basic
session description-oriented API, but to move the mechanism for
generating offers and answers out of the JSEP implementation.
Instead of providing createOffer/createAnswer methods within the
implementation, this approach would instead expose a getCapabilities
API which would provide the application with the information it
needed in order to generate its own session descriptions. This
increases the amount of work that the application needs to do; it
needs to know how to generate session descriptions from capabilities,
and especially how to generate the correct answer from an arbitrary
offer and the supported capabilities. While this could certainly be
addressed by using a library like the one mentioned above, it
basically forces the use of said library even for a simple example.
Providing createOffer/createAnswer avoids this problem.
2. Terminology
The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
"SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
document are to be interpreted as described in [RFC2119].
3. Semantics and Syntax
3.1. Signaling Model
JSEP does not specify a particular signaling model or state machine,
other than the generic need to exchange session descriptions in the
fashion described by [RFC3264] (offer/answer) in order for both sides
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
a session. However, the JSEP implementation is totally decoupled
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from the actual mechanism by which these offers and answers are
communicated to the remote side, including addressing,
retransmission, forking, and glare handling. These issues are left
entirely up to the application; the application has complete control
over which offers and answers get handed to the implementation, and
when.
+-----------+ +-----------+
| Web App |<--- App-Specific Signaling -->| Web App |
+-----------+ +-----------+
^ ^
| SDP | SDP
V V
+-----------+ +-----------+
| JSEP |<----------- Media ------------>| JSEP |
| Impl. | | Impl. |
+-----------+ +-----------+
Figure 1: JSEP Signaling Model
3.2. Session Descriptions and State Machine
In order to establish the media plane, the user agent needs specific
parameters to indicate what to transmit to the remote side, as well
as how to handle the media that is received. These parameters are
determined by the exchange of session descriptions in offers and
answers, and there are certain details to this process that must be
handled in the JSEP APIs.
Whether a session description applies to the local side or the remote
side affects the meaning of that description. For example, the list
of codecs sent to a remote party indicates what the local side is
willing to receive, which, when intersected with the set of codecs
the remote side supports, specifies what the remote side should send.
However, not all parameters follow this rule; for example, the
fingerprints [I-D.ietf-mmusic-4572-update] sent to a remote party are
calculated based on the local certificate(s) offered; the remote
party MUST either accept these parameters or reject them altogether,
with no option to choose different values.
In addition, various RFCs put different conditions on the format of
offers versus answers. For example, an offer may propose an
arbitrary number of m= sections (i.e., media descriptions as
described in [RFC4566], Section 5.14), but an answer must contain the
exact same number as the offer.
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Lastly, while the exact media parameters are only known only after an
offer and an answer have been exchanged, it is possible for the
offerer to receive media after they have sent an offer and before
they have received an answer. To properly process incoming media in
this case, the offerer's media handler must be aware of the details
of the offer before the answer arrives.
Therefore, in order to handle session descriptions properly, the user
agent needs:
1. To know if a session description pertains to the local or remote
side.
2. To know if a session description is an offer or an answer.
3. To allow the offer to be specified independently of the answer.
JSEP addresses this by adding both setLocalDescription and
setRemoteDescription methods and having session description objects
contain a type field indicating the type of session description being
supplied. This satisfies the requirements listed above for both the
offerer, who first calls setLocalDescription(sdp [offer]) and then
later setRemoteDescription(sdp [answer]), as well as for the
answerer, who first calls setRemoteDescription(sdp [offer]) and then
later setLocalDescription(sdp [answer]).
JSEP also allows for an answer to be treated as provisional by the
application. Provisional answers provide a way for an answerer to
communicate initial session parameters back to the offerer, in order
to allow the session to begin, while allowing a final answer to be
specified later. This concept of a final answer is important to the
offer/answer model; when such an answer is received, any extra
resources allocated by the caller can be released, now that the exact
session configuration is known. These "resources" can include things
like extra ICE components, TURN candidates, or video decoders.
Provisional answers, on the other hand, do no such deallocation; as a
result, multiple dissimilar provisional answers, with their own codec
choices, transport parameters, etc., can be received and applied
during call setup. Note that the final answer itself may be
different than any received provisional answers.
In [RFC3264], the constraint at the signaling level is that only one
offer can be outstanding for a given session, but at the media stack
level, a new offer can be generated at any point. For example, when
using SIP for signaling, if one offer is sent, then cancelled using a
SIP CANCEL, another offer can be generated even though no answer was
received for the first offer. To support this, the JSEP media layer
can provide an offer via the createOffer() method whenever the
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Javascript application needs one for the signaling. The answerer can
send back zero or more provisional answers, and finally end the
offer-answer exchange by sending a final answer. The state machine
for this is as follows:
setRemote(OFFER) setLocal(PRANSWER)
/-----\ /-----\
| | | |
v | v |
+---------------+ | +---------------+ |
| |----/ | |----/
| | setLocal(PRANSWER) | |
| Remote-Offer |------------------- >| Local-Pranswer|
| | | |
| | | |
+---------------+ +---------------+
^ | |
| | setLocal(ANSWER) |
setRemote(OFFER) | |
| V setLocal(ANSWER) |
+---------------+ |
| | |
| |<---------------------------+
| Stable |
| |<---------------------------+
| | |
+---------------+ setRemote(ANSWER) |
^ | |
| | setLocal(OFFER) |
setRemote(ANSWER) | |
| V |
+---------------+ +---------------+
| | | |
| | setRemote(PRANSWER) | |
| Local-Offer |------------------- >|Remote-Pranswer|
| | | |
| |----\ | |----\
+---------------+ | +---------------+ |
^ | ^ |
| | | |
\-----/ \-----/
setLocal(OFFER) setRemote(PRANSWER)
Figure 2: JSEP State Machine
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Aside from these state transitions there is no other difference
between the handling of provisional ("pranswer") and final ("answer")
answers.
3.3. Session Description Format
JSEP's session descriptions use SDP syntax for their internal
representation. While this format is not optimal for manipulation
from Javascript, it is widely accepted, and frequently updated with
new features; any alternate encoding of session descriptions would
have to keep pace with the changes to SDP, at least until the time
that this new encoding eclipsed SDP in popularity.
However, to simplify Javascript processing, and provide for future
flexibility, the SDP syntax is encapsulated within a
SessionDescription object, which can be constructed from SDP, and be
serialized out to SDP. If future specifications agree on a JSON
format for session descriptions, we could easily enable this object
to generate and consume that JSON.
Other methods may be added to SessionDescription in the future to
simplify handling of SessionDescriptions from Javascript. In the
meantime, Javascript libraries can be used to perform these
manipulations.
Note that most applications should be able to treat the
SessionDescriptions produced and consumed by these various API calls
as opaque blobs; that is, the application will not need to read or
change them.
3.4. Session Description Control
In order to give the application control over various common session
parameters, JSEP provides control surfaces which tell the JSEP
implementation how to generate session descriptions. This avoids the
need for Javascript to modify session descriptions in most cases.
Changes to these objects result in changes to the session
descriptions generated by subsequent createOffer/Answer calls.
3.4.1. RtpTransceivers
RtpTransceivers allow the application to control the RTP media
associated with one m= section. Each RtpTransceiver has an RtpSender
and an RtpReceiver, which an application can use to control the
sending and receiving of RTP media. The application may also modify
the RtpTransceiver directly, for instance, by stopping it.
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RtpTransceivers generally have a 1:1 mapping with m= sections,
although there may be more RtpTransceivers than m= sections when
RtpTransceivers are created but not yet associated with a m= section,
or if RtpTransceivers have been stopped and disassociated from m=
sections. An RtpTransceiver is said to be associated with an m=
section if its mid property is non-null; otherwise it is said to be
disassociated. The associated m= section is determined using a
mapping between transceivers and m= section indices, formed when
creating an offer or applying a remote offer. An RtpTransceiver is
never associated with more than one m= section, and once a session
description is applied, a m= section is always associated with
exactly one RtpTransceiver.
RtpTransceivers can be created explicitly by the application or
implicitly by calling setRemoteDescription with an offer that adds
new m= sections.
3.4.2. RtpSenders
RtpSenders allow the application to control how RTP media is sent.
An RtpSender is conceptually responsible for the outgoing RTP
stream(s) described by an m= section. This includes encoding the
attached MediaStreamTrack, sending RTP media packets, and generating/
processing RTCP for the outgoing RTP streams(s).
3.4.3. RtpReceivers
RtpReceivers allow the application to inspect how RTP media is
received. An RtpReceiver is conceptually responsible for the
incoming RTP stream(s) described by an m= section. This includes
processing received RTP media packets, decoding the incoming
stream(s) to produce a remote MediaStreamTrack, and generating/
processing RTCP for the incoming RTP stream(s).
3.5. ICE
3.5.1. ICE Gathering Overview
JSEP gathers ICE candidates as needed by the application. Collection
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
the local session description, or new ICE credentials in the
description, indicating an ICE restart. Use of new ICE credentials
can be triggered explicitly by the application, or implicitly by the
JSEP implementation in response to changes in the ICE configuration.
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
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set, calls to the createOffer API will generate new ICE credentials.
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
a local- or remote-initiated ICE restart.
When a new gathering phase starts, the ICE agent will notify the
application that gathering is occurring through an event. Then, when
each new ICE candidate becomes available, the ICE agent will supply
it to the application via an additional event; these candidates will
also automatically be added to the current and/or pending local
session description. Finally, when all candidates have been
gathered, an event will be dispatched to signal that the gathering
process is complete.
Note that gathering phases only gather the candidates needed by
new/recycled/restarting m= sections; other m= sections continue to
use their existing candidates. Also, when bundling is active,
candidates are only gathered (and exchanged) for the m= sections
referenced in BUNDLE-tags, as described in
[I-D.ietf-mmusic-sdp-bundle-negotiation].
3.5.2. ICE Candidate Trickling
Candidate trickling is a technique through which a caller may
incrementally provide candidates to the callee after the initial
offer has been dispatched; the semantics of "Trickle ICE" are defined
in [I-D.ietf-ice-trickle]. This process allows the callee to begin
acting upon the call and setting up the ICE (and perhaps DTLS)
connections immediately, without having to wait for the caller to
gather all possible candidates. This results in faster media setup
in cases where gathering is not performed prior to initiating the
call.
JSEP supports optional candidate trickling by providing APIs, as
described above, that provide control and feedback on the ICE
candidate gathering process. Applications that support candidate
trickling can send the initial offer immediately and send individual
candidates when they get the notified of a new candidate;
applications that do not support this feature can simply wait for the
indication that gathering is complete, and then create and send their
offer, with all the candidates, at this time.
Upon receipt of trickled candidates, the receiving application will
supply them to its ICE agent. This triggers the ICE agent to start
using the new remote candidates for connectivity checks.
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3.5.2.1. ICE Candidate Format
In JSEP, ICE candidates are abstracted by an IceCandidate object, and
as with session descriptions, SDP syntax is used for the internal
representation.
The candidate details are specified in an IceCandidate field, using
the same SDP syntax as the "candidate-attribute" field defined in
[RFC5245], Section 15.1. For example:
candidate:1 1 UDP 1694498815 192.0.2.33 10000 typ host
The IceCandidate object contains a field to indicate which ICE ufrag
it is associated with, as defined in [RFC5245], Section 15.4. This
value is used to determine which session description (and thereby
which gathering phase) this IceCandidate belongs to, which helps
resolve ambiguities during ICE restarts. If this field is absent in
a received IceCandidate (perhaps when communicating with a non-JSEP
endpoint), the most recently received session description is assumed.
The IceCandidate object also contains fields to indicate which m=
section it is associated with, which can be identified in one of two
ways, either by a m= section index, or a MID. The m= section index
is a zero-based index, with index N referring to the N+1th m= section
in the session description referenced by this IceCandidate. The MID
is a "media stream identification" value, as defined in [RFC5888],
Section 4, which provides a more robust way to identify the m=
section in the session description, using the MID of the associated
RtpTransceiver object (which may have been locally generated by the
answerer when interacting with a non-JSEP endpoint that does not
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
for identification; otherwise, the m= section index is used instead.
When creating an IceCandidate object, JSEP implementations MUST
populate all of these fields.
3.5.3. ICE Candidate Policy
Typically, when gathering ICE candidates, the JSEP implementation
will gather all possible forms of initial candidates - host, server
reflexive, and relay. However, in certain cases, applications may
want to have more specific control over the gathering process, due to
privacy or related concerns. For example, one may want to only use
relay candidates, to leak as little location information as possible
(keeping in mind that this choice comes with corresponding
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operational costs). To accomplish this, JSEP allows the application
to restrict which ICE candidates are used in a session. Note that
this filtering is applied on top of any restrictions the
implementation chooses to enforce regarding which IP addresses are
permitted for the application, as discussed in
[I-D.ietf-rtcweb-ip-handling].
There may also be cases where the application wants to change which
types of candidates are used while the session is active. A prime
example is where a callee may initially want to use only relay
candidates, to avoid leaking location information to an arbitrary
caller, but then change to use all candidates (for lower operational
cost) once the user has indicated they want to take the call. For
this scenario, the JSEP implementation MUST allow the candidate
policy to be changed in mid-session, subject to the aforementioned
interactions with local policy.
To administer the ICE candidate policy, the JSEP implementation will
determine the current setting at the start of each gathering phase.
Then, during the gathering phase, the implementation MUST NOT expose
candidates disallowed by the current policy to the application, use
them as the source of connectivity checks, or indirectly expose them
via other fields, such as the raddr/rport attributes for other ICE
candidates. Later, if a different policy is specified by the
application, the application can apply it by kicking off a new
gathering phase via an ICE restart.
3.5.4. ICE Candidate Pool
JSEP applications typically inform the JSEP implementation to begin
ICE gathering via the information supplied to setLocalDescription, as
this is where the app specifies the number of media streams, and
thereby ICE components, for which to gather candidates. However, to
accelerate cases where the application knows the number of ICE
components to use ahead of time, it may ask the implementation to
gather a pool of potential ICE candidates to help ensure rapid media
setup.
When setLocalDescription is eventually called, and the JSEP
implementation goes to gather the needed ICE candidates, it SHOULD
start by checking if any candidates are available in the pool. If
there are candidates in the pool, they SHOULD be handed to the
application immediately via the ICE candidate event. If the pool
becomes depleted, either because a larger-than-expected number of ICE
components is used, or because the pool has not had enough time to
gather candidates, the remaining candidates are gathered as usual.
This only occurs for the first offer/answer exchange, after which the
candidate pool is emptied and no longer used.
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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
minimize the time it takes to establish connectivity, to avoid
clipping of initial media. By pre-gathering candidates into the
pool, it can exchange and start sending connectivity checks from
these candidates almost immediately upon receipt of a call. Note
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 consume resources on the STUN/TURN servers it is using.
3.6. Video Size Negotiation
Video size negotiation is the process through which a receiver can
use the "a=imageattr" SDP attribute [RFC6236] to indicate what video
frame sizes it is capable of receiving. A receiver may have hard
limits on what its video decoder can process, or it may have some
maximum set by policy.
Note that certain codecs support transmission of samples with aspect
ratios other than 1.0 (i.e., non-square pixels). JSEP
implementations will not transmit non-square pixels, but SHOULD
receive and render such video with the correct aspect ratio.
However, sample aspect ratio has no impact on the size negotiation
described below; all dimensions are measured in pixels, whether
square or not.
3.6.1. Creating an imageattr Attribute
The receiver will first intersect any known local limits (e.g.,
hardware decoder capababilities, local policy) to determine the
absolute minimum and maximum sizes it can receive. If there are no
known local limits, the "a=imageattr" attribute SHOULD be omitted.
Otherwise, an "a=imageattr" attribute is created with "recv"
direction, and the resulting resolution space formed from the
aforementioned intersection is used to specify its minimum and
maximum x= and y= values. If the intersection is the null set, i.e.,
the degenerate case of no permitted resolutions, this MUST be
represented by x=0 and y=0 values.
The rules here express a single set of preferences, and therefore,
the "a=imageattr" q= value is not important. It SHOULD be set to
1.0.
The "a=imageattr" field is payload type specific. When all video
codecs supported have the same capabilities, use of a single
attribute, with the wildcard payload type (*), is RECOMMENDED.
However, when the supported video codecs have different limitations,
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specific "a=imageattr" attributes MUST be inserted for each payload
type.
As an example, consider a system with a multiformat video decoder,
which is capable of decoding any resolution from 48x48 to 720p, In
this case, the implementation would generate this attribute:
a=imageattr:* recv [x=[48:1280],y=[48:720],q=1.0]
This declaration indicates that the receiver is capable of decoding
any image resolution from 48x48 up to 1280x720 pixels.
3.6.2. Interpreting an imageattr Attribute
[RFC6236] defines "a=imageattr" to be an advisory field. This means
that it does not absolutely constrain the video formats that the
sender can use, but gives an indication of the preferred values.
This specification prescribes more specific behavior. When a sender
of a given MediaStreamTrack, which is producing video of a certain
resolution, receives an "a=imageattr recv" attribute, it MUST check
to see if the original resolution meets the size criteria specified
in the attribute, and adapt the resolution accordingly by scaling (if
appropriate). Note that when considering a MediaStreamTrack that is
producing rotated video, the unrotated resolution MUST be used. This
is required regardless of whether the receiver supports performing
receive-side rotation (e.g., through CVO [TS26.114]), as it
significantly simplifies the matching logic.
For the purposes of resolution negotiation, only size limits are
considered. Any other values, e.g. picture or sample aspect ratio,
MUST be ignored.
When communicating with a non-JSEP endpoint, multiple relevant
"a=imageattr recv" attributes may be present in a received m=
section. If this occurs, attributes other than the one with the
highest "q=" value MUST be ignored. If multiple attributes have the
same "q=" value, those that appear after the first such attribute in
the m= section MUST be ignored.
If an "a=imageattr recv" attribute references a different video
payload type than what has been selected for sending the
MediaStreamTrack, it MUST be ignored.
If the original resolution matches the size limits in the attribute,
the track MUST be transmitted untouched.
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If the original resolution exceeds the size limits in the attribute,
the sender SHOULD apply downscaling to the output of the
MediaStreamTrack in order to satisfy the limits. Downscaling MUST
NOT change the track aspect ratio.
If the original resolution is less than the size limits in the
attribute, upscaling is needed, but this may not be appropriate in
all cases. To address this concern, the application can set an
upscaling policy for each sent track. For this case, if upscaling is
permitted by policy, the sender SHOULD apply upscaling in order to
provide the desired resolution. Otherwise, the sender MUST NOT apply
upscaling. The sender SHOULD NOT upscale in other cases, even if the
policy permits it. Upscaling MUST NOT change the track aspect ratio.
If there is no appropriate and permitted scaling mechanism that
allows the received size limits to be satisfied, the sender MUST NOT
transmit the track.
If the attribute includes a "sar=" (sample aspect ratio) value set to
something other than "1.0", indicating the receiver wants to receive
non-square pixels, this cannot be satisfied and the sender MUST NOT
transmit the track.
In the special case of receiving a maximum resolution of [0, 0], as
described above, the sender MUST NOT transmit the track.
3.7. Simulcast
JSEP supports simulcast transmission of a MediaStreamTrack, where
multiple encodings of the source media can be transmitted within the
context of a single m= section. The current JSEP API is designed to
allow applications to send simulcasted media but only to receive a
single encoding. This allows for multi-user scenarios where each
sending client sends multiple encodings to a server, which then, for
each receiving client, chooses the appropriate encoding to forward.
Applications request support for simulcast by configuring multiple
encodings on an RtpSender, which, upon generation of an offer or
answer, are indicated in SDP markings on the corresponding m=
section, as described below. Receivers that understand simulcast and
are willing to receive it will also include SDP markings to indicate
their support, and JSEP endpoints will use these markings to
determine whether simulcast is permitted for a given RtpSender. If
simulcast support is not negotiated, the RtpSender will only use the
first configured encoding.
Note that the exact simulcast parameters are up to the sending
application. While the aforementioned SDP markings are provided to
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ensure the remote side can receive and demux multiple simulcast
encodings, the specific resolutions and bitrates to be used for each
encoding are purely a send-side decision in JSEP.
JSEP currently does not provide a mechanism to configure receipt of
simulcast. This means that if simulcast is offered by the remote
endpoint, the answer generated by a JSEP endpoint will not indicate
support for receipt of simulcast, and as such the remote endpoint
will only send a single encoding per m= section.
In addition, JSEP does not provide a mechanism to handle an incoming
offer requesting simulcast from the JSEP endpoint. This means that
established simulcast streams will continue to work through a
received re-offer, but setting up initial simulcast by way of a
received offer requires out-of-band signaling or SDP inspection.
Future versions of this specification may add additional APIs to
provide direct control.
When using JSEP to transmit multiple encodings from a RtpSender, the
techniques from [I-D.ietf-mmusic-sdp-simulcast] and
[I-D.ietf-mmusic-rid] are used. Specifically, when multiple
encodings have been configured for a RtpSender, the m= section for
the RtpSender will include an "a=simulcast" attribute, as defined in
[I-D.ietf-mmusic-sdp-simulcast], Section 6.2, with a "send" simulcast
stream description that lists each desired encoding, and no "recv"
simulcast stream description. The m= section will also include an
"a=rid" attribute for each encoding, as specified in
[I-D.ietf-mmusic-rid], Section 4; the use of RID identifiers allows
the individual encodings to be disambiguated even though they are all
part of the same m= section.
3.8. Interactions With Forking
Some call signaling systems allow various types of forking where an
SDP Offer may be provided to more than one device. For example, SIP
[RFC3261] defines both a "Parallel Search" and "Sequential Search".
Although these are primarily signaling level issues that are outside
the scope of JSEP, they do have some impact on the configuration of
the media plane that is relevant. When forking happens at the
signaling layer, the Javascript application responsible for the
signaling needs to make the decisions about what media should be sent
or received at any point of time, as well as which remote endpoint it
should communicate with; JSEP is used to make sure the media engine
can make the RTP and media perform as required by the application.
The basic operations that the applications can have the media engine
do are:
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o Start exchanging media with a given remote peer, but keep all the
resources reserved in the offer.
o Start exchanging media with a given remote peer, and free any
resources in the offer that are not being used.
3.8.1. Sequential Forking
Sequential forking involves a call being dispatched to multiple
remote callees, where each callee can accept the call, but only one
active session ever exists at a time; no mixing of received media is
performed.
JSEP handles sequential forking well, allowing the application to
easily control the policy for selecting the desired remote endpoint.
When an answer arrives from one of the callees, the application can
choose to apply it either as a provisional answer, leaving open the
possibility of using a different answer in the future, or apply it as
a final answer, ending the setup flow.
In a "first-one-wins" situation, the first answer will be applied as
a final answer, and the application will reject any subsequent
answers. In SIP parlance, this would be ACK + BYE.
In a "last-one-wins" situation, all answers would be applied as
provisional answers, and any previous call leg will be terminated.
At some point, the application will end the setup process, perhaps
with a timer; at this point, the application could reapply the
pending remote description as a final answer.
3.8.2. Parallel Forking
Parallel forking involves a call being dispatched to multiple remote
callees, where each callee can accept the call, and multiple
simultaneous active signaling sessions can be established as a
result. If multiple callees send media at the same time, the
possibilities for handling this are described in Section 3.1 of
[RFC3960]. Most SIP devices today only support exchanging media with
a single device at a time, and do not try to mix multiple early media
audio sources, as that could result in a confusing situation. For
example, consider having a European ringback tone mixed together with
the North American ringback tone - the resulting sound would not be
like either tone, and would confuse the user. If the signaling
application wishes to only exchange media with one of the remote
endpoints at a time, then from a media engine point of view, this is
exactly like the sequential forking case.
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In the parallel forking case where the Javascript application wishes
to simultaneously exchange media with multiple peers, the flow is
slightly more complex, but the Javascript application can follow the
strategy that [RFC3960] describes using UPDATE. The UPDATE approach
allows the signaling to set up a separate media flow for each peer
that it wishes to exchange media with. In JSEP, this offer used in
the UPDATE would be formed by simply creating a new PeerConnection
and making sure that the same local media streams have been added
into this new PeerConnection. Then the new PeerConnection object
would produce a SDP offer that could be used by the signaling to
perform the UPDATE strategy discussed in [RFC3960].
As a result of sharing the media streams, the application will end up
with N parallel PeerConnection sessions, each with a local and remote
description and their own local and remote addresses. The media flow
from these sessions can be managed using setDirection (see
Section 4.2.3), or the application can choose to play out the media
from all sessions mixed together. Of course, if the application
wants to only keep a single session, it can simply terminate the
sessions that it no longer needs.
4. Interface
This section details the basic operations that must be present to
implement JSEP functionality. The actual API exposed in the W3C API
may have somewhat different syntax, but should map easily to these
concepts.
4.1. PeerConnection
4.1.1. Constructor
The PeerConnection constructor allows the application to specify
global parameters for the media session, such as the STUN/TURN
servers and credentials to use when gathering candidates, as well as
the initial ICE candidate policy and pool size, and also the bundle
policy to use.
If an ICE candidate policy is specified, it functions as described in
Section 3.5.3, causing the JSEP implementation to only surface the
permitted candidates (including any implementation-internal
filtering) to the application, and only use those candidates for
connectivity checks. The set of available policies is as follows:
all: All candidates permitted by implementation policy will be
gathered and used.
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relay: All candidates except relay candidates will be filtered out.
This obfuscates the location information that might be ascertained
by the remote peer from the received candidates. Depending on how
the application deploys and chooses relay servers, this could
obfuscate location to a metro or possibly even global level.
The default ICE candidate policy MUST be set to "all" as this is
generally the desired policy, and also typically reduces use of
application TURN server resources significantly.
If a size is specified for the ICE candidate pool, this indicates the
number of ICE components to pre-gather candidates for. Because pre-
gathering results in utilizing STUN/TURN server resources for
potentially long periods of time, this must only occur upon
application request, and therefore the default candidate pool size
MUST be zero.
The application can specify its preferred policy regarding use of
bundle, the multiplexing mechanism defined in
[I-D.ietf-mmusic-sdp-bundle-negotiation]. Regardless of policy, the
application will always try to negotiate bundle onto a single
transport, and will offer a single bundle group across all m=
sections; use of this single transport is contingent upon the
answerer accepting bundle. However, by specifying a policy from the
list below, the application can control exactly how aggressively it
will try to bundle media streams together, which affects how it will
interoperate with a non-bundle-aware endpoint. When negotiating with
a non-bundle-aware endpoint, only the streams not marked as bundle-
only streams will be established.
The set of available policies is as follows:
balanced: The first m= section of each type (audio, video, or
application) will contain transport parameters, which will allow
an answerer to unbundle that section. The second and any
subsequent m= section of each type will be marked bundle-only.
The result is that if there are N distinct media types, then
candidates will be gathered for for N media streams. This policy
balances desire to multiplex with the need to ensure basic audio
and video can still be negotiated in legacy cases. When acting as
answerer, if there is no bundle group in the offer, the
implementation will reject all but the first m= section of each
type.
max-compat: All m= sections will contain transport parameters; none
will be marked as bundle-only. This policy will allow all streams
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to be received by non-bundle-aware endpoints, but require separate
candidates to be gathered for each media stream.
max-bundle: Only the first m= section will contain transport
parameters; all streams other than the first will be marked as
bundle-only. This policy aims to minimize candidate gathering and
maximize multiplexing, at the cost of less compatibility with
legacy endpoints. When acting as answerer, the implementation
will reject any m= sections other than the first m= section,
unless they are in the same bundle group as that m= section.
As it provides the best tradeoff between performance and
compatibility with legacy endpoints, the default bundle policy MUST
be set to "balanced".
The application can specify its preferred policy regarding use of
RTP/RTCP multiplexing [RFC5761] using one of the following policies:
negotiate: The JSEP implementation will gather both RTP and RTCP
candidates but also will offer "a=rtcp-mux", thus allowing for
compatibility with either multiplexing or non-multiplexing
endpoints.
require: The JSEP implementation will only gather RTP candidates and
will insert an "a=rtcp-mux-only" indication into any new m=
sections in offers it generates. This halves the number of
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".
Implementations MAY choose to reject attempts by the application to
set the multiplexing policy to "negotiate".
4.1.2. addTrack
The addTrack method adds a MediaStreamTrack to the PeerConnection,
using the MediaStream argument to associate the track with other
tracks in the same MediaStream, so that they can be added to the same
"LS" group when creating an offer or answer. addTrack attempts to
minimize the number of transceivers as follows: If the PeerConnection
is in the "have-remote-offer" state, the track will be attached to
the first compatible transceiver that was created by the most recent
call to setRemoteDescription() and does not have a local track.
Otherwise, a new transceiver will be created, as described in
Section 4.1.4.
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4.1.3. removeTrack
The removeTrack method removes a MediaStreamTrack from the
PeerConnection, using the RtpSender argument to indicate which sender
should have its track removed. The sender's track is cleared, and
the sender stops sending. Future calls to createOffer will mark the
m= section associated with the sender as recvonly (if
transceiver.currentDirection is sendrecv) or as inactive (if
transceiver.currentDirection is sendonly).
4.1.4. addTransceiver
The addTransceiver method adds a new RtpTransceiver to the
PeerConnection. If a MediaStreamTrack argument is provided, then the
transceiver will be configured with that media type and the track
will be attached to the transceiver. Otherwise, the application MUST
explicitly specify the type; this mode is useful for creating
recvonly transceivers as well as for creating transceivers to which a
track can be attached at some later point.
At the time of creation, the application can also specify a
transceiver direction attribute, a set of MediaStreams which the
transceiver is associated with (allowing LS group assignments), and a
set of encodings for the media (used for simulcast as described in
Section 3.7).
4.1.5. createDataChannel
The createDataChannel method creates a new data channel and attaches
it to the PeerConnection. If no data channel currently exists for
this PeerConnection, then a new offer/answer exchange is required.
All data channels on a given PeerConnection share the same SCTP/DTLS
association and therefore the same m= section, so subsequent creation
of data channels does not have any impact on the JSEP state.
The createDataChannel method also includes a number of arguments
which are used by the PeerConnection (e.g., maxPacketLifetime) but
are not reflected in the SDP and do not affect the JSEP state.
4.1.6. createOffer
The createOffer method generates a blob of SDP that contains a
[RFC3264] offer with the supported configurations for the session,
including descriptions of the media added to this PeerConnection, the
codec/RTP/RTCP options supported by this implementation, and any
candidates that have been gathered by the ICE agent. An options
parameter may be supplied to provide additional control over the
generated offer. This options parameter allows an application to
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trigger an ICE restart, for the purpose of reestablishing
connectivity.
In the initial offer, the generated SDP will contain all desired
functionality for the session (functionality that is supported but
not desired by default may be omitted); for each SDP line, the
generation of the SDP will follow the process defined for generating
an initial offer from the document that specifies the given SDP line.
The exact handling of initial offer generation is detailed in
Section 5.2.1 below.
In the event createOffer is called after the session is established,
createOffer will generate an offer to modify the current session
based on any changes that have been made to the session, e.g., adding
or stopping RtpTransceivers, or requesting an ICE restart. For each
existing stream, the generation of each SDP line must follow the
process defined for generating an updated offer from the RFC that
specifies the given SDP line. For each new stream, the generation of
the SDP must follow the process of generating an initial offer, as
mentioned above. If no changes have been made, or for SDP lines that
are unaffected by the requested changes, the offer will only contain
the parameters negotiated by the last offer-answer exchange. The
exact handling of subsequent offer generation is detailed in
Section 5.2.2. below.
Session descriptions generated by createOffer must be immediately
usable by setLocalDescription; if a system has limited resources
(e.g. a finite number of decoders), createOffer should return an
offer that reflects the current state of the system, so that
setLocalDescription will succeed when it attempts to acquire those
resources.
Calling this method may do things such as generating new ICE
credentials, but does not result in candidate gathering, or cause
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
The createAnswer method generates a blob of SDP that contains a
[RFC3264] SDP answer with the supported configuration for the session
that is compatible with the parameters supplied in the most recent
call to setRemoteDescription, which MUST have been called prior to
calling createAnswer. Like createOffer, the returned blob contains
descriptions of the media added to this PeerConnection, the
codec/RTP/RTCP options negotiated for this session, and any
candidates that have been gathered by the ICE agent. An options
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parameter may be supplied to provide additional control over the
generated answer.
As an answer, the generated SDP will contain a specific configuration
that specifies how the media plane should be established; for each
SDP line, the generation of the SDP must follow the process defined
for generating an answer from the document that specifies the given
SDP line. The exact handling of answer generation is detailed in
Section 5.3. below.
Session descriptions generated by createAnswer must be immediately
usable by setLocalDescription; like createOffer, the returned
description should reflect the current state of the system.
Calling this method may do things such as generating new ICE
credentials, but does not trigger candidate gathering or cause a
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
Session description objects (RTCSessionDescription) may be of type
"offer", "pranswer", "answer" or "rollback". These types provide
information as to how the description parameter should be parsed, and
how the media state should be changed.
"offer" indicates that a description should be parsed as an offer;
said description may include many possible media configurations. A
description used as an "offer" may be applied anytime the
PeerConnection is in a stable state, or as an update to a previously
supplied but unanswered "offer".
"pranswer" indicates that a description should be parsed as an
answer, but not a final answer, and so should not result in the
freeing of allocated resources. It may result in the start of media
transmission, if the answer does not specify an inactive media
direction. A description used as a "pranswer" may be applied as a
response to an "offer", or an update to a previously sent "pranswer".
"answer" indicates that a description should be parsed as an answer,
the offer-answer exchange should be considered complete, and any
resources (decoders, candidates) that are no longer needed can be
released. A description used as an "answer" may be applied as a
response to an "offer", or an update to a previously sent "pranswer".
The only difference between a provisional and final answer is that
the final answer results in the freeing of any unused resources that
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were allocated as a result of the offer. As such, the application
can use some discretion on whether an answer should be applied as
provisional or final, and can change the type of the session
description as needed. For example, in a serial forking scenario, an
application may receive multiple "final" answers, one from each
remote endpoint. The application could choose to accept the initial
answers as provisional answers, and only apply an answer as final
when it receives one that meets its criteria (e.g. a live user
instead of voicemail).
"rollback" is a special session description type implying that the
state machine should be rolled back to the previous stable state, as
described in Section 4.1.8.2. The contents MUST be empty.
4.1.8.1. Use of Provisional Answers
Most applications will not need to create answers using the
"pranswer" type. While it is good practice to send an immediate
response to an offer, in order to warm up the session transport and
prevent media clipping, the preferred handling for a JSEP application
is to create and send a "sendonly" final answer with a null
MediaStreamTrack immediately after receiving the offer, which will
prevent media from being sent by the caller, and allow media to be
sent immediately upon answer by the callee. Later, when the callee
actually accepts the call, the application can plug in the real
MediaStreamTrack and create a new "sendrecv" offer to update the
previous offer/answer pair and start bidirectional media flow. While
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 JSEP application that wants to set
up audio and video as quickly as possible. When the callee receives
an offer with audio and video MediaStreamTracks, it will send an
immediate answer accepting these tracks as sendonly (meaning that the
caller will not send the callee any media yet, and because the callee
has not yet added its own MediaStreamTracks, the callee will not send
any media either). It will then ask the user to accept the call and
acquire the needed local tracks. Upon acceptance by the user, the
application will plug in the tracks it has acquired, which, because
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.
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Of course, some applications may not be able to perform this double
offer-answer exchange, particularly ones that are attempting to
gateway to legacy signaling protocols. In these cases, pranswer can
still provide the application with a mechanism to warm up the
transport.
4.1.8.2. Rollback
In certain situations it may be desirable to "undo" a change made to
setLocalDescription or setRemoteDescription. Consider a case where a
call is ongoing, and one side wants to change some of the session
parameters; that side generates an updated offer and then calls
setLocalDescription. However, the remote side, either before or
after setRemoteDescription, decides it does not want to accept the
new parameters, and sends a reject message back to the offerer. Now,
the offerer, and possibly the answerer as well, need to return to a
stable state and the previous local/remote description. To support
this, we introduce the concept of "rollback".
A rollback discards any proposed changes to the session, returning
the state machine to the stable state, and setting the pending local
and/or remote description (see Section 4.1.12 and Section 4.1.14) to
null. Any resources or candidates that were allocated by the
abandoned local description are discarded; any media that is received
will be processed according to the previous local and remote
descriptions. Rollback can only be used to cancel proposed changes;
there is no support for rolling back from a stable state to a
previous stable state. Note that this implies that once the answerer
has performed setLocalDescription with his answer, this cannot be
rolled back.
A rollback will disassociate any RtpTransceivers that were associated
with m= sections by the application of the rolled-back session
description (see Section 5.9 and Section 5.8). This means that some
RtpTransceivers that were previously associated will no longer be
associated with any m= section; in such cases, the value of the
RtpTransceiver's mid property MUST be set to null, and the mapping
between the transceiver and its m= section index MUST be discarded.
RtpTransceivers that were created by applying a remote offer that was
subsequently rolled back MUST be stopped and removed from the
PeerConnection. However, a RtpTransceiver MUST NOT be removed if a
track was attached to the RtpTransceiver via the addTrack method.
This is so that an application may call addTrack, then call
setRemoteDescription with an offer, then roll back that offer, then
call createOffer and have a m= section for the added track appear in
the generated offer.
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A rollback is performed by supplying a session description of type
"rollback" with empty contents to either setLocalDescription or
setRemoteDescription, depending on which was most recently used (i.e.
if the new offer was supplied to setLocalDescription, the rollback
should be done using setLocalDescription as well).
4.1.9. setLocalDescription
The setLocalDescription method instructs the PeerConnection to apply
the supplied session description as its local configuration. The
type field indicates whether the description should be processed as
an offer, provisional answer, or final answer; offers and answers are
checked differently, using the various rules that exist for each SDP
line.
This API changes the local media state; among other things, it sets
up local resources for receiving and decoding media. In order to
successfully handle scenarios where the application wants to offer to
change from one media format to a different, incompatible format, the
PeerConnection must be able to simultaneously support use of both the
current and pending local descriptions (e.g., support the codecs that
exist in either description). This dual processing begins when the
PeerConnection enters the have-local-offer state, and continues until
setRemoteDescription is called with either a final answer, at which
point the PeerConnection can fully adopt the pending local
description, or a rollback, which results in a revert to the current
local description.
This API indirectly controls the candidate gathering process. When a
local description is supplied, and the number of transports currently
in use does not match the number of transports needed by the local
description, the PeerConnection will create transports as needed and
begin gathering candidates for each transport, using ones from the
candidate pool if available.
If setRemoteDescription was previously called with an offer, and
setLocalDescription is called with an answer (provisional or final),
and the media directions are compatible, and media is available to
send, this will result in the starting of media transmission.
4.1.10. setRemoteDescription
The setRemoteDescription method instructs the PeerConnection to apply
the supplied session description as the desired remote configuration.
As in setLocalDescription, the type field of the description
indicates how it should be processed.
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This API changes the local media state; among other things, it sets
up local resources for sending and encoding media.
If setLocalDescription was previously called with an offer, and
setRemoteDescription is called with an answer (provisional or final),
and the media directions are compatible, and media is available to
send, this will result in the starting of media transmission.
4.1.11. currentLocalDescription
The currentLocalDescription method returns the current negotiated
local description - i.e., the local description from the last
successful offer/answer exchange - in addition to any local
candidates that have been generated by the ICE agent since the local
description was set.
A null object will be returned if an offer/answer exchange has not
yet been completed.
4.1.12. pendingLocalDescription
The pendingLocalDescription method returns a copy of the local
description currently in negotiation - i.e., a local offer set
without any corresponding remote answer - in addition to any local
candidates that have been generated by the ICE agent since the local
description was set.
A null object will be returned if the state of the PeerConnection is
"stable" or "have-remote-offer".
4.1.13. currentRemoteDescription
The currentRemoteDescription method returns a copy of the current
negotiated remote description - i.e., the remote description from the
last successful offer/answer exchange - in addition to any remote
candidates that have been supplied via processIceMessage since the
remote description was set.
A null object will be returned if an offer/answer exchange has not
yet been completed.
4.1.14. pendingRemoteDescription
The pendingRemoteDescription method returns a copy of the remote
description currently in negotiation - i.e., a remote offer set
without any corresponding local answer - in addition to any remote
candidates that have been supplied via processIceMessage since the
remote description was set.
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A null object will be returned if the state of the PeerConnection is
"stable" or "have-local-offer".
4.1.15. canTrickleIceCandidates
The canTrickleIceCandidates property indicates whether the remote
side supports receiving trickled candidates. There are three
potential values:
null: No SDP has been received from the other side, so it is not
known if it can handle trickle. This is the initial value before
setRemoteDescription() is called.
true: SDP has been received from the other side indicating that it
can support trickle.
false: SDP has been received from the other side indicating that it
cannot support trickle.
As described in Section 3.5.2, JSEP implementations always provide
candidates to the application individually, consistent with what is
needed for Trickle ICE. However, applications can use the
canTrickleIceCandidates property to determine whether their peer can
actually do Trickle ICE, i.e., whether it is safe to send an initial
offer or answer followed later by candidates as they are gathered.
As "true" is the only value that definitively indicates remote
Trickle ICE support, an application which compares
canTrickleIceCandidates against "true" will by default attempt Half
Trickle on initial offers and Full Trickle on subsequent interactions
with a Trickle ICE-compatible agent.
4.1.16. setConfiguration
The setConfiguration method allows the global configuration of the
PeerConnection, which was initially set by constructor parameters, to
be changed during the session. The effects of this method call
depend on when it is invoked, and differ depending on which specific
parameters are changed:
o Any changes to the STUN/TURN servers to use affect the next
gathering phase. If an ICE gathering phase has already started or
completed, the 'needs-ice-restart' bit mentioned in Section 3.5.1
will be set. This will cause the next call to createOffer to
generate new ICE credentials, for the purpose of forcing an ICE
restart and kicking off a new gathering phase, in which the new
servers will be used. If the ICE candidate pool has a nonzero
size, and a local description has not yet been applied, any
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existing candidates will be discarded, and new candidates will be
gathered from the new servers.
o Any change to the ICE candidate policy affects the next gathering
phase. If an ICE gathering phase has already started or
completed, the 'needs-ice-restart' bit will be set. Either way,
changes to the policy have no effect on the candidate pool,
because pooled candidates are not surfaced to the application
until a gathering phase occurs, and so any necessary filtering can
still be done on any pooled candidates.
o The ICE candidate pool size MUST NOT be changed after applying a
local description. If a local description has not yet been
applied, any changes to the ICE candidate pool size take effect
immediately; if increased, additional candidates are pre-gathered;
if decreased, the now-superfluous candidates are discarded.
o The bundle and RTCP-multiplexing policies MUST NOT be changed
after the construction of the PeerConnection.
This call may result in a change to the state of the ICE Agent.
4.1.17. addIceCandidate
The addIceCandidate method provides a remote candidate to the ICE
agent, which, if parsed successfully, will be added to the current
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=
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
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
invalid, an error is generated.
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
candidate MUST be assumed to belong to the most recently applied
remote description. Connectivity checks will be sent to the new
candidate.
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]).
The MID and ufrag are used as described above to determine the m=
section and ICE generation for which candidate gathering is complete.
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
recently applied remote description. If neither the MID nor the m=
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index is present, then the indication MUST be assumed to apply to all
m= sections in the most recently applied remote description.
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
being established.
4.2. RtpTransceiver
4.2.1. stop
The stop method stops an RtpTransceiver. This will cause future
calls to createOffer to generate a zero port for the associated m=
section. See below for more details.
4.2.2. stopped
The stopped property indicates whether the transceiver has been
stopped, either by a call to stopTransceiver or by applying an answer
that rejects the associated m= section. In either of these cases, it
is set to "true", and otherwise will be set to "false".
A stopped RtpTransceiver does not send any outgoing RTP or RTCP or
process any incoming RTP or RTCP. It cannot be restarted.
4.2.3. setDirection
The setDirection method sets the direction of a transceiver, which
affects the direction property of the associated m= section on future
calls to createOffer and createAnswer.
When creating offers, the transceiver direction is directly reflected
in the output, even for reoffers. When creating answers, the
transceiver direction is intersected with the offered direction, as
explained in the Section 5.3 section below.
Note that while setDirection sets the direction property of the
transceiver immediately (Section 4.2.4), this property does not
immediately affect whether the transceiver's RtpSender will send or
its RtpReceiver will receive. The direction in effect is represented
by the currentDirection property, which is only updated when an
answer is applied.
4.2.4. direction
The direction property indicates the last value passed into
setDirection. If setDirection has never been called, it is set to
the direction the transceiver was initialized with.
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4.2.5. currentDirection
The currentDirection property indicates the last negotiated direction
for the transceiver's associated m= section. More specifically, it
indicates the [RFC3264] directional attribute of the associated m=
section in the last applied answer, with "send" and "recv" directions
reversed if it was a remote answer. For example, if the directional
attribute for the associated m= section in a remote answer is
"recvonly", currentDirection is set to "sendonly".
If an answer that references this transceiver has not yet been
applied, or if the transceiver is stopped, currentDirection is set to
null.
4.2.6. setCodecPreferences
The setCodecPreferences method sets the codec preferences of a
transceiver, which in turn affect the presence and order of codecs of
the associated m= section on future calls to createOffer and
createAnswer. Note that setCodecPreferences does not directly affect
which codec the implementation decides to send. It only affects
which codecs the implementation indicates that it prefers to receive,
via the offer or answer. Even when a codec is excluded by
setCodecPreferences, it still may be used to send until the next
offer/answer exchange discards it.
The codec preferences of an RtpTransceiver can cause codecs to be
excluded by subsequent calls to createOffer and createAnswer, in
which case the corresponding media formats in the associated m=
section will be excluded. The codec preferences cannot add media
formats that would otherwise not be present. This includes codecs
that were not negotiated in a previous offer/answer exchange that
included the transceiver.
The codec preferences of an RtpTransceiver can also determine the
order of codecs in subsequent calls to createOffer and createAnswer,
in which case the order of the media formats in the associated m=
section will follow the specified preferences.
5. SDP Interaction Procedures
This section describes the specific procedures to be followed when
creating and parsing SDP objects.
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5.1. Requirements Overview
JSEP implementations must comply with the specifications listed below
that govern the creation and processing of offers and answers.
5.1.1. 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.
o ICE, as specified in [RFC5245], MUST be used. Note that the
remote endpoint may use a Lite implementation; implementations
MUST properly handle remote endpoints which do ICE-Lite.
o DTLS [RFC6347] or DTLS-SRTP [RFC5763], MUST be used, as
appropriate for the media type, as specified in
[I-D.ietf-rtcweb-security-arch]
The SDES SRTP keying mechanism from [RFC4568] MUST NOT be used, as
discussed in [I-D.ietf-rtcweb-security-arch].
5.1.2. Profile Names and Interoperability
For media m= sections, JSEP implementations MUST support the
"UDP/TLS/RTP/SAVPF" profile specified in [RFC7850], and MUST indicate
this profile for each media m= line they produce in an offer. For
data m= sections, implementations MUST support the "UDP/DTLS/SCTP"
profile and MUST indicate this profile for each data m= line they
produce in an offer. Because ICE can select either UDP [RFC5245] or
TCP [RFC6544] transport depending on network conditions, this
advertisement is consistent with ICE eventually selecting either
either UDP or TCP.
Unfortunately, in an attempt at compatibility, some endpoints
generate other profile strings even when they mean to support one of
these profiles. For instance, an endpoint might generate "RTP/AVP"
but supply "a=fingerprint" and "a=rtcp-fb" attributes, indicating its
willingness to support "(UDP,TCP)/TLS/RTP/SAVPF". In order to
simplify compatibility with such endpoints, JSEP implementations MUST
follow the following rules when processing the media m= sections in
an offer:
o The profile in any "m=" line in any answer MUST exactly match the
profile provided in the offer.
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o Any profile matching the following patterns MUST be accepted:
"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
effect; support for DTLS-SRTP is determined by the presence of one
or more "a=fingerprint" attribute. Note that lack of an
"a=fingerprint" attribute will lead to negotiation failure.
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
is present, assume AVPF timing, i.e., a default value of "trr-
int=0". Otherwise, assume that AVPF is being used in an AVP
compatible mode and use a value of "trr-int=4000".
o For data m= sections, implementations MUST support receiving the
"UDP/DTLS/SCTP", "TCP/DTLS/SCTP", or "DTLS/SCTP" (for backwards
compatibility) profiles.
Note that re-offers by JSEP implementations MUST use the correct
profile strings even if the initial offer/answer exchange used an
(incorrect) older profile string.
5.2. Constructing an Offer
When createOffer is called, a new SDP description must be created
that includes the functionality specified in
[I-D.ietf-rtcweb-rtp-usage]. The exact details of this process are
explained below.
5.2.1. Initial Offers
When createOffer is called for the first time, the result is known as
the initial offer.
The first step in generating an initial offer is to generate session-
level attributes, as specified in [RFC4566], Section 5.
Specifically:
o The first SDP line MUST be "v=0", as specified in [RFC4566],
Section 5.1
o The second SDP line MUST be an "o=" line, as specified in
[RFC4566], Section 5.2. The value of the <username> field SHOULD
be "-". [RFC3264] requires that the <sess-id> be representable as
a 64-bit signed integer. It is RECOMMENDED that the <sess-id> be
generated as a 64-bit quantity with the high bit being sent to
zero and the remaining 63 bits being cryptographically random.
The value of the <nettype> <addrtype> <unicast-address> tuple
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SHOULD be set to a non-meaningful address, such as IN IP4 0.0.0.0,
to prevent leaking the local address in this field. As mentioned
in [RFC4566], the entire o= line needs to be unique, but selecting
a random number for <sess-id> is sufficient to accomplish this.
o The third SDP line MUST be a "s=" line, as specified in [RFC4566],
Section 5.3; to match the "o=" line, a single dash SHOULD be used
as the session name, e.g. "s=-". Note that this differs from the
advice in [RFC4566] which proposes a single space, but as both
"o=" and "s=" are meaningless, having the same meaningless value
seems clearer.
o Session Information ("i="), URI ("u="), Email Address ("e="),
Phone Number ("p="), Repeat Times ("r="), and Time Zones ("z=")
lines are not useful in this context and SHOULD NOT be included.
o Encryption Keys ("k=") lines do not provide sufficient security
and MUST NOT be included.
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
0".
o An "a=ice-options" line with the "trickle" option MUST be added,
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]
Section 5.14. An m= section is generated for each RtpTransceiver
that has been added to the PeerConnection, excluding any stopped
RtpTransceivers. This is done in the order the RtpTransceivers were
added to the PeerConnection.
For each m= section generated for an RtpTransceiver, establish a
mapping between the transceiver and the index of the generated m=
section.
Each m= section, provided it is not marked as bundle-only, MUST
generate a unique set of ICE credentials and gather its own unique
set of ICE candidates. Bundle-only m= sections MUST NOT contain any
ICE credentials and MUST NOT gather any candidates.
For DTLS, all m= sections MUST use all the certificate(s) that have
been specified for the PeerConnection; as a result, they MUST all
have the same [I-D.ietf-mmusic-4572-update] fingerprint value(s), or
these value(s) MUST be session-level attributes.
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Each m= section should be generated as specified in [RFC4566],
Section 5.14. For the m= line itself, the following rules MUST be
followed:
o The port value is set to the port of the default ICE candidate for
this m= section, but given that no candidates are available yet,
the "dummy" port value of 9 (Discard) MUST be used, as indicated
in [I-D.ietf-ice-trickle], Section 5.1.
o To properly indicate use of DTLS, the <proto> field MUST be set to
"UDP/TLS/RTP/SAVPF", as specified in [RFC5764], Section 8.
o If codec preferences have been set for the associated transceiver,
media formats MUST be generated in the corresponding order, and
MUST exclude any codecs not present in the codec preferences.
o The media formats in the answer MAY include codecs present in the
offer that were discarded in a previous offer/answer exchange.
This is necessary for compatibility with third- party call control
and SIP use cases.
o Unless excluded by the above restrictions, the media formats MUST
include the mandatory audio/video codecs as specified in
[I-D.ietf-rtcweb-audio] (see Section 3) and
[I-D.ietf-rtcweb-video] (see Section 5).
The m= line MUST be followed immediately by a "c=" line, as specified
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",
as defined in [I-D.ietf-ice-trickle], Section 5.1.
[I-D.ietf-mmusic-sdp-mux-attributes] groups SDP attributes into
different categories. To avoid unnecessary duplication when
bundling, Section 8.1 of [I-D.ietf-mmusic-sdp-bundle-negotiation]
specifies that attributes of category IDENTICAL or TRANSPORT should
not be repeated in bundled m= sections.
The following attributes, which are of a category other than
IDENTICAL or TRANSPORT, MUST be included in each m= section:
o An "a=mid" line, as specified in [RFC5888], Section 4. All MID
values MUST be generated in a fashion that does not leak user
information, e.g., randomly or using a per-PeerConnection counter,
and SHOULD be 3 bytes or less, to allow them to efficiently fit
into the RTP header extension defined in
[I-D.ietf-mmusic-sdp-bundle-negotiation], Section 14. Note that
this does not set the RtpTransceiver mid property, as that only
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occurs when the description is applied. The generated MID value
can be considered a "proposed" MID at this point.
o A direction attribute which is the same as that of the associated
transceiver.
o For each media format on the m= line, "a=rtpmap" and "a=fmtp"
lines, as specified in [RFC4566], Section 6, and [RFC3264],
Section 5.1.
o For each primary codec where RTP retransmission should be used, a
corresponding "a=rtpmap" line indicating "rtx" with the clock rate
of the primary codec and an "a=fmtp" line that references the
payload type of the primary codec, as specified in [RFC4588],
Section 8.1.
o For each supported FEC mechanism, "a=rtpmap" and "a=fmtp" lines,
as specified in [RFC4566], Section 6. The FEC mechanisms that
MUST be supported are specified in [I-D.ietf-rtcweb-fec],
Section 6, and specific usage for each media type is outlined in
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
specified in [RFC5285], Section 5. The list of header extensions
that SHOULD/MUST be supported is specified in
[I-D.ietf-rtcweb-rtp-usage], Section 5.2. Any header extensions
that require encryption MUST be specified as indicated in
[RFC6904], Section 4.
o For each supported RTCP feedback mechanism, an "a=rtcp-fb"
mechanism, as specified in [RFC4585], Section 4.2. The list of
RTCP feedback mechanisms that SHOULD/MUST be supported is
specified in [I-D.ietf-rtcweb-rtp-usage], Section 5.1.
o If the RtpTransceiver has a sendrecv or sendonly direction:
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* For each MediaStream that was associated with the transceiver
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
the application has specified RID values or has specified more
than one encoding in the RtpSenders's parameters, an "a=rid" line
for each encoding specified. The "a=rid" line is specified in
[I-D.ietf-mmusic-rid], and its direction MUST be "send". If the
application has chosen a RID value, it MUST be used as the rid-
identifier; otherwise a RID value MUST be generated by the
implementation. RID values MUST be generated in a fashion that
does not leak user information, e.g., randomly or using a per-
PeerConnection counter, and SHOULD be 3 bytes or less, to allow
them to efficiently fit into the RTP header extension defined in
[I-D.ietf-avtext-rid], Section 3. If no encodings have been
specified, or only one encoding is specified but without a RID
value, then no "a=rid" lines are generated.
o If the RtpTransceiver has a sendrecv or sendonly direction and
more than one "a=rid" line has been generated, an "a=simulcast"
line, with direction "send", as defined in
[I-D.ietf-mmusic-sdp-simulcast], Section 6.2. The list of RIDs
MUST include all of the RID identifiers used in the "a=rid" lines
for this m= section.
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
TRANSPORT, MUST appear only in "m=" sections which either have a
unique address or which are associated with the bundle-tag. (In
initial offers, this means those "m=" sections which do not contain
an "a=bundle-only" attribute.)
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o "a=ice-ufrag" and "a=ice-pwd" lines, as specified in [RFC5245],
Section 15.4.
o An "a=fingerprint" line for each of the endpoint's certificates,
as specified in [RFC4572], Section 5; the digest algorithm used
for the fingerprint MUST match that used in the certificate
signature.
o An "a=setup" line, as specified in [RFC4145], Section 4, and
clarified for use in DTLS-SRTP scenarios in [RFC5763], Section 5.
The role value in the offer MUST be "actpass".
o An "a=dtls-id" line, as specified in [I-D.ietf-mmusic-dtls-sdp]
Section 5.2.
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
candidates have yet been gathered.
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.
Lastly, if a data channel has been created, a m= section MUST be
generated for data. The <media> field MUST be set to "application"
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-
datachannel" as specified in [I-D.ietf-mmusic-sctp-sdp], Section 4.1.
Within the data m= section, an "a=mid" line MUST be generated and
included as described above, along with 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
"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"
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o "a=fingerprint"
o "a=setup"
o "a=dtls-id"
Once all m= sections have been generated, a session-level "a=group"
attribute MUST be added as specified in [RFC5888]. This attribute
MUST have semantics "BUNDLE", and MUST include the mid identifiers of
each m= section. The effect of this is that the JSEP implementation
offers all m= sections as one bundle group. However, whether the m=
sections are bundle-only or not depends on the bundle policy.
The next step is to generate session-level lip sync groups as defined
in [RFC5888], Section 7. For each MediaStream referenced by more
than one RtpTransceiver (by passing those MediaStreams as arguments
to the addTrack and addTransceiver methods), a group of type "LS"
MUST be added that contains the mid values for each RtpTransceiver.
Attributes which SDP permits to either be at the session level or the
media level SHOULD generally be at the media level even if they are
identical. This promotes readability, especially if one of a set of
initially identical attributes is subsequently changed.
Attributes other than the ones specified above MAY be included,
except for the following attributes which are specifically
incompatible with the requirements of [I-D.ietf-rtcweb-rtp-usage],
and MUST NOT be included:
o "a=crypto"
o "a=key-mgmt"
o "a=ice-lite"
Note that when bundle is used, any additional attributes that are
added MUST follow the advice in [I-D.ietf-mmusic-sdp-mux-attributes]
on how those attributes interact with bundle.
Note that these requirements are in some cases stricter than those of
SDP. Implementations MUST be prepared to accept compliant SDP even
if it would not conform to the requirements for generating SDP in
this specification.
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5.2.2. Subsequent Offers
When createOffer is called a second (or later) time, or is called
after a local description has already been installed, the processing
is somewhat different than for an initial offer.
If the initial offer was not applied using setLocalDescription,
meaning the PeerConnection is still in the "stable" state, the steps
for generating an initial offer should be followed, subject to the
following restriction:
o The fields of the "o=" line MUST stay the same except for the
<session-version> field, which MUST increment by one on each call
to createOffer if the offer might differ from the output of the
previous call to createOffer; implementations MAY opt to increment
<session-version> on every call. The value of the generated
<session-version> is independent of the <session-version> of the
current local description; in particular, in the case where the
current version is N, an offer is created and applied with version
N+1, and then that offer is rolled back so that the current
version is again N, the next generated offer will still have
version N+2.
Note that if the application creates an offer by reading
currentLocalDescription instead of calling createOffer, the returned
SDP may be different than when setLocalDescription was originally
called, due to the addition of gathered ICE candidates, but the
<session-version> will not have changed. There are no known
scenarios in which this causes problems, but if this is a concern,
the solution is simply to use createOffer to ensure a unique
<session-version>.
If the initial offer was applied using setLocalDescription, but an
answer from the remote side has not yet been applied, meaning the
PeerConnection is still in the "local-offer" state, an offer is
generated by following the steps in the "stable" state above, along
with these exceptions:
o The "s=" and "t=" lines MUST stay the same.
o If any RtpTransceiver has been added, and there exists an m=
section with a zero port in the current local description or the
current remote description, that m= section MUST be recycled by
generating an m= section for the added RtpTransceiver as if the m=
section were being added to the session description, placed at the
same index as the m= section with a zero port.
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o If an RtpTransceiver is stopped and is not associated with an m=
section, an m= section MUST NOT be generated for it. This
prevents adding back RtpTransceivers whose m= sections were
recycled and used for a new RtpTransceiver in a previous offer/
answer exchange, as described above.
o If an RtpTransceiver has been stopped and is associated with an m=
section, and the m= section is not being recycled as described
above, an m= section MUST be generated for it with the port set to
zero and all "a=msid" lines removed.
o For RtpTransceivers that are not stopped, the "a=msid" line(s)
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,
and address of the default candidate for the m= section, as
described in [RFC5245], Section 4.3. If ICE checking has already
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
yet completed. Note that this differs from the guidance in
[RFC5245], Section 9.1.2.2, which only refers to offers created
when ICE has completed. In each case, if no RTP candidates have
yet been gathered, dummy values MUST be used, as described above.
o Each "a=mid" line MUST stay the same.
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
STUN/TURN servers, or the ICE candidate policy), or the
"IceRestart" option ( Section 5.2.3.1 was specified. If the m=
section is bundled into another m= section, it still MUST NOT
contain any ICE credentials.
o If the m= section is not bundled into another m= section, its
"a=rtcp" attribute line MUST be filled in with the port and
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
candidate that has been gathered during the most recent gathering
phase (see Section 3.5.1), an "a=candidate" line MUST be added, as
defined in [RFC5245], Section 4.3., paragraph 3. If candidate
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gathering for the section has completed, an "a=end-of-candidates"
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, both "a=candidate" and "a=end-of-candidates" MUST be
omitted.
o For RtpTransceivers that are still present, the "a=rid" lines MUST
stay the same.
o For RtpTransceivers that are still present, any "a=simulcast" line
MUST stay the same.
o If any RtpTransceiver has been stopped, the port MUST be set to
zero and all "a=msid" lines MUST be removed.
o If any RtpTransceiver has been added, and there exists a m=
section with a zero port in the current local description or the
current remote description, that m= section MUST be recycled by
generating a m= section for the added RtpTransceiver as if the m=
section were being added to session description, except that
instead of adding it, the generated m= section replaces the m=
section with a zero port. The new m= section MUST contain a new
MID.
If the initial offer was applied using setLocalDescription, and an
answer from the remote side has been applied using
setRemoteDescription, meaning the PeerConnection is in the "remote-
pranswer" or "stable" states, an offer is generated based on the
negotiated session descriptions by following the steps mentioned for
the "local-offer" state above.
In addition, for each non-recycled, non-rejected m= section in the
new offer, the following adjustments are made based on the contents
of the corresponding m= section in the current remote description, if
any:
o The m= line and corresponding "a=rtpmap" and "a=fmtp" lines MUST
only include codecs present in the most recent answer which have
not been excluded by the codec preferences of the associated
transceiver. Note that non-JSEP endpoints are not subject to
these restrictions, and might offer media formats that were not
present in the most recent answer, as specified in [RFC3264],
Section 8. Therefore, JSEP implementations MUST be prepared to
receive such offers.
o Unless codec preferences have been set for the associated
transceiver, the media formats on the m= line MUST be generated in
the same order as in the current local description.
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o The RTP header extensions MUST only include those that are present
in the most recent answer.
o The RTCP feedback extensions MUST only include those that are
present in the most recent answer.
o The "a=rtcp" line MUST only be added if the most recent answer did
not include an "a=rtcp-mux" line.
o The "a=rtcp-mux" line MUST only be added if present in the most
recent answer.
o The "a=rtcp-mux-only" line MUST NOT be added.
o The "a=rtcp-rsize" line MUST only be added if present in the most
recent answer.
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=
sections that have been marked as rejected, plus any newly added or
re-enabled m= sections. In other words, the bundle attribute must
contain all m= sections that were previously bundled, as long as they
are still alive, as well as any new m= sections.
"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
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any synchronized "recvonly" m= sections continue to be synchronized
in the new offer.
5.2.3. Options Handling
The createOffer method takes as a parameter an RTCOfferOptions
object. Special processing is performed when generating a SDP
description if the following options are present.
5.2.3.1. IceRestart
If the "IceRestart" option is specified, with a value of "true", the
offer MUST indicate an ICE restart by generating new ICE ufrag and
pwd attributes, as specified in [RFC5245], Section 9.1.1.1. If this
option is specified on an initial offer, it has no effect (since a
new ICE ufrag and pwd are already generated). Similarly, if the ICE
configuration has changed, this option has no effect, since new ufrag
and pwd attributes will be generated automatically. This option is
primarily useful for reestablishing connectivity in cases where
failures are detected by the application.
5.2.3.2. VoiceActivityDetection
If the "VoiceActivityDetection" option is specified, with a value of
"true", the offer MUST indicate support for silence suppression in
the audio it receives by including comfort noise ("CN") codecs for
each offered audio codec, as specified in [RFC3389], Section 5.1,
except for codecs that have their own internal silence suppression
support. For codecs that have their own internal silence suppression
support, the appropriate fmtp parameters for that codec MUST be
specified to indicate that silence suppression for received audio is
desired. For example, when using the Opus codec [RFC6716], the
"usedtx=1" parameter, specified in [RFC7587], would be used in the
offer. This option allows the endpoint to significantly reduce the
amount of audio bandwidth it receives, at the cost of some fidelity,
depending on the quality of the remote VAD algorithm.
If the "VoiceActivityDetection" option is specified, with a value of
"false", the JSEP implementation MUST NOT emit "CN" codecs. For
codecs that have their own internal silence suppression support, the
appropriate fmtp parameters for that codec MUST be specified to
indicate that silence suppression for received audio is not desired.
For example, when using the Opus codec, the "usedtx=0" parameter
would be specified in the offer.
Note that setting the "VoiceActivityDetection" parameter when
generating an offer is a request to receive audio with silence
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suppression. It has no impact on whether the local endpoint does
silence suppression for the audio it sends.
The "VoiceActivityDetection" option does not have any impact on the
setting of the "vad" value in the signaling of the client to mixer
audio level header extension described in [RFC6464], Section 4.
5.3. Generating an Answer
When createAnswer is called, a new SDP description must be created
that is compatible with the supplied remote description as well as
the requirements specified in [I-D.ietf-rtcweb-rtp-usage]. The exact
details of this process are explained below.
5.3.1. Initial Answers
When createAnswer is called for the first time after a remote
description has been provided, the result is known as the initial
answer. If no remote description has been installed, an answer
cannot be generated, and an error MUST be returned.
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
Section 5.2. For many cases, this is not a problem. However, if any
mandatory SDP attributes are missing, or functionality listed as
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
rejected.
The first step in generating an initial answer is to generate
session-level attributes. The process here is identical to that
indicated in the initial offers section above, except that the
"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
was present in the offer.
The next step is to generate session-level lip sync groups, as
defined in [RFC5888], Section 7. For each group of type "LS" present
in the offer, select the local RtpTransceivers that are referenced by
the MID values in the specified group, and determine which of them
either reference a common local MediaStream (specified in the calls
to addTrack/addTransceiver used to create them), or have no
MediaStream to reference because they were not created by addTrack/
addTransceiver. If at least two such RtpTransceivers exist, a group
of type "LS" with the mid values of these RtpTransceivers MUST be
added. Otherwise the offered "LS" group MUST be ignored and no
corresponding group generated in the answer.
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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.
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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
present in the remote offer, as specified in [RFC3264], Section 6.
For the purposes of this discussion, any session-level attributes in
the offer that are also valid as media-level attributes are
considered to be present in each m= section.
The next step is to go through each offered m= section. Each offered
m= section will have an associated RtpTransceiver, as described in
Section 5.9. If there are more RtpTransceivers than there are m=
sections, the unmatched RtpTransceivers will need to be associated in
a subsequent offer.
For each offered m= section, if any of the following conditions are
true, the corresponding m= section in the answer MUST be marked as
rejected by setting the port in the m= line to zero, as indicated in
[RFC3264], Section 6., and further processing for this m= section can
be skipped:
o The associated RtpTransceiver has been stopped.
o No supported codec is present in the offer.
o The bundle policy is "max-bundle", and this is not the first m=
section or in the same bundle group as the first m= section.
o The bundle policy is "balanced", and this is not the first m=
section for this media type or in the same bundle group as the
first m= section for this media type.
Otherwise, each m= section in the answer should then be generated as
specified in [RFC3264], Section 6.1. For the m= line itself, the
following rules must be followed:
o The port value would normally be set to the port of the default
ICE candidate for this m= section, but given that no candidates
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are available yet, the "dummy" port value of 9 (Discard) MUST be
used, as indicated in [I-D.ietf-ice-trickle], Section 5.1.
o The <proto> field MUST be set to exactly match the <proto> field
for the corresponding m= line in the offer.
o If codec preferences have been set for the associated transceiver,
media formats MUST be generated in the corresponding order, and
MUST exclude any codecs not present in the codec preferences or
not present in the offer. Note that non-JSEP endpoints are not
subject to this restriction, and might add media formats in the
answer that are not present in the offer, as specified in
[RFC3264], Section 6.1. Therefore, JSEP implementations MUST be
prepared to receive such answers.
o Unless excluded by the above restrictions, the media formats MUST
include the mandatory audio/video codecs as specified in
[I-D.ietf-rtcweb-audio] (see Section 3) and
[I-D.ietf-rtcweb-video] (see Section 5).
The m= line MUST be followed immediately by a "c=" line, as specified
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",
as defined in [I-D.ietf-ice-trickle], Section 5.1.
If the offer supports bundle, all m= sections to be bundled must use
the same ICE credentials and candidates; all m= sections not being
bundled must use unique ICE credentials and candidates. Each m=
section MUST contain the following attributes (which are of attribute
types other than IDENTICAL and TRANSPORT):
o If and only if present in the offer, an "a=mid" line, as specified
in [RFC5888], Section 9.1. The "mid" value MUST match that
specified in the offer.
o A direction attribute, determined by applying the rules regarding
the offered direction specified in [RFC3264], Section 6.1, and
then intersecting with the direction of the associated
RtpTransceiver. For example, in the case where an m= section is
offered as "sendonly", and the local transceiver is set to
"sendrecv", the result in the answer is a "recvonly" direction.
o For each media format on the m= line, "a=rtpmap" and "a=fmtp"
lines, as specified in [RFC4566], Section 6, and [RFC3264],
Section 6.1.
o If "rtx" is present in the offer, for each primary codec where RTP
retransmission should be used, a corresponding "a=rtpmap" line
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indicating "rtx" with the clock rate of the primary codec and an
"a=fmtp" line that references the payload type of the primary
codec, as specified in [RFC4588], Section 8.1.
o For each supported FEC mechanism, "a=rtpmap" and "a=fmtp" lines,
as specified in [RFC4566], Section 6. The FEC mechanisms that
MUST be supported are specified in [I-D.ietf-rtcweb-fec],
Section 6, and specific usage for each media type is outlined in
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
offer, an "a=extmap" line, as specified in [RFC5285], Section 5.
The list of header extensions that SHOULD/MUST be supported is
specified in [I-D.ietf-rtcweb-rtp-usage], Section 5.2. Any header
extensions that require encryption MUST be specified as indicated
in [RFC6904], Section 4.
o For each supported RTCP feedback mechanism that is present in the
offer, an "a=rtcp-fb" mechanism, as specified in [RFC4585],
Section 4.2. The list of RTCP feedback mechanisms that SHOULD/
MUST be supported is specified in [I-D.ietf-rtcweb-rtp-usage],
Section 5.1.
o If the RtpTransceiver has a sendrecv or sendonly direction:
* For each MediaStream that was associated with the transceiver
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.
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Each m= section which is not bundled into another m= section, MUST
contain the following attributes (which are of category IDENTICAL or
TRANSPORT):
o "a=ice-ufrag" and "a=ice-pwd" lines, as specified in [RFC5245],
Section 15.4.
o An "a=fingerprint" line for each of the endpoint's certificates,
as specified in [RFC4572], Section 5; the digest algorithm used
for the fingerprint MUST match that used in the certificate
signature.
o An "a=setup" line, as specified in [RFC4145], Section 4, and
clarified for use in DTLS-SRTP scenarios in [RFC5763], Section 5.
The role value in the answer MUST be "active" or "passive"; the
"active" role is RECOMMENDED.
o An "a=dtls-id" line, as specified in [I-D.ietf-mmusic-dtls-sdp]
Section 5.3.
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
specified in [RFC3605], Section 2.1, containing the dummy value "9
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
[RFC5506], Section 5.
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
"application" and the <proto> and <fmt> fields MUST be set to exactly
match the fields in the offer.
Within the data m= section, an "a=mid" line MUST be generated and
included as described above, along with 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
"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 is not bundled
into another m= section:
o "a=ice-ufrag"
o "a=ice-pwd"
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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,
corresponding session-level "a=group" attributes MUST be added as
specified in [RFC5888]. These attributes MUST have semantics
"BUNDLE", and MUST include the all mid identifiers from the offered
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
answer should have an "a=bundle-only" line.
Attributes that are common between all m= sections MAY be moved to
session-level, if explicitly defined to be valid at session-level.
The attributes prohibited in the creation of offers are also
prohibited in the creation of answers.
5.3.2. Subsequent Answers
When createAnswer is called a second (or later) time, or is called
after a local description has already been installed, the processing
is somewhat different than for an initial answer.
If the initial answer was not applied using setLocalDescription,
meaning the PeerConnection is still in the "have-remote-offer" state,
the steps for generating an initial answer should be followed,
subject to the following restriction:
o The fields of the "o=" line MUST stay the same except for the
<session-version> field, which MUST increment if the session
description changes in any way from the previously generated
answer.
If any session description was previously supplied to
setLocalDescription, an answer is generated by following the steps in
the "have-remote-offer" state above, along with these exceptions:
o The "s=" and "t=" lines MUST stay the same.
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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
[RFC5245], Section 4.3. Note, however, that the m= line protocol
need not match the default candidate, because this protocol value
must instead match what was supplied in the offer, as described
above.
o Unless codec preferences have been set for the associated
transceiver, the media formats on the m= line MUST be generated in
the same order as in the current local description.
o Each "a=ice-ufrag" and "a=ice-pwd" line MUST stay the same, unless
the m= section is restarting, in which case new ICE credentials
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
NOT contain any ICE credentials.
o Each "a=setup" line MUST use an "active" or "passive" role value
consistent with the existing DTLS association, if the association
is being continued by the offerer.
o If the m= section is not bundled into another m= section and RTCP
multiplexing is not active, an "a=rtcp" attribute line MUST be
filled in with the port and address of the default RTCP candidate.
If no RTCP candidates have yet been gathered, dummy values MUST be
used, as described in the initial answer section above.
o If the m= section is not bundled into another m= section, for each
candidate that has been gathered during the most recent gathering
phase (see Section 3.5.1), an "a=candidate" line MUST be added, as
defined in [RFC5245], Section 4.3., paragraph 3. If candidate
gathering for the section has completed, an "a=end-of-candidates"
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, both "a=candidate" and "a=end-of-candidates" MUST be
omitted.
o For RtpTransceivers that are not stopped, the "a=msid" line(s)
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
The createAnswer method takes as a parameter an RTCAnswerOptions
object. The set of parameters for RTCAnswerOptions is different than
those supported in RTCOfferOptions; the IceRestart option is
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unnecessary, as ICE credentials will automatically be changed for all
m= sections where the offerer chose to perform ICE restart.
The following options are supported in RTCAnswerOptions.
5.3.3.1. VoiceActivityDetection
Silence suppression in the answer is handled as described in
Section 5.2.3.2, with one exception: if support for silence
suppression was not indicated in the offer, the
VoiceActivityDetection parameter has no effect, and the answer should
be generated as if VoiceActivityDetection was set to false. This is
done on a per-codec basis (e.g., if the offerer somehow offered
support for CN but set "usedtx=0" for Opus, setting
VoiceActivityDetection to true would result in an answer with CN
codecs and "usedtx=0").
5.4. Modifying an Offer or Answer
The SDP returned from createOffer or createAnswer MUST NOT be changed
before passing it to setLocalDescription. If precise control over
the SDP is needed, the aforementioned createOffer/createAnswer
options or RtpTransceiver APIs MUST be used.
Note that the application MAY modify the SDP to reduce the
capabilities in the offer it sends to the far side (post-
setLocalDescription) or the offer that it installs from the far side
(pre-setRemoteDescription), as long as it remains a valid SDP offer
and specifies a subset of what was in the original offer. This is
safe because the answer is not permitted to expand capabilities, and
therefore will just respond to what is present in the offer.
The application SHOULD NOT modify the SDP in the answer it transmits,
as the answer contains the negotiated capabilities, and this can
cause the two sides to have different ideas about what exactly was
negotiated.
As always, the application is solely responsible for what it sends to
the other party, and all incoming SDP will be processed by the JSEP
implementation to the extent of its capabilities. It is an error to
assume that all SDP is well-formed; however, one should be able to
assume that any implementation of this specification will be able to
process, as a remote offer or answer, unmodified SDP coming from any
other implementation of this specification.
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5.5. Processing a Local Description
When a SessionDescription is supplied to setLocalDescription, the
following steps MUST be performed:
o First, the type of the SessionDescription is checked against the
current state of the PeerConnection:
* If the type is "offer", the PeerConnection state MUST be either
"stable" or "have-local-offer".
* If the type is "pranswer" or "answer", the PeerConnection state
MUST be either "have-remote-offer" or "have-local-pranswer".
o If the type is not correct for the current state, processing MUST
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
described in the Section 5.7 section below. If parsing fails for
any reason, processing MUST stop and an error MUST be returned.
o Finally, the parsed SessionDescription is applied as described in
the Section 5.8 section below.
5.6. Processing a Remote Description
When a SessionDescription is supplied to setRemoteDescription, the
following steps MUST be performed:
o First, the type of the SessionDescription is checked against the
current state of the PeerConnection:
* If the type is "offer", the PeerConnection state MUST be either
"stable" or "have-remote-offer".
* If the type is "pranswer" or "answer", the PeerConnection state
MUST be either "have-local-offer" or "have-remote-pranswer".
o If the type is not correct for the current state, processing MUST
stop and an error MUST be returned.
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o Next, the SessionDescription is parsed into a data structure, as
described in the Section 5.7 section below. If parsing fails for
any reason, processing MUST stop and an error MUST be returned.
o Finally, the parsed SessionDescription is applied as described in
the Section 5.9 section below.
5.7. Parsing a Session Description
When a SessionDescription of any type is supplied to setLocal/
RemoteDescription, the implementation must parse it and reject it if
it is invalid. The exact details of this process are explained
below.
The SDP contained in the session description object consists of a
sequence of text lines, each containing a key-value expression, as
described in [RFC4566], Section 5. The SDP is read, line-by-line,
and converted to a data structure that contains the deserialized
information. However, SDP allows many types of lines, not all of
which are relevant to JSEP applications. For each line, the
implementation will first ensure it is syntactically correct
according to its defining ABNF, check that it conforms to [RFC4566]
and [RFC3264] semantics, and then either parse and store or discard
the provided value, as described below.
If any line is not well-formed, or cannot be parsed as described, the
parser MUST stop with an error and reject the session description,
even if the value is to be discarded. This ensures that
implementations do not accidentally misinterpret ambiguous SDP.
5.7.1. Session-Level Parsing
First, the session-level lines are checked and parsed. These lines
MUST occur in a specific order, and with a specific syntax, as
defined in [RFC4566], Section 5. Note that while the specific line
types (e.g. "v=", "c=") MUST occur in the defined order, lines of the
same type (typically "a=") can occur in any order, and their ordering
is not meaningful.
The following non-attribute lines are not meaningful in the JSEP
context and MAY be discarded once they have been checked.
The "c=" line MUST be checked for syntax but its value is not
used. This supersedes the guidance in [RFC5245], Section 6.1, to
use "ice-mismatch" to indicate mismatches between "c=" and the
candidate lines; because JSEP always uses ICE, "ice-mismatch" is
not useful in this context.
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The "i=", "u=", "e=", "p=", "t=", "r=", "z=", and "k=" lines are
not used by this specification; they MUST be checked for syntax
but their values are not used.
The remaining non-attribute lines are processed as follows:
The "v=" line MUST have a version of 0, as specified in [RFC4566],
Section 5.1.
The "o=" line MUST be parsed as specified in [RFC4566],
Section 5.2.
The "b=" line, if present, MUST be parsed as specified in
[RFC4566], Section 5.8, and the bwtype and bandwidth values
stored.
Finally, the attribute lines are processed. Specific processing MUST
be applied for the following session-level attribute ("a=") lines:
o Any "a=group" lines are parsed as specified in [RFC5888],
Section 5, and the group's semantics and mids are stored.
o If present, a single "a=ice-lite" line is parsed as specified in
[RFC5245], Section 15.3, and a value indicating the presence of
ice-lite is stored.
o If present, a single "a=ice-ufrag" line is parsed as specified in
[RFC5245], Section 15.4, and the ufrag value is stored.
o If present, a single "a=ice-pwd" line is parsed as specified in
[RFC5245], Section 15.4, and the password value is stored.
o If present, a single "a=ice-options" line is parsed as specified
in [RFC5245], Section 15.5, and the set of specified options is
stored.
o Any "a=fingerprint" lines are parsed as specified in [RFC4572],
Section 5, and the set of fingerprint and algorithm values is
stored.
o If present, a single "a=setup" line is parsed as specified in
[RFC4145], Section 4, and the setup value is stored.
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
stored.
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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],
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
continues with the lines in m= sections.
5.7.2. Media Section Parsing
Like the session-level lines, the media section lines MUST occur in
the specific order and with the specific syntax defined in [RFC4566],
Section 5.
The "m=" line itself MUST be parsed as described in [RFC4566],
Section 5.14, and the media, port, proto, and fmt values stored.
Following the "m=" line, specific processing MUST be applied for the
following non-attribute lines:
o As with the "c=" line at the session level, the "c=" line MUST be
parsed according to [RFC4566], Section 5.7, but its value is not
used.
o The "b=" line, if present, MUST be parsed as specified in
[RFC4566], Section 5.8, and the bwtype and bandwidth values
stored.
Specific processing MUST also be applied for the following attribute
lines:
o If present, a single "a=ice-ufrag" line is parsed as specified in
[RFC5245], Section 15.4, and the ufrag value is stored.
o If present, a single "a=ice-pwd" line is parsed as specified in
[RFC5245], Section 15.4, and the password value is stored.
o If present, a single "a=ice-options" line is parsed as specified
in [RFC5245], Section 15.5, and the set of specified options is
stored.
o Any "a=candidate" attributes MUST be parsed as specified in
[RFC5245], Section 15.1, and their values stored.
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o Any "a=remote-candidates" attributes MUST be parsed as specified
in [RFC5245], Section 15.2, but their values are ignored.
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
its presence or absence flagged and stored.
o Any "a=fingerprint" lines are parsed as specified in [RFC4572],
Section 5, and the set of fingerprint and algorithm values is
stored.
If the "m=" proto value indicates use of RTP, as described in the
Section 5.1.2 section above, the following attribute lines MUST be
processed:
o The "m=" fmt value MUST be parsed as specified in [RFC4566],
Section 5.14, and the individual values stored.
o Any "a=rtpmap" or "a=fmtp" lines MUST be parsed as specified in
[RFC4566], Section 6, and their values stored.
o If present, a single "a=ptime" line MUST be parsed as described in
[RFC4566], Section 6, and its value stored.
o If present, a single "a=maxptime" line MUST be parsed as described
in [RFC4566], Section 6, and its value stored.
o If present, a single direction attribute line (e.g. "a=sendrecv")
MUST be parsed as described in [RFC4566], Section 6, and its value
stored.
o Any "a=ssrc" or "a=ssrc-group" attributes MUST be parsed as
specified in [RFC5576], Sections 4.1-4.2, and their values stored.
o Any "a=extmap" attributes MUST be parsed as specified in
[RFC5285], Section 5, and their values stored.
o Any "a=rtcp-fb" attributes MUST be parsed as specified in
[RFC4585], Section 4.2., and their values stored.
o If present, a single "a=rtcp-mux" attribute MUST be parsed as
specified in [RFC5761], Section 5.1.3, and its presence or absence
flagged and stored.
o If present, a single "a=rtcp-mux-only" attribute MUST be parsed as
specified in [I-D.ietf-mmusic-mux-exclusive], Section 3, and its
presence or absence flagged and stored.
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o If present, a single "a=rtcp-rsize" attribute MUST be parsed as
specified in [RFC5506], Section 5, and its presence or absence
flagged and stored.
o If present, a single "a=rtcp" attribute MUST be parsed as
specified in [RFC3605], Section 2.1, but its value is ignored, as
this information is superfluous when using ICE.
o If present, "a=msid" attributes MUST be parsed as specified in
[I-D.ietf-mmusic-msid], Section 3.2, and their values stored.
o Any "a=imageattr" attributes MUST be parsed as specified in
[RFC6236], Section 3, and their values stored.
o Any "a=rid" lines MUST be parsed as specified in
[I-D.ietf-mmusic-rid], Section 10, and their values stored.
o If present, a single "a=simulcast" line MUST be parsed as
specified in [I-D.ietf-mmusic-sdp-simulcast], and its values
stored.
Otherwise, if the "m=" proto value indicates use of SCTP, the
following attribute lines MUST be processed:
o The "m=" fmt value MUST be parsed as specified in
[I-D.ietf-mmusic-sctp-sdp], Section 4.3, and the application
protocol value stored.
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
value stored.
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
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
Assuming parsing completes successfully, the parsed description is
then evaluated to ensure internal consistency as well as proper
support for mandatory features. Specifically, the following checks
are performed:
o For each m= section, valid values for each of the mandatory-to-use
features enumerated in Section 5.1.1 MUST be present. These
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values MAY either be present at the media level, or inherited from
the session level.
* ICE ufrag and password values, which MUST comply with the size
limits specified in [RFC5245], Section 15.4.
* dtls-id value, which MUST be set according to
[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,
the DTLS connection is not being continued and the remote
description MUST be part of an ICE restart, together with new
ufrag and password values. If this is an answer, the dtls-id
value, if present, MUST be the same as in the offer.
* DTLS setup value, which MUST be set according to the rules
specified in [RFC5763], Section 5 and MUST be consistent with
the selected role of the current DTLS connection, if one exists
and is being continued.
* DTLS fingerprint values, where at least one fingerprint MUST be
present.
o All RID values referenced in an "a=simulcast" line MUST exist as
"a=rid" lines.
o Each m= section is also checked to ensure prohibited features are
not used. If this is a local description, the "ice-lite"
attribute MUST NOT be specified.
o If the RTP/RTCP multiplexing policy is "require", each m= section
MUST contain an "a=rtcp-mux" attribute. If 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
following additional checks are applied:
o The session description must follow the rules defined in
[RFC3264], Section 6, including the requirement that the number of
m= sections MUST exactly match the number of m= sections in the
associated offer.
o For each m= section, the media type and protocol values MUST
exactly match the media type and protocol values in the
corresponding m= section in the associated offer.
If any of the preceding checks failed, processing MUST stop and an
error MUST be returned.
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5.8. Applying a Local Description
The following steps are performed at the media engine level to apply
a local description. If an error is returned, the session MUST be
restored to the state it was in before performing these steps.
Next, m= sections are processed. For each m= section, the following
steps MUST be performed; if any parameters are out of bounds, or
cannot be applied, processing MUST stop and an error MUST be
returned.
o If this m= section is new, begin gathering candidates for it, as
defined in [RFC5245], Section 4.1.1, unless it has been marked as
bundle-only.
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
ICE restart, and begin gathering new candidates for the m= section
as described in [RFC5245], Section 9.1.1.1. If this description
is an answer, also start checks on that media section as defined
in [RFC5245], Section 9.3.1.1.
o If the m= section proto value indicates use of RTP:
* If there is no RtpTransceiver associated with this m= section
(which will only happen when applying an offer), find one and
associate it with this m= section according to the following
steps:
+ Find the RtpTransceiver that corresponds to this m= section,
using the mapping between transceivers and m= section
indices established when creating the offer.
+ Set the value of this RtpTransceiver's mid property to the
MID of the m= section.
* If RTCP mux is indicated, prepare to demux RTP and RTCP from
the RTP ICE component, as specified in [RFC5761],
Section 5.1.3. If RTCP mux is not indicated, but was
previously negotiated, i.e., the RTCP ICE component no longer
exists, this MUST result in an error.
* For each specified RTP header extension, establish a mapping
between the extension ID and URI, as described in section 6 of
[RFC5285]. If any indicated RTP header extension is not
supported, this MUST result in an error.
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* If the MID header extension is supported, prepare to demux RTP
streams intended for this m= section based on the MID header
extension, as described in
[I-D.ietf-mmusic-sdp-bundle-negotiation], Section 14.
* For each specified media format, establish a mapping between
the payload type and the actual media format, as described in
[RFC3264], Section 6.1. If any indicated media format is not
supported, this MUST result in an error.
* For each specified "rtx" media format, establish a mapping
between the RTX payload type and its associated primary payload
type, as described in [RFC4588], Sections 8.6 and 8.7. If any
referenced primary payload types are not present, this MUST
result in an error.
* If the directional attribute is of type "sendrecv" or
"recvonly", enable receipt and decoding of media.
Finally, if this description is of type "pranswer" or "answer",
follow the processing defined in the Section 5.10 section below.
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"
is listed as an attribute, update the PeerConnection canTrickle
property to be true. Otherwise, set this property to false.
The following steps MUST be performed for attributes at the session
level; if any parameters are out of bounds, or cannot be applied,
processing MUST stop and an error MUST be returned.
o For any specified "CT" bandwidth value, set this as the limit for
the maximum total bitrate for all m= sections, as specified in
Section 5.8 of [RFC4566]. Within this overall limit, the
implementation can dynamically decide how to best allocate the
available bandwidth between m= sections, respecting any specific
limits that have been specified for individual m= sections.
o For any specified "RR" or "RS" bandwidth values, handle as
specified in [RFC3556], Section 2.
o Any "AS" bandwidth value MUST be ignored, as the meaning of this
construct at the session level is not well defined.
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For each m= section, the following steps MUST be performed; if any
parameters are out of bounds, or cannot be applied, processing MUST
stop and an error MUST be returned.
o If the PeerConnection state is "have-local-offer", and the ICE
ufrag or password changed from the previous remote description,
then an ICE restart is needed, as described in Section 9.1.1.1 of
[RFC5245]. If the description is of type "offer", note that an
ICE restart is needed. If the description is of type "answer" or
"pranswer" and the current local description is also an ICE
restart, then signal the ICE agent to begin checks as described in
Section 9.3.1.1 of [RFC5245]. An answerer MUST change the ufrag
and password in an answer if and only if ICE is restarting, as
described in Section 9.2.1.1 of [RFC5245].
o If the PeerConnection state is "have-remote-pranswer", and the ICE
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
use the supplied ICE remote ufrag and password for their
connectivity checks.
o Pair any supplied ICE candidates with any gathered local
candidates, as described in Section 5.7 of [RFC5245] and start
connectivity checks with the appropriate credentials.
o If an "a=end-of-candidates" attribute is present, process the end-
of-candidates indication as described in [I-D.ietf-ice-trickle]
Section 11.
o If the m= section proto value indicates use of RTP:
* If the m= section is being recycled (see Section 5.2.2),
dissociate the currently associated RtpTransceiver by setting
its mid property to null, and discard the mapping between the
transceiver and its m= section index.
* If the m= section is not associated with any RtpTransceiver
(possibly because it was dissociated in the previous step),
either find an RtpTransceiver or create one according to the
following steps:
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+ If the m= section is sendrecv or recvonly, and there are
RtpTransceivers of the same type that were added to the
PeerConnection by addTrack and are not associated with any
m= section and are not stopped, find the first (according to
the canonical order described in Section 5.2.1) such
RtpTransceiver.
+ If no RtpTransceiver was found in the previous step, create
one with a recvonly direction.
+ Associate the found or created RtpTransceiver with the m=
section by setting the value of the RtpTransceiver's mid
property to the MID of the m= section, and establish a
mapping between the transceiver and the index of the m=
section. If the m= section does not include a MID (i.e.,
the remote endpoint does not support the MID extension),
generate a value for the RtpTransceiver mid property,
following the guidance for "a=mid" mentioned in
Section 5.2.1.
* For each specified media format that is also supported by the
local implementation, establish a mapping between the specified
payload type and the media format, as described in [RFC3264],
Section 6.1. Specifically, this means that the implementation
records the payload type to be used in outgoing RTP packets
when sending each specified media format, as well as the
relative preference for each format that is indicated in their
ordering. If any indicated media format is not supported by
the local implementation, it MUST be ignored.
* For each specified "rtx" media format, establish a mapping
between the RTX payload type and its associated primary payload
type, as described in [RFC4588], Section 4. If any referenced
primary payload types are not present, this MUST result in an
error.
* For each specified fmtp parameter that is supported by the
local implementation, enable them on the associated media
formats.
* For each specified RTP header extension that is also supported
by the local implementation, establish a mapping between the
extension ID and URI, as described in [RFC5285], Section 5.
Specifically, this means that the implementation records the
extension ID to be used in outgoing RTP packets when sending
each specified header extension. If any indicated RTP header
extension is not supported by the local implementation, it MUST
be ignored.
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* For each specified RTCP feedback mechanism that is supported by
the local implementation, enable them on the associated media
formats.
* For any specified "TIAS" bandwidth value, set this value as a
constraint on the maximum RTP bitrate to be used when sending
media, as specified in [RFC3890]. If a "TIAS" value is not
present, but an "AS" value is specified, generate a "TIAS"
value using this formula:
TIAS = AS * 1000 * 0.95 - 50 * 40 * 8
The 50 is based on 50 packets per second, the 40 is based on an
estimate of total header size, the 1000 changes the unit from
kbps to bps (as required by TIAS), and the 0.95 is to allocate
5% to RTCP. "TIAS" is used in preference to "AS" because it
provides more accurate control of bandwidth.
* For any "RR" or "RS" bandwidth values, handle as specified in
[RFC3556], Section 2.
* Any specified "CT" bandwidth value MUST be ignored, as the
meaning of this construct at the media level is not well
defined.
* If the m= section is of type audio:
+ For each specified "CN" media format, enable DTX for all
supported media formats with the same clockrate, as
described in [RFC3389], Section 5, except for formats that
have their own internal DTX mechanisms. DTX for such
formats (e.g., Opus) is controlled via fmtp parameters, as
discussed in Section 5.2.3.2.
+ For each specified "telephone-event" media format, enable
DTMF transmission for all supported media formats with the
same clockrate, as described in [RFC4733], Section 2.5.1.2.
If the application attempts to transmit DTMF when using a
media format that does not have a corresponding telephone-
event format, this MUST result in an error.
+ For any specified "ptime" value, configure the available
media formats to use the specified packet size. If the
specified size is not supported for a media format, use the
next closest value instead.
Finally, if this description is of type "pranswer" or "answer",
follow the processing defined in the Section 5.10 section below.
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5.10. Applying an Answer
In addition to the steps mentioned above for processing a local or
remote description, the following steps are performed when processing
a description of type "pranswer" or "answer".
For each m= section, the following steps MUST be performed:
o If the m= section has been rejected (i.e. port is set to zero in
the answer), stop any reception or transmission of media for this
section, and, unless a non-rejected m= section is bundled with
this m= section, discard any associated ICE components, as
described in Section 9.2.1.3 of [RFC5245].
o If the remote DTLS fingerprint has been changed or the dtls-id has
changed, tear down the DTLS connection. This includes the case
when the PeerConnection state is "have-remote-pranswer". If a
DTLS connection needs to be torn down but the answer does not
indicate an ICE restart or, in the case of "have-remote-pranswer",
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
connection, using the specified roles and fingerprints, on any
underlying ICE components, once they are active.
o If the m= section proto value indicates use of RTP:
* If the m= section references any media formats, RTP header
extensions, or RTCP feedback mechanisms that were not present
in the corresponding m= section in the offer, this indicates a
negotiation problem and MUST result in an error.
* If the m= section has RTCP mux enabled, discard the RTCP ICE
component, if one exists, and begin or continue muxing RTCP
over the RTP ICE component, as specified in [RFC5761],
Section 5.1.3. Otherwise, prepare to transmit RTCP over the
RTCP ICE component; if no RTCP ICE component exists, because
RTCP mux was previously enabled, this MUST result in an error.
* If the m= section has reduced-size RTCP enabled, configure the
RTCP transmission for this m= section to use reduced-size RTCP,
as specified in [RFC5506].
* If the directional attribute in the answer is of type
"sendrecv" or "sendonly", choose the media format to send as
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the most preferred media format from the remote description
that is also present in the answer, as described in [RFC3264],
Sections 6.1 and 7, and start transmitting RTP media once the
underlying transport layers have been established. If an SSRC
has not already been chosen for this outgoing RTP stream,
choose a random one. If media is already being transmitted,
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
determine payload types for the outgoing RTP streams, including
the payload type for the send media format chosen above. Any
RTP header extensions that were negotiated should be included
in the outgoing RTP streams, using the extension mapping from
the remote description; if the RID header extension has been
negotiated, and RID values are specified, include the RID
header extension in the outgoing RTP streams, as indicated in
[I-D.ietf-mmusic-rid], Section 4.
* If simulcast has been negotiated, send the number of Source RTP
Streams as specified in [I-D.ietf-mmusic-sdp-simulcast],
Section 6.2.2.
* If the send media format chosen above has a corresponding "rtx"
media format, or a FEC mechanism has been negotiated, establish
a Redundancy RTP Stream with a random SSRC for each Source RTP
Stream, and start or continue transmitting RTX/FEC packets as
needed.
* If the send media format chosen above has a corresponding "red"
media format of the same clockrate, allow redundant encoding
using the specified format for resiliency purposes, as
discussed in [I-D.ietf-rtcweb-fec], Section 3.2. Note that
unlike RTX or FEC media formats, the "red" format is
transmitted on the Source RTP Stream, not the Redundancy RTP
Stream.
* Enable the RTCP feedback mechanisms referenced in the media
section for all Source RTP Streams using the specified media
formats. Specifically, begin or continue sending the requested
feedback types and reacting to received feedback, as specified
in [RFC4585], Section 4.2. When sending RTCP feedback, follow
the rules and recommendations from
[I-D.ietf-avtcore-rtp-multi-stream], Section 5.4.1 to select
which SSRC to use.
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* If the directional attribute is of type "recvonly" or
"inactive", stop transmitting all RTP media, but continue
sending RTCP, as described in [RFC3264], Section 5.1.
o If the m= section proto value indicates use of SCTP:
* 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
connection, using the local SCTP port value from the local
description, and the remote SCTP port value from the remote
description, as described in [I-D.ietf-mmusic-sctp-sdp],
Section 10.2.
If the answer contains valid bundle groups, discard any ICE
components for the m= sections that will be bundled onto the primary
ICE components in each bundle, and begin muxing these m= sections
accordingly, as described in
[I-D.ietf-mmusic-sdp-bundle-negotiation], Section 8.2.
If the description is of type "answer", and there are still remaining
candidates in the ICE candidate pool, discard them.
6. Processing RTP/RTCP
When bundling, associating incoming RTP/RTCP with the proper m=
section is defined in [I-D.ietf-mmusic-sdp-bundle-negotiation]. When
not bundling, the proper m= section is clear from the ICE component
over which the RTP/RTCP is received.
Once the proper m= section(s) are known, RTP/RTCP is delivered to the
RtpTransceiver(s) associated with the m= section(s) and further
processing of the RTP/RTCP is done at the RtpTransceiver level. This
includes using RID [I-D.ietf-mmusic-rid] to distinguish between
multiple Encoded Streams, as well as determine which Source RTP
stream should be repaired by a given Redundancy RTP stream.
7. Examples
Note that this example section shows several SDP fragments. To
format in 72 columns, some of the lines in SDP have been split into
multiple lines, where leading whitespace indicates that a line is a
continuation of the previous line. In addition, some blank lines
have been added to improve readability but are not valid in SDP.
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More examples of SDP for WebRTC call flows can be found in
[I-D.nandakumar-rtcweb-sdp].
7.1. Simple Example
This section shows a very simple example that sets up a minimal audio
/ video call between two JSEP endpoints without using trickle ICE.
The example in the following section provides a more detailed example
of what could happen in a JSEP session.
The code flow below shows Alice's endpoint initiating the session to
Bob's endpoint. The messages from Alice's JS to Bob's JS are assumed
to flow over some signaling protocol via a web server. The JS on
both Alice's side and Bob's side waits for all candidates before
sending the offer or answer, so the offers and answers are complete;
trickle ICE is not used. Both Alice and Bob are using the default
bundle policy of "balanced", and the default RTCP mux policy of
"require".
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// set up local media state
AliceJS->AliceUA: create new PeerConnection
AliceJS->AliceUA: addTrack with two tracks: audio and video
AliceJS->AliceUA: createOffer to get offer
AliceJS->AliceUA: setLocalDescription with offer
AliceUA->AliceJS: multiple onicecandidate events with candidates
// wait for ICE gathering to complete
AliceUA->AliceJS: onicecandidate event with null candidate
AliceJS->AliceUA: get |offer-A1| from pendingLocalDescription
// |offer-A1| is sent over signaling protocol to Bob
AliceJS->WebServer: signaling with |offer-A1|
WebServer->BobJS: signaling with |offer-A1|
// |offer-A1| arrives at Bob
BobJS->BobUA: create a PeerConnection
BobJS->BobUA: setRemoteDescription with |offer-A1|
BobUA->BobJS: ontrack events for audio and video tracks
// Bob accepts call
BobJS->BobUA: addTrack with local tracks
BobJS->BobUA: createAnswer
BobJS->BobUA: setLocalDescription with answer
BobUA->BobJS: multiple onicecandidate events with candidates
// wait for ICE gathering to complete
BobUA->BobJS: onicecandidate event with null candidate
BobJS->BobUA: get |answer-A1| from currentLocalDescription
// |answer-A1| is sent over signaling protocol to Alice
BobJS->WebServer: signaling with |answer-A1|
WebServer->AliceJS: signaling with |answer-A1|
// |answer-A1| arrives at Alice
AliceJS->AliceUA: setRemoteDescription with |answer-A1|
AliceUA->AliceJS: ontrack events for audio and video tracks
// media flows
BobUA->AliceUA: media sent from Bob to Alice
AliceUA->BobUA: media sent from Alice to Bob
The SDP for |offer-A1| looks like:
v=0
o=- 4962303333179871722 1 IN IP4 0.0.0.0
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s=-
t=0 0
a=ice-options:trickle
a=group:BUNDLE a1 v1
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=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:47017fee-b6c1-4162-929c-a25110252400
f83006c5-a0ff-4e0a-9ed9-d3e6747be7d9
a=ice-ufrag:ETEn
a=ice-pwd:OtSK0WpNtpUjkY4+86js7ZQl
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:actpass
a=dtls-id:1
a=rtcp:10101 IN IP4 203.0.113.100
a=rtcp-mux
a=rtcp-rsize
a=candidate:1 1 udp 2113929471 203.0.113.100 10100 typ host
a=candidate:1 2 udp 2113929470 203.0.113.100 10101 typ host
a=end-of-candidates
m=video 10102 UDP/TLS/RTP/SAVPF 100 101
c=IN IP4 203.0.113.100
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 nack
a=rtcp-fb:100 nack pli
a=msid:47017fee-b6c1-4162-929c-a25110252400
f30bdb4a-5db8-49b5-bcdc-e0c9a23172e0
a=ice-ufrag:BGKk
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a=ice-pwd:mqyWsAjvtKwTGnvhPztQ9mIf
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:actpass
a=dtls-id:1
a=rtcp:10103 IN IP4 203.0.113.100
a=rtcp-mux
a=rtcp-rsize
a=candidate:1 1 udp 2113929471 203.0.113.100 10102 typ host
a=candidate:1 2 udp 2113929470 203.0.113.100 10103 typ host
a=end-of-candidates
The SDP for |answer-A1| looks like:
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v=0
o=- 6729291447651054566 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 10200 UDP/TLS/RTP/SAVPF 96 0 8 97 98
c=IN IP4 203.0.113.200
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:61317484-2ed4-49d7-9eb7-1414322a7aae
5a7b57b8-f043-4bd1-a45d-09d4dfa31226
a=ice-ufrag:6sFv
a=ice-pwd:cOTZKZNVlO9RSGsEGM63JXT2
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
a=dtls-id:1
a=rtcp-mux
a=rtcp-rsize
a=candidate:1 1 udp 2113929471 203.0.113.200 10200 typ host
a=end-of-candidates
m=video 10200 UDP/TLS/RTP/SAVPF 100 101
c=IN IP4 203.0.113.200
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:61317484-2ed4-49d7-9eb7-1414322a7aae
4ea4d4a1-2fda-4511-a9cc-1b32c2e59552
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7.2. Detailed Example
This section shows a more involved example of a session between two
JSEP endpoints. Trickle ICE is used in full trickle mode, with a
bundle policy of "max-bundle", an RTCP mux policy of "require", and a
single TURN server. Initially, both Alice and Bob establish an audio
channel and a data channel. Later, Bob adds two video flows, one for
his video feed, and one for screensharing, both supporting FEC, and
with the video feed configured for simulcast. Alice accepts these
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
AliceJS->AliceUA: create new PeerConnection
AliceJS->AliceUA: addTrack with an audio track
AliceJS->AliceUA: createDataChannel to get data channel
AliceJS->AliceUA: createOffer to get |offer-B1|
AliceJS->AliceUA: setLocalDescription with |offer-B1|
// |offer-B1| is sent over signaling protocol to Bob
AliceJS->WebServer: signaling with |offer-B1|
WebServer->BobJS: signaling with |offer-B1|
// |offer-B1| arrives at Bob
BobJS->BobUA: create a PeerConnection
BobJS->BobUA: setRemoteDescription with |offer-B1|
BobUA->BobJS: ontrack with audio track from Alice
// candidates are sent to Bob
AliceUA->AliceJS: onicecandidate (host) |offer-B1-candidate-1|
AliceJS->WebServer: signaling with |offer-B1-candidate-1|
AliceUA->AliceJS: onicecandidate (srflx) |offer-B1-candidate-2|
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 |offer-B1-candidate-1|
BobJS->BobUA: addIceCandidate with |offer-B1-candidate-1|
WebServer->BobJS: signaling with |offer-B1-candidate-2|
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
BobJS->BobUA: addTrack with local audio
BobJS->BobUA: createDataChannel to get data channel
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BobJS->BobUA: createAnswer to get |answer-B1|
BobJS->BobUA: setLocalDescription with |answer-B1|
// |answer-B1| is sent to Alice
BobJS->WebServer: signaling with |answer-B1|
WebServer->AliceJS: signaling with |answer-B1|
AliceJS->AliceUA: setRemoteDescription with |answer-B1|
AliceUA->AliceJS: ontrack event with audio track from Bob
// candidates are sent to Alice
BobUA->BobJS: onicecandidate (host) |answer-B1-candidate-1|
BobJS->WebServer: signaling with |answer-B1-candidate-1|
BobUA->BobJS: onicecandidate (srflx) |answer-B1-candidate-2|
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 |answer-B1-candidate-1|
AliceJS->AliceUA: addIceCandidate with |answer-B1-candidate-1|
WebServer->AliceJS: signaling with |answer-B1-candidate-2|
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
BobUA->BobJS: ondatachannel event
AliceUA->AliceJS: ondatachannel event
BobUA->BobJS: onopen
AliceUA->AliceJS: onopen
// media is flowing between endpoints
BobUA->AliceUA: audio+data sent from Bob to Alice
AliceUA->BobUA: audio+data sent from Alice to Bob
// some time later Bob adds two video streams
// note, no candidates exchanged, because of bundle
BobJS->BobUA: addTrack with first video stream
BobJS->BobUA: addTrack with second video stream
BobJS->BobUA: createOffer to get |offer-B2|
BobJS->BobUA: setLocalDescription with |offer-B2|
// |offer-B2| is sent to Alice
BobJS->WebServer: signaling with |offer-B2|
WebServer->AliceJS: signaling with |offer-B2|
AliceJS->AliceUA: setRemoteDescription with |offer-B2|
AliceUA->AliceJS: ontrack event with first video track
AliceUA->AliceJS: ontrack event with second video track
AliceJS->AliceUA: createAnswer to get |answer-B2|
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AliceJS->AliceUA: setLocalDescription with |answer-B2|
// |answer-B2| is sent over signaling protocol to Bob
AliceJS->WebServer: signaling with |answer-B2|
WebServer->BobJS: signaling with |answer-B2|
BobJS->BobUA: setRemoteDescription with |answer-B2|
// media is flowing between endpoints
BobUA->AliceUA: audio+video+data sent from Bob to Alice
AliceUA->BobUA: audio+video+data sent from Alice to Bob
The SDP for |offer-B1| looks like:
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v=0
o=- 4962303333179871723 1 IN IP4 0.0.0.0
s=-
t=0 0
a=ice-options:trickle
a=group:BUNDLE a1 d1
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:57017fee-b6c1-4162-929c-a25110252400
e83006c5-a0ff-4e0a-9ed9-d3e6747be7d9
a=ice-ufrag:ATEn
a=ice-pwd:AtSK0WpNtpUjkY4+86js7ZQl
a=fingerprint:sha-256
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
a=setup:actpass
a=dtls-id:1
a=rtcp-mux
a=rtcp-mux-only
a=rtcp-rsize
m=application 0 UDP/DTLS/SCTP webrtc-datachannel
c=IN IP4 0.0.0.0
a=mid:d1
a=sctp-port:5000
a=max-message-size:65536
a=bundle-only
|offer-B1-candidate-1| looks like:
ufrag ATEn
index 0
mid a1
attr candidate:1 1 udp 2113929471 203.0.113.100 10100 typ host
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|offer-B1-candidate-2| 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
|offer-B1-candidate-3| looks like:
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:
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v=0
o=- 7729291447651054566 1 IN IP4 0.0.0.0
s=-
t=0 0
a=ice-options:trickle
a=group:BUNDLE a1 d1
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:71317484-2ed4-49d7-9eb7-1414322a7aae
6a7b57b8-f043-4bd1-a45d-09d4dfa31226
a=ice-ufrag:7sFv
a=ice-pwd:dOTZKZNVlO9RSGsEGM63JXT2
a=fingerprint:sha-256
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=dtls-id:1
a=rtcp-mux
a=rtcp-mux-only
a=rtcp-rsize
m=application 9 UDP/DTLS/SCTP webrtc-datachannel
c=IN IP4 0.0.0.0
a=mid:d1
a=sctp-port:5000
a=max-message-size:65536
|answer-B1-candidate-1| looks like:
ufrag 7sFv
index 0
mid a1
attr candidate:1 1 udp 2113929471 203.0.113.200 10200 typ host
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|answer-B1-candidate-2| 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
|answer-B1-candidate-3| looks like:
ufrag 7sFv
index 0
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
o=- 7729291447651054566 2 IN IP4 0.0.0.0
s=-
t=0 0
a=ice-options:trickle
a=group:BUNDLE a1 d1 v1 v2
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
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a=msid:71317484-2ed4-49d7-9eb7-1414322a7aae
6a7b57b8-f043-4bd1-a45d-09d4dfa31226
a=ice-ufrag:7sFv
a=ice-pwd:dOTZKZNVlO9RSGsEGM63JXT2
a=fingerprint:sha-256
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=dtls-id:1
a=rtcp-mux
a=rtcp-mux-only
a=rtcp-rsize
a=candidate:1 1 udp 2113929471 203.0.113.200 10200 typ host
a=candidate:1 1 udp 1845494015 198.51.100.200 11200 typ srflx
raddr 203.0.113.200 rport 10200
a=candidate:1 1 udp 255 192.0.2.200 12200 typ relay
raddr 198.51.100.200 rport 11200
a=end-of-candidates
m=application 12200 UDP/DTLS/SCTP webrtc-datachannel
c=IN IP4 192.0.2.200
a=mid:d1
a=sctp-port:5000
a=max-message-size:65536
m=video 12200 UDP/TLS/RTP/SAVPF 100 101 102
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=rtpmap:102 flexfec/90000
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:71317484-2ed4-49d7-9eb7-1414322a7aae
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
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a=rtpmap:100 VP8/90000
a=rtpmap:101 rtx/90000
a=fmtp:101 apt=100
a=rtpmap:102 flexfec/90000
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:81317484-2ed4-49d7-9eb7-1414322a7aae
6ea4d4a1-2fda-4511-a9cc-1b32c2e59552
The SDP for |answer-B2| is shown below. In addition to the
acceptance of the video m= sections, the use of a=recvonly to
indicate one-way video, and the use of a=imageattr to limit the
received resolution, note the use of setup:passive to maintain the
existing DTLS roles.
v=0
o=- 4962303333179871723 2 IN IP4 0.0.0.0
s=-
t=0 0
a=ice-options:trickle
a=group:BUNDLE a1 d1 v1 v2
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:57017fee-b6c1-4162-929c-a25110252400
e83006c5-a0ff-4e0a-9ed9-d3e6747be7d9
a=ice-ufrag:ATEn
a=ice-pwd:AtSK0WpNtpUjkY4+86js7ZQl
a=fingerprint:sha-256
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
a=setup:passive
a=dtls-id:1
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a=rtcp-mux
a=rtcp-mux-only
a=rtcp-rsize
a=candidate:1 1 udp 2113929471 203.0.113.100 10100 typ host
a=candidate:1 1 udp 1845494015 198.51.100.100 11100 typ srflx
raddr 203.0.113.100 rport 10100
a=candidate:1 1 udp 255 192.0.2.100 12100 typ relay
raddr 198.51.100.100 rport 11100
a=end-of-candidates
m=application 12100 UDP/DTLS/SCTP webrtc-datachannel
c=IN IP4 192.0.2.100
a=mid:d1
a=sctp-port:5000
a=max-message-size:65536
m=video 12100 UDP/TLS/RTP/SAVPF 100 101
c=IN IP4 192.0.2.100
a=mid:v1
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
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
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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 warmup 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
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// |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 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:
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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
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:actpass
a=dtls-id:1
a=rtcp-mux
a=rtcp-mux-only
a=rtcp-rsize
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 nack
a=rtcp-fb:100 nack pli
a=msid:bbce3ba6-abfc-ac63-d00a-e15b286f8fce
ac701365-eb06-42df-cc93-7f22bc308789
a=bundle-only
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|offer-C1-candidate-1| looks like:
ufrag 4ZcD
index 0
mid a1
attr candidate:1 1 udp 255 192.0.2.100 12100 typ relay
raddr 0.0.0.0 rport 0
The SDP for |answer-C1| looks like:
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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: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
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|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
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: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
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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=dtls-id:1
a=rtcp-mux
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a=rtcp-mux-only
a=rtcp-rsize
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 nack
a=rtcp-fb:100 nack pli
a=msid:bbce3ba6-abfc-ac63-d00a-e15b286f8fce
ac701365-eb06-42df-cc93-7f22bc308789
8. Security Considerations
The IETF has published separate documents
[I-D.ietf-rtcweb-security-arch] [I-D.ietf-rtcweb-security] describing
the security architecture for WebRTC as a whole. The remainder of
this section describes security considerations for this document.
While formally the JSEP interface is an API, it is better to think of
it is an Internet protocol, with the JS being untrustworthy from the
perspective of the endpoint. Thus, the threat model of [RFC3552]
applies. In particular, JS can call the API in any order and with
any inputs, including malicious ones. This is particularly relevant
when we consider the SDP which is passed to setLocalDescription().
While correct API usage requires that the application pass in SDP
which was derived from createOffer() or createAnswer(), there is no
guarantee that applications do so. The JSEP implementation MUST be
prepared for the JS to pass in bogus data instead.
Conversely, the application programmer MUST recognize that the JS
does not have complete control of endpoint behavior. One case that
bears particular mention is that editing ICE candidates out of the
SDP or suppressing trickled candidates does not have the expected
behavior: implementations will still perform checks from those
candidates even if they are not sent to the other side. Thus, for
instance, it is not possible to prevent the remote peer from learning
your public IP address by removing server reflexive candidates.
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Applications which wish to conceal their public IP address should
instead configure the ICE agent to use only relay candidates.
9. IANA Considerations
This document requires no actions from IANA.
10. Acknowledgements
Harald Alvestrand, Taylor Brandstetter, Suhas Nandakumar, and Peter
Thatcher provided significant text for this draft. Bernard Aboba,
Adam Bergkvist, Dan Burnett, Ben Campbell, Alissa Cooper, Richard
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.
11. References
11.1. Normative References
[I-D.ietf-avtcore-rtp-multi-stream]
Lennox, J., Westerlund, M., Wu, Q., and C. Perkins,
"Sending Multiple RTP Streams in a Single RTP Session",
draft-ietf-avtcore-rtp-multi-stream-11 (work in progress),
December 2015.
[I-D.ietf-avtext-rid]
Roach, A., Nandakumar, S., and P. Thatcher, "RTP Stream
Identifier (RID) Source Description (SDES)", draft-ietf-
avtext-rid-00 (work in progress), February 2016.
[I-D.ietf-ice-trickle]
Ivov, E., Rescorla, E., Uberti, J., and P. Saint-Andre,
"Trickle ICE: Incremental Provisioning of Candidates for
the Interactive Connectivity Establishment (ICE)
Protocol".
[I-D.ietf-mmusic-4572-update]
Holmberg, C., "Updates to RFC 4572", draft-ietf-mmusic-
4572-update-05 (work in progress), June 2016.
[I-D.ietf-mmusic-dtls-sdp]
Holmberg, C. and R. Shpount, "Using the SDP Offer/Answer
Mechanism for DTLS", draft-ietf-mmusic-dtls-sdp-14 (work
in progress), July 2016.
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[I-D.ietf-mmusic-msid]
Alvestrand, H., "Cross Session Stream Identification in
the Session Description Protocol", draft-ietf-mmusic-
msid-01 (work in progress), August 2013.
[I-D.ietf-mmusic-mux-exclusive]
Holmberg, C., "Indicating Exclusive Support of RTP/RTCP
Multiplexing using SDP", draft-ietf-mmusic-mux-
exclusive-08 (work in progress), June 2016.
[I-D.ietf-mmusic-rid]
Thatcher, P., Zanaty, M., Nandakumar, S., Burman, B.,
Roach, A., and B. Campen, "RTP Payload Format
Constraints", draft-ietf-mmusic-rid-04 (work in progress),
February 2016.
[I-D.ietf-mmusic-sctp-sdp]
Loreto, S. and G. Camarillo, "Stream Control Transmission
Protocol (SCTP)-Based Media Transport in the Session
Description Protocol (SDP)", draft-ietf-mmusic-sctp-sdp-04
(work in progress), June 2013.
[I-D.ietf-mmusic-sdp-bundle-negotiation]
Holmberg, C., Alvestrand, H., and C. Jennings,
"Multiplexing Negotiation Using Session Description
Protocol (SDP) Port Numbers", draft-ietf-mmusic-sdp-
bundle-negotiation-04 (work in progress), June 2013.
[I-D.ietf-mmusic-sdp-mux-attributes]
Nandakumar, S., "A Framework for SDP Attributes when
Multiplexing", draft-ietf-mmusic-sdp-mux-attributes-01
(work in progress), February 2014.
[I-D.ietf-mmusic-sdp-simulcast]
Burman, B., Westerlund, M., Nandakumar, S., and M. Zanaty,
"Using Simulcast in SDP and RTP Sessions", draft-ietf-
mmusic-sdp-simulcast-04 (work in progress), February 2016.
[I-D.ietf-rtcweb-audio]
Valin, J. and C. Bran, "WebRTC Audio Codec and Processing
Requirements", draft-ietf-rtcweb-audio-02 (work in
progress), August 2013.
[I-D.ietf-rtcweb-fec]
Uberti, J., "WebRTC Forward Error Correction
Requirements", draft-ietf-rtcweb-fec-00 (work in
progress), February 2015.
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[I-D.ietf-rtcweb-rtp-usage]
Perkins, C., Westerlund, M., and J. Ott, "Web Real-Time
Communication (WebRTC): Media Transport and Use of RTP",
draft-ietf-rtcweb-rtp-usage-09 (work in progress),
September 2013.
[I-D.ietf-rtcweb-security]
Rescorla, E., "Security Considerations for WebRTC", draft-
ietf-rtcweb-security-06 (work in progress), January 2014.
[I-D.ietf-rtcweb-security-arch]
Rescorla, E., "WebRTC Security Architecture", draft-ietf-
rtcweb-security-arch-09 (work in progress), February 2014.
[I-D.ietf-rtcweb-video]
Roach, A., "WebRTC Video Processing and Codec
Requirements", draft-ietf-rtcweb-video-00 (work in
progress), July 2014.
[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
Requirement Levels", BCP 14, RFC 2119, March 1997.
[RFC3261] Rosenberg, J., Schulzrinne, H., Camarillo, G., Johnston,
A., Peterson, J., Sparks, R., Handley, M., and E.
Schooler, "SIP: Session Initiation Protocol", RFC 3261,
June 2002.
[RFC3264] Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model
with Session Description Protocol (SDP)", RFC 3264, June
2002.
[RFC3552] Rescorla, E. and B. Korver, "Guidelines for Writing RFC
Text on Security Considerations", BCP 72, RFC 3552, July
2003.
[RFC3605] Huitema, C., "Real Time Control Protocol (RTCP) attribute
in Session Description Protocol (SDP)", RFC 3605, October
2003.
[RFC3890] Westerlund, M., "A Transport Independent Bandwidth
Modifier for the Session Description Protocol (SDP)", RFC
3890, DOI 10.17487/RFC3890, September 2004,
<http://www.rfc-editor.org/info/rfc3890>.
[RFC4145] Yon, D. and G. Camarillo, "TCP-Based Media Transport in
the Session Description Protocol (SDP)", RFC 4145,
September 2005.
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[RFC4566] Handley, M., Jacobson, V., and C. Perkins, "SDP: Session
Description Protocol", RFC 4566, July 2006.
[RFC4572] Lennox, J., "Connection-Oriented Media Transport over the
Transport Layer Security (TLS) Protocol in the Session
Description Protocol (SDP)", RFC 4572, July 2006.
[RFC4585] Ott, J., Wenger, S., Sato, N., Burmeister, C., and J. Rey,
"Extended RTP Profile for Real-time Transport Control
Protocol (RTCP)-Based Feedback (RTP/AVPF)", RFC 4585, July
2006.
[RFC5245] Rosenberg, J., "Interactive Connectivity Establishment
(ICE): A Protocol for Network Address Translator (NAT)
Traversal for Offer/Answer Protocols", RFC 5245, April
2010.
[RFC5285] Singer, D. and H. Desineni, "A General Mechanism for RTP
Header Extensions", RFC 5285, July 2008.
[RFC5761] Perkins, C. and M. Westerlund, "Multiplexing RTP Data and
Control Packets on a Single Port", RFC 5761, April 2010.
[RFC5888] Camarillo, G. and H. Schulzrinne, "The Session Description
Protocol (SDP) Grouping Framework", RFC 5888, June 2010.
[RFC6236] Johansson, I. and K. Jung, "Negotiation of Generic Image
Attributes in the Session Description Protocol (SDP)", RFC
6236, May 2011.
[RFC6347] Rescorla, E. and N. Modadugu, "Datagram Transport Layer
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
Real-time Transport Protocol (SRTP)", RFC 6904, April
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>.
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[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'
Field for Transporting RTP Media over TCP under Various
RTP Profiles", RFC 7850, DOI 10.17487/RFC7850, April 2016,
<http://www.rfc-editor.org/info/rfc7850>.
[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
[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]
Uberti, J. and G. Shieh, "WebRTC IP Address Handling
Recommendations", draft-ietf-rtcweb-ip-handling-01 (work
in progress), March 2016.
[I-D.nandakumar-rtcweb-sdp]
Nandakumar, S. and C. Jennings, "SDP for the WebRTC",
draft-nandakumar-rtcweb-sdp-02 (work in progress), July
2013.
[RFC3389] Zopf, R., "Real-time Transport Protocol (RTP) Payload for
Comfort Noise (CN)", RFC 3389, September 2002.
[RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V.
Jacobson, "RTP: A Transport Protocol for Real-Time
Applications", STD 64, RFC 3550, DOI 10.17487/RFC3550,
July 2003, <http://www.rfc-editor.org/info/rfc3550>.
[RFC3556] Casner, S., "Session Description Protocol (SDP) Bandwidth
Modifiers for RTP Control Protocol (RTCP) Bandwidth", RFC
3556, July 2003.
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[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
Tone Generation in the Session Initiation Protocol (SIP)",
RFC 3960, December 2004.
[RFC4568] Andreasen, F., Baugher, M., and D. Wing, "Session
Description Protocol (SDP) Security Descriptions for Media
Streams", RFC 4568, July 2006.
[RFC4588] Rey, J., Leon, D., Miyazaki, A., Varsa, V., and R.
Hakenberg, "RTP Retransmission Payload Format", RFC 4588,
July 2006.
[RFC4733] Schulzrinne, H. and T. Taylor, "RTP Payload for DTMF
Digits, Telephony Tones, and Telephony Signals", RFC 4733,
DOI 10.17487/RFC4733, December 2006,
<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
Real-Time Transport Control Protocol (RTCP): Opportunities
and Consequences", RFC 5506, April 2009.
[RFC5576] Lennox, J., Ott, J., and T. Schierl, "Source-Specific
Media Attributes in the Session Description Protocol
(SDP)", RFC 5576, June 2009.
[RFC5763] Fischl, J., Tschofenig, H., and E. Rescorla, "Framework
for Establishing a Secure Real-time Transport Protocol
(SRTP) Security Context Using Datagram Transport Layer
Security (DTLS)", RFC 5763, May 2010.
[RFC5764] McGrew, D. and E. Rescorla, "Datagram Transport Layer
Security (DTLS) Extension to Establish Keys for the Secure
Real-time Transport Protocol (SRTP)", RFC 5764, May 2010.
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[RFC6464] Lennox, J., Ed., Ivov, E., and E. Marocco, "A Real-time
Transport Protocol (RTP) Header Extension for Client-to-
Mixer Audio Level Indication", RFC 6464, DOI 10.17487/
RFC6464, December 2011,
<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
B. Burman, Ed., "A Taxonomy of Semantics and Mechanisms
for Real-Time Transport Protocol (RTP) Sources", RFC 7656,
DOI 10.17487/RFC7656, November 2015,
<http://www.rfc-editor.org/info/rfc7656>.
[TS26.114]
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]
Bergkvist, A., Burnett, D., Narayanan, A., and C.
Jennings, "WebRTC 1.0: Real-time Communication Between
Browsers", World Wide Web Consortium WD WD-webrtc-
20140617, June 2014,
<http://www.w3.org/TR/2011/WD-webrtc-20140617>.
Appendix A. Appendix A
For the syntax validation performed in Section 5.7, the following
list of ABNF definitions is used:
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+------------------------+------------------------------------------+
| Attribute | Reference |
+------------------------+------------------------------------------+
| ptime | [RFC4566] Section 9 |
| maxptime | [RFC4566] Section 9 |
| rtpmap | [RFC4566] Section 9 |
| recvonly | [RFC4566] Section 9 |
| sendrecv | [RFC4566] Section 9 |
| sendonly | [RFC4566] Section 9 |
| inactive | [RFC4566] Section 9 |
| framerate | [RFC4566] Section 9 |
| fmtp | [RFC4566] Section 9 |
| quality | [RFC4566] Section 9 |
| rtcp | [RFC3605] Section 2.1 |
| setup | [RFC4145] Sections 3, 4, and 5 |
| connection | [RFC4145] Sections 3, 4, and 5 |
| fingerprint | [RFC4572] Section 5 |
| rtcp-fb | [RFC4585] Section 4.2 |
| candidate | [RFC5245] Section 15.1 |
| remote-candidates | [RFC5245] Section 15.2 |
| ice-lite | [RFC5245] Section 15.3 |
| ice-ufrag | [RFC5245] Section 15.4 |
| ice-pwd | [RFC5245] Section 15.4 |
| ice-options | [RFC5245] Section 15.5 |
| extmap | [RFC5285] Section 7 |
| mid | [RFC5888] Section 4 and 5 |
| group | [RFC5888] Section 4 and 5 |
| imageattr | [RFC6236] Section 3.1 |
| extmap (encrypt | [RFC6904] Section 4 |
| option) | |
| msid | [I-D.ietf-mmusic-msid] Section 2 |
| rid | [I-D.ietf-mmusic-rid] Section 10 |
| simulcast | [I-D.ietf-mmusic-sdp-simulcast] Section |
| | 6.1 |
| dtls-id | [I-D.ietf-mmusic-dtls-sdp] Section 4 |
+------------------------+------------------------------------------+
Table 1: SDP ABNF References
Appendix B. Change log
Note: This section will be removed by RFC Editor before publication.
Changes in draft-20:
o Remove Appendix-B.
Changes in draft-19:
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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:
o Update demux algorithm and move it to an appendix in preparation
for merging it into BUNDLE.
o Clarify why we can't handle an incoming offer to send simulcast.
o Expand IceCandidate object text.
o Further document use of ICE candidate pool.
o Document removeTrack.
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o Update requirements to only accept the last generated offer/answer
as an argument to setLocalDescription.
o Allow round pixels.
o Fix code around default timing when AVPF is not specified.
o Clean up terminology around m= line and m=section.
o Provide a more realistic example for minimum decoder capabilities.
o Document behavior when rtcp-mux policy is require but rtcp-mux
attribute not provided.
o Expanded discussion of RtpSender and RtpReceiver.
o Add RtpTransceiver.currentDirection and document setDirection.
o Require imageattr x=0, y=0 to indicate that there are no valid
resolutions.
o Require a privacy-preserving MID/RID construction.
o Require support for RFC 3556 bandwidth modifiers.
o Update maxptime description.
o Note that endpoints may encounter extra codecs in answers and
subsequent offers from non-JSEP peers.
o Update references.
Changes in draft-17:
o Split createOffer and createAnswer sections to clearly indicate
attributes which always appear and which only appear when not
bundled into another m= section.
o Add descriptions of RtpTransceiver methods.
o Describe how to process RTCP feedback attributes.
o Clarify transceiver directions and their interaction with 3264.
o Describe setCodecPreferences.
o Update RTP demux algorithm. Include RTCP.
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o Update requirements for when a=rtcp is included, limiting to cases
where it is needed for backward compatibility.
o Clarify SAR handling.
o Updated addTrack matching algorithm.
o Remove a=ssrc requirements.
o Handle a=setup in reoffers.
o Discuss how RTX/FEC should be handled.
o Discuss how telephone-event should be handled.
o Discuss how CN/DTX should be handled.
o Add missing references to ABNF table.
Changes in draft-16:
o Update addIceCandidate to indicate ICE generation and allow per-m=
section end-of-candidates.
o Update fingerprint handling to use draft-ietf-mmusic-4572-update.
o Update text around SDP processing of RTP header extensions and
payload formats.
o Add sections on simulcast, addTransceiver, and createDataChannel.
o Clarify text to ensure that the session ID is a positive 63 bit
integer.
o Clarify SDP processing for direction indication.
o Describe SDP processing for rtcp-mux-only.
o Specify how SDP session version in o= line.
o Require that when doing an re-offer, the capabilities of the new
session are mostly required to be a subset of the previously
negotiated session.
o Clarified ICE restart interaction with bundle-only.
o Remove support for changing SDP before calling
setLocalDescription.
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o Specify algorithm for demuxing RTP based on MID, PT, and SSRC.
o Clarify rules for rejecting m= lines when bundle policy is
balanced or max-bundle.
Changes in draft-15:
o Clarify text around codecs offered in subsequent transactions to
refer to what's been negotiated.
o Rewrite LS handling text to indicate edge cases and that we're
living with them.
o Require that answerer reject m= lines when there are no codecs in
common.
o Enforce max-bundle on offer processing.
o Fix TIAS formula to handle bits vs. kilobits.
o Describe addTrack algorithm.
o Clean up references.
Changes in draft-14:
o Added discussion of RtpTransceivers + RtpSenders + RtpReceivers,
and how they interact with createOffer/createAnswer.
o Removed obsolete OfferToReceiveX options.
o Explained how addIceCandidate can be used for end-of-candidates.
Changes in draft-13:
o Clarified which SDP lines can be ignored.
o Clarified how to handle various received attributes.
o Revised how attributes should be generated for bundled m= lines.
o Remove unused references.
o Remove text advocating use of unilateral PTs.
o Trigger an ICE restart even if the ICE candidate policy is being
made more strict.
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o Remove the 'public' ICE candidate policy.
o Move open issues into GitHub issues.
o Split local/remote description accessors into current/pending.
o Clarify a=imageattr handling.
o Add more detail on VoiceActivityDetection handling.
o Reference draft-shieh-rtcweb-ip-handling.
o Make it clear when an ICE restart should occur.
o Resolve changes needed in references.
o Remove MSID semantics.
o ice-options are now at session level.
o Default RTCP mux policy is now 'require'.
Changes in draft-12:
o Filled in sections on applying local and remote descriptions.
o Discussed downscaling and upscaling to fulfill imageattr
requirements.
o Updated what SDP can be modified by the application.
o Updated to latest datachannel SDP.
o Allowed multiple fingerprint lines.
o Switched back to IPv4 for dummy candidates.
o Added additional clarity on ICE default candidates.
Changes in draft-11:
o Clarified handling of RTP CNAMEs.
o Updated what SDP lines should be processed or ignored.
o Specified how a=imageattr should be used.
Changes in draft-10:
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o Described video size negotiation with imageattr.
o Clarified rejection of sections that do not have mux-only.
o Add handling of LS groups
Changes in draft-09:
o Don't return null for {local,remote}Description after close().
o Changed TCP/TLS to UDP/DTLS in RTP profile names.
o Separate out bundle and mux policy.
o Added specific references to FEC mechanisms.
o Added canTrickle mechanism.
o Added section on subsequent answers and, answer options.
o Added text defining set{Local,Remote}Description behavior.
Changes in draft-08:
o Added new example section and removed old examples in appendix.
o Fixed <proto> field handling.
o Added text describing a=rtcp attribute.
o Reworked handling of OfferToReceiveAudio and OfferToReceiveVideo
per discussion at IETF 90.
o Reworked trickle ICE handling and its impact on m= and c= lines
per discussion at interim.
o Added max-bundle-and-rtcp-mux policy.
o Added description of maxptime handling.
o Updated ICE candidate pool default to 0.
o Resolved open issues around AppID/receiver-ID.
o Reworked and expanded how changes to the ICE configuration are
handled.
o Some reference updates.
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o Editorial clarification.
Changes in draft-07:
o Expanded discussion of VAD and Opus DTX.
o Added a security considerations section.
o Rewrote the section on modifying SDP to require implementations to
clearly indicate whether any given modification is allowed.
o Clarified impact of IceRestart on CreateOffer in local-offer
state.
o Guidance on whether attributes should be defined at the media
level or the session level.
o Renamed "default" bundle policy to "balanced".
o Removed default ICE candidate pool size and clarify how it works.
o Defined a canonical order for assignment of MSTs to m= lines.
o Removed discussion of rehydration.
o Added Eric Rescorla as a draft editor.
o Cleaned up references.
o Editorial cleanup
Changes in draft-06:
o Reworked handling of m= line recycling.
o Added handling of BUNDLE and bundle-only.
o Clarified handling of rollback.
o Added text describing the ICE Candidate Pool and its behavior.
o Allowed OfferToReceiveX to create multiple recvonly m= sections.
Changes in draft-05:
o Fixed several issues identified in the createOffer/Answer sections
during document review.
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o Updated references.
Changes in draft-04:
o Filled in sections on createOffer and createAnswer.
o Added SDP examples.
o Fixed references.
Changes in draft-03:
o Added text describing relationship to W3C specification
Changes in draft-02:
o Converted from nroff
o Removed comparisons to old approaches abandoned by the working
group
o Removed stuff that has moved to W3C specification
o Align SDP handling with W3C draft
o Clarified section on forking.
Changes in draft-01:
o Added diagrams for architecture and state machine.
o Added sections on forking and rehydration.
o Clarified meaning of "pranswer" and "answer".
o Reworked how ICE restarts and media directions are controlled.
o Added list of parameters that can be changed in a description.
o Updated suggested API and examples to match latest thinking.
o Suggested API and examples have been moved to an appendix.
Changes in draft -00:
o Migrated from draft-uberti-rtcweb-jsep-02.
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Authors' Addresses
Justin Uberti
Google
747 6th St S
Kirkland, WA 98033
USA
Email: justin@uberti.name
Cullen Jennings
Cisco
400 3rd Avenue SW
Calgary, AB T2P 4H2
Canada
Email: fluffy@iii.ca
Eric Rescorla (editor)
Mozilla
331 Evelyn Ave
Mountain View, CA 94041
USA
Email: ekr@rtfm.com
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