JavaScript Session Establishment Protocol (JSEP)
RFC 8829
| Document | Type | RFC - Proposed Standard (January 2021) IPR | |
|---|---|---|---|
| Authors | Justin Uberti , Cullen Fluffy Jennings , Eric Rescorla | ||
| Last updated | 2021-01-20 | ||
| 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 2017-07-06 | ||
| IESG | IESG state | RFC 8829 (Proposed Standard) | |
| Action Holders |
(None)
|
||
| Consensus boilerplate | Yes | ||
| Telechat date | (None) | ||
| Responsible AD | Adam Roach | ||
| Send notices to | Ted Hardie <ted.ietf@gmail.com> | ||
| IANA | IANA review state | IANA OK - No Actions Needed | |
| IANA action state | No IANA Actions |
RFC 8829
Internet Engineering Task Force (IETF) J. Uberti
Request for Comments: 8829 Google
Category: Standards Track C. Jennings
ISSN: 2070-1721 Cisco
E. Rescorla, Ed.
Mozilla
January 2021
JavaScript Session Establishment Protocol (JSEP)
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 is an Internet Standards Track document.
This document is a product of the Internet Engineering Task Force
(IETF). It represents the consensus of the IETF community. It has
received public review and has been approved for publication by the
Internet Engineering Steering Group (IESG). Further information on
Internet Standards is available in Section 2 of RFC 7841.
Information about the current status of this document, any errata,
and how to provide feedback on it may be obtained at
https://www.rfc-editor.org/info/rfc8829.
Copyright Notice
Copyright (c) 2021 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
(https://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
the Trust Legal Provisions and are provided without warranty as
described in the Simplified BSD License.
Table of Contents
1. Introduction
1.1. General Design of JSEP
1.2. Other Approaches Considered
1.3. Contradiction regarding bundle-only "m=" sections
2. Terminology
3. Semantics and Syntax
3.1. Signaling Model
3.2. Session Descriptions and State Machine
3.3. Session Description Format
3.4. Session Description Control
3.4.1. RtpTransceivers
3.4.2. RtpSenders
3.4.3. RtpReceivers
3.5. ICE
3.5.1. ICE Gathering Overview
3.5.2. ICE Candidate Trickling
3.5.2.1. ICE Candidate Format
3.5.3. ICE Candidate Policy
3.5.4. ICE Candidate Pool
3.5.5. ICE Versions
3.6. Video Size Negotiation
3.6.1. Creating an imageattr Attribute
3.6.2. Interpreting imageattr Attributes
3.7. Simulcast
3.8. Interactions with Forking
3.8.1. Sequential Forking
3.8.2. Parallel Forking
4. Interface
4.1. PeerConnection
4.1.1. Constructor
4.1.2. addTrack
4.1.3. removeTrack
4.1.4. addTransceiver
4.1.5. onaddtrack Event
4.1.6. createDataChannel
4.1.7. ondatachannel Event
4.1.8. createOffer
4.1.9. createAnswer
4.1.10. SessionDescriptionType
4.1.10.1. Use of Provisional Answers
4.1.10.2. Rollback
4.1.11. setLocalDescription
4.1.12. setRemoteDescription
4.1.13. currentLocalDescription
4.1.14. pendingLocalDescription
4.1.15. currentRemoteDescription
4.1.16. pendingRemoteDescription
4.1.17. canTrickleIceCandidates
4.1.18. setConfiguration
4.1.19. addIceCandidate
4.1.20. onicecandidate Event
4.2. RtpTransceiver
4.2.1. stop
4.2.2. stopped
4.2.3. setDirection
4.2.4. direction
4.2.5. currentDirection
4.2.6. setCodecPreferences
5. SDP Interaction Procedures
5.1. Requirements Overview
5.1.1. Usage Requirements
5.1.2. Profile Names and Interoperability
5.2. Constructing an Offer
5.2.1. Initial Offers
5.2.2. Subsequent Offers
5.2.3. Options Handling
5.2.3.1. IceRestart
5.2.3.2. VoiceActivityDetection
5.3. Generating an Answer
5.3.1. Initial Answers
5.3.2. Subsequent Answers
5.3.3. Options Handling
5.3.3.1. VoiceActivityDetection
5.4. Modifying an Offer or Answer
5.5. Processing a Local Description
5.6. Processing a Remote Description
5.7. Processing a Rollback
5.8. Parsing a Session Description
5.8.1. Session-Level Parsing
5.8.2. Media Section Parsing
5.8.3. Semantics Verification
5.9. Applying a Local Description
5.10. Applying a Remote Description
5.11. Applying an Answer
6. Processing RTP/RTCP
7. Examples
7.1. Simple Example
7.2. Detailed Example
7.3. Early Transport Warmup Example
8. Security Considerations
9. IANA Considerations
10. References
10.1. Normative References
10.2. Informative References
Appendix A. SDP ABNF Syntax
Acknowledgements
Authors' Addresses
1. Introduction
This document describes how the W3C Web Real-Time Communication
(WebRTC) RTCPeerConnection interface [W3C.webrtc] is used to control
the setup, management, and teardown of a multimedia session.
1.1. General Design of JSEP
WebRTC call setup has been designed to focus on controlling the media
plane, leaving signaling-plane behavior 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 call signaling
protocol, 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 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), which 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 Interactive
Connectivity Establishment (ICE) state machine [RFC8445]. 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, that in many cases it will
actually be between a JSEP endpoint and some kind of server, such as
a gateway or Multipoint Control Unit (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 configuration
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.
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 [RFC8445], JSEP decouples the ICE state machine from
the overall signaling state machine. The ICE state machine must
remain in the JSEP implementation because only the implementation has
the necessary knowledge of candidates and other transport
information. Performing this separation provides additional
flexibility in protocols that decouple session descriptions from
transport. For instance, in traditional SIP, each offer or answer is
self-contained, including both the session descriptions and the
transport information. However, [RFC8840] allows SIP to be used with
Trickle ICE [RFC8838], in which the session description can be sent
immediately and the transport information can be sent when available.
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. JSEP's
decoupling of the ICE and signaling state machines allows it to
accommodate either model.
Although it abstracts signaling, the JSEP approach requires that the
application 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 make life easier for the application 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 provide implementations of the SIP [RFC3261] and
Extensible Messaging and Presence Protocol (XMPP) [RFC6120] signaling
protocols atop the JSEP API. 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 (see
[RFC3264], Section 4).
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 the argument that requiring
exposure of this level of complexity to the application programmer
would not be beneficial; it would (1) result in an API where even a
simple example would require a significant amount of code to
orchestrate all the needed interactions and (2) create a large API
surface that would need 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.
1.3. Contradiction regarding bundle-only "m=" sections
Since the approval of the WebRTC specification documents, the IETF
has become aware of an inconsistency between the document specifying
JSEP and the document specifying BUNDLE (this RFC and [RFC8843],
respectively). Rather than delaying publication further to come to a
resolution, the documents are being published as they were originally
approved. The IETF intends to restart work on these technologies,
and revised versions of these documents will be published as soon as
a resolution becomes available.
The specific issue involves the handling of "m=" sections that are
designated as bundle-only, as discussed in Section 4.1.1 of this RFC.
Currently, there is divergence between JSEP and BUNDLE, as well as
between these specifications and existing browser implementations:
* JSEP prescribes that said "m=" sections should use port zero and
add an "a=bundle-only" attribute in initial offers, but not in
answers or subsequent offers.
* BUNDLE prescribes that these "m=" sections should be marked as
described in the previous point, but in all offers and answers.
* Most current browsers do not mark any "m=" sections with port zero
and instead use the same port for all bundled "m=" sections; some
others follow the JSEP behavior.
2. Terminology
The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
"SHOULD", "SHOULD NOT", "RECOMMENDED", "NOT RECOMMENDED", "MAY", and
"OPTIONAL" in this document are to be interpreted as described in
BCP 14 [RFC2119] [RFC8174] when, and only when, they appear in all
capitals, as shown here.
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
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 JSEP implementation 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; some parameters are
declarative, and the remote side must either accept them or reject
them altogether. An example of such a parameter is the TLS
fingerprints [RFC8122] as used in the context of DTLS [RFC6347];
these fingerprints are calculated based on the local certificate(s)
offered and are not subject to negotiation.
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.
Lastly, while the exact media parameters are known only after an
offer and an answer have been exchanged, the offerer may receive ICE
checks, and possibly media (e.g., in the case of a re-offer after a
connection has been established) before it receives 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 JSEP
implementation 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]), and the answerer, who first
calls setRemoteDescription(sdp [offer]) and then later
setLocalDescription(sdp [answer]).
During the offer/answer exchange, the outstanding offer is considered
to be "pending" at the offerer and the answerer, as it may be either
accepted or rejected. If this is a re-offer, each side will also
have "current" local and remote descriptions, which reflect the
result of the last offer/answer exchange. Sections 4.1.14, 4.1.16,
4.1.13, and 4.1.15 provide more detail on pending and current
descriptions.
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, Traversal Using Relays around NAT (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 JSEP level,
a new offer can be generated at any point. For example, when using
SIP for signaling, if one offer is sent and is then canceled 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
JavaScript application needs one for the signaling. The answerer can
send back zero or more provisional answers and then 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 |
+---------------+ | +---------------+ |
| |----/ | |----/
| have- | setLocal(PRANSWER) | have- |
| remote-offer |------------------- >| local-pranswer|
| | | |
| | | |
+---------------+ +---------------+
^ | |
| | setLocal(ANSWER) |
setRemote(OFFER) | |
| V setLocal(ANSWER) |
+---------------+ |
| | |
| |<---------------------------+
| stable |
| |<---------------------------+
| | |
+---------------+ setRemote(ANSWER) |
^ | |
| | setLocal(OFFER) |
setRemote(ANSWER) | |
| V |
+---------------+ +---------------+
| | | |
| have- | setRemote(PRANSWER) |have- |
| local-offer |------------------- >|remote-pranswer|
| | | |
| |----\ | |----\
+---------------+ | +---------------+ |
^ | ^ |
| | | |
\-----/ \-----/
setLocal(OFFER) setRemote(PRANSWER)
Figure 2: JSEP State Machine
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 Session Description Protocol (SDP)
syntax for their internal representation. While this format is not
optimal for manipulation from JavaScript, it is widely accepted and
is 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 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.
As detailed below, 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 that 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/createAnswer calls.
3.4.1. RtpTransceivers
RtpTransceivers allow the application to control the RTP media
associated with one "m=" section. Each RtpTransceiver has an
RtpSender and an RtpReceiver, which an application can use to control
the sending and receiving of RTP media. The application may also
modify the RtpTransceiver directly, for instance, by stopping it.
RtpTransceivers generally have a 1:1 mapping with "m=" sections,
although there may be more RtpTransceivers than "m=" sections when
RtpTransceivers are created but not yet associated with an "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 media identification (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, an "m=" section
is always associated with exactly one RtpTransceiver. However, in
certain cases where an "m=" section has been rejected, as discussed
in Section 5.2.2 below, that "m=" section will be "recycled" and
associated with a new RtpTransceiver with a new MID value.
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 the RTP Control Protocol (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 by 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
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 locally or remotely initiated ICE restart.
When a new gathering phase starts, the ICE agent will notify the
application that gathering is occurring through a state change event.
Then, when each new ICE candidate becomes available, the ICE agent
will supply it to the application via an onicecandidate event; these
candidates will also automatically be added to the current and/or
pending local session description. Finally, when all candidates have
been gathered, a final onicecandidate 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, if an "m=" section is
bundled (either by a successful bundle negotiation or by being marked
as bundle-only), then candidates will be gathered and exchanged for
that "m=" section if and only if its MID item is a BUNDLE-tag, as
described in [RFC8843].
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 [RFC8838]. 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 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 that 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.
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
[RFC8839], Section 5.1. Note that this field does not contain an
"a=" prefix, as indicated in the following example:
candidate:1 1 UDP 1694498815 192.0.2.33 10000 typ host
The IceCandidate object contains a field to indicate which ICE
username fragment (ufrag) it is associated with, as defined in
[RFC8839], Section 5.4. 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 an "m=" section index or by 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 5.10
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.
Implementations MUST be prepared to receive objects with some fields
missing, as mentioned above.
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
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 [RFC8828].
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 that 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
the local description indicates the number of ICE components that
will be needed and for which candidates must be gathered. 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 prepares 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 are 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.
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. ("STUN"
stands for "Session Traversal Utilities for NAT".)
3.5.5. ICE Versions
While this specification formally relies on [RFC8445], at the time of
its publication, the majority of WebRTC implementations support the
version of ICE described in [RFC5245]. The "ice2" attribute defined
in [RFC8445] can be used to detect the version in use by a remote
endpoint and to provide a smooth transition from the older
specification to the newer one. Implementations MUST be able to
accept remote descriptions that do not have the "ice2" attribute.
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. By specifying these limits in an
"a=imageattr" attribute, JSEP endpoints can attempt to ensure that
the remote sender transmits video at an acceptable resolution.
However, when communicating with a non-JSEP endpoint that does not
understand this attribute, any signaled limits may be exceeded, and
the JSEP implementation MUST handle this gracefully, e.g., by
discarding the video.
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 combine any known local limits (e.g.,
hardware decoder capabilities or 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.
If these local limits preclude receiving any video, i.e., the
degenerate case of no permitted resolutions, the "a=imageattr"
attribute MUST be omitted, and the "m=" section MUST be marked as
sendonly/inactive, as appropriate.
Otherwise, an "a=imageattr" attribute is created with a "recv"
direction, and the resulting resolution space formed from the
aforementioned intersection is used to specify its minimum and
maximum "x=" and "y=" 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,
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 imageattr Attributes
[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 behavior that is more specific. When a
MediaStreamTrack, which is producing video of a certain resolution
(the "track resolution"), is attached to an RtpSender, which is
encoding the track video at the same or lower resolution(s) (the
"encoder resolutions"), and a remote description is applied that
references the sender and contains valid "a=imageattr recv"
attributes, it MUST follow the rules below to ensure that the sender
does not transmit a resolution that would exceed the size criteria
specified in the attributes. These rules MUST be followed as long as
the attributes remain present in the remote description, including
cases in which the track changes its resolution or is replaced with a
different track.
Depending on how the RtpSender is configured, it may be producing a
single encoding at a certain resolution or, if simulcast
(Section 3.7) has been negotiated, multiple encodings, each at their
own specific resolution. In addition, depending on the
configuration, each encoding may have the flexibility to reduce
resolution when needed or may be locked to a specific output
resolution.
For each encoding being produced by the RtpSender, the set of
"a=imageattr recv" attributes in the corresponding "m=" section of
the remote description is processed to determine what should be
transmitted. Only attributes that reference the media format
selected for the encoding are considered; each such attribute is
evaluated individually, starting with the attribute with the highest
"q=" value. If multiple attributes have the same "q=" value, they
are evaluated in the order they appear in their containing "m="
section. Note that while JSEP endpoints will include at most one
"a=imageattr recv" attribute per media format, JSEP endpoints may
receive session descriptions from non-JSEP endpoints with "m="
sections that contain multiple such attributes.
For each "a=imageattr recv" attribute, the following rules are
applied. If this processing is successful, the encoding is
transmitted accordingly, and no further attributes are considered for
that encoding. Otherwise, the next attribute is evaluated, in the
aforementioned order. If none of the supplied attributes can be
processed successfully, the encoding MUST NOT be transmitted, and an
error SHOULD be raised to the application.
* The limits from the attribute are compared to the encoder
resolution. Only the specific limits mentioned below are
considered; any other values, such as picture aspect ratio, MUST
be ignored. When considering a MediaStreamTrack that is producing
rotated video, the unrotated resolution MUST be used for the
checks. This is required regardless of whether the receiver
supports performing receive-side rotation (e.g., through
Coordination of Video Orientation (CVO) [TS26.114]), as it
significantly simplifies the matching logic.
* If the attribute includes a "sar=" (sample aspect ratio) value set
to something other than "1.0", indicating that the receiver wants
to receive non-square pixels, this cannot be satisfied and the
attribute MUST NOT be used.
* If the encoder resolution exceeds the maximum size permitted by
the attribute and the encoder is allowed to adjust its resolution,
the encoder SHOULD apply downscaling in order to satisfy the
limits. Downscaling MUST NOT change the picture aspect ratio of
the encoding, ignoring any trivial differences due to rounding.
For example, if the encoder resolution is 1280x720 and the
attribute specified a maximum of 640x480, the expected output
resolution would be 640x360. If downscaling cannot be applied,
the attribute MUST NOT be used.
* If the encoder resolution is less than the minimum size permitted
by the attribute, the attribute MUST NOT be used; the encoder MUST
NOT apply upscaling. JSEP implementations SHOULD avoid this
situation by allowing receipt of arbitrarily small resolutions,
perhaps via fallback to a software decoder.
* If the encoder resolution is within the maximum and minimum sizes,
no action is needed.
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. Upon generation of an offer or answer,
these encodings are indicated via 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
ensure that 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
setting up simulcast in the case where the JSEP endpoint receives the
initial offer requires out-of-band signaling or SDP inspection.
However, in the case where the JSEP endpoint sets up simulcast in its
initial offer, any established simulcast streams will continue to
work upon receipt of an incoming re-offer. Future versions of this
specification may add additional APIs to handle the incoming initial
offer scenario.
When using JSEP to transmit multiple encodings from an RtpSender, the
techniques from [RFC8853] and [RFC8851] are used. Specifically, when
multiple encodings have been configured for an RtpSender, the "m="
section for the RtpSender will include an "a=simulcast" attribute, as
defined in [RFC8853], Section 5.1, 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 [RFC8851], Section 4;
the use of Restriction Identifiers (RIDs, also called rid-ids or
RtpStreamIds) 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 in 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 as follows:
* Start exchanging media with a given remote peer, but keep all the
resources reserved in the offer.
* 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 as either (1) a provisional answer, leaving open
the possibility of using a different answer in the future or (2) 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 [RFC3960],
Section 3.1. 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.
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
(see Section 4.1) and making sure that the same local media streams
have been added into this new PeerConnection. Then the new
PeerConnection object would produce an 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.
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 the 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 [RFC8843]. 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=" sections 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 N media streams. This policy
balances desire to multiplex with the need to ensure that 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 to be received by non-bundle-aware endpoints but will
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 trade-off 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" (Lip Synchronization) group when creating an offer or answer.
Adding tracks to the same "LS" group indicates that the playback of
these tracks should be synchronized for proper lip sync, as described
in [RFC5888], Section 7. 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.
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.direction is sendrecv) or as inactive (if
transceiver.direction 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 that 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. onaddtrack Event
The onaddtrack event is dispatched to the application when a new
remote track has been signaled as a result of a setRemoteDescription
call. The new track is supplied as a MediaStreamTrack object in the
event, along with the MediaStream(s) the track is part of.
4.1.6. 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 ("SCTP" stands for "Stream Control Transmission
Protocol") 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 that
are used by the PeerConnection (e.g., maxPacketLifetime) but are not
reflected in the SDP and do not affect the JSEP state.
4.1.7. ondatachannel Event
The ondatachannel event is dispatched to the application when a new
data channel has been negotiated by the remote side, which can occur
at any time after the underlying SCTP/DTLS association has been
established. The new data channel object is supplied in the event.
4.1.8. createOffer
The createOffer method generates a blob of SDP that contains an offer
per [RFC3264] with the supported configurations for the session,
including descriptions of the media added to this PeerConnection, the
codec, RTP, and 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
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 specification that defines 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 it does not change the PeerConnection state, trigger
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.
4.1.9. createAnswer
The createAnswer method generates a blob of SDP that contains an SDP
answer per [RFC3264] with the supported configuration for the session
that is compatible with the parameters supplied in the most recent
call to setRemoteDescription; setRemoteDescription 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
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 specification that defines 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 it does not change the PeerConnection state, trigger
candidate gathering, or cause a media state change. Specifically,
the answer is not applied, and does not become the current local
description, until setLocalDescription is called.
4.1.10. 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 MUST be parsed as an offer; said
description may include many possible media configurations. A
description used as an "offer" may be applied any time the
PeerConnection is in a "stable" state or applied as an update to a
previously supplied but unanswered "offer".
"pranswer" indicates that a description MUST be parsed as an answer,
but not a final answer, and so MUST 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 as an update to a previously sent
"pranswer".
"answer" indicates that a description MUST be parsed as an answer,
the offer/answer exchange MUST be considered complete, and any
resources (decoders, candidates) that are no longer needed SHOULD be
released. A description used as an "answer" may be applied as a
response to an "offer" or as 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
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 indicating that the
state machine MUST be rolled back to the previous "stable" state, as
described in Section 4.1.10.2. The contents MUST be empty.
4.1.10.1. 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 handshaking and DTLS handshaking have likely completed by this
point, can start transmitting immediately. The application will also
send a new offer to the remote side indicating call acceptance and
moving the audio and video to be two-way media. A detailed example
flow along these lines is shown in Section 7.3.
Of course, some applications may not be able to perform this double
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.
4.1.10.2. 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, needs to return to a
"stable" state and the previous local/remote description. To support
this, we introduce the concept of "rollback", which discards any
proposed changes to the session, returning the state machine to the
"stable" state. A rollback is performed by supplying a session
description of type "rollback" with empty contents to either
setLocalDescription or setRemoteDescription.
4.1.11. 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, final answer, or rollback; 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 it continues
until setRemoteDescription is called with either (1) a final answer,
at which point the PeerConnection can fully adopt the pending local
description or (2) 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 (1) setRemoteDescription was previously called with an offer, (2)
setLocalDescription is called with an answer (provisional or final),
(3) the media directions are compatible, and (4) media is available
to send, this will result in the starting of media transmission.
4.1.12. 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.
This API changes the local media state; among other things, it sets
up local resources for sending and encoding media.
If (1) setLocalDescription was previously called with an offer, (2)
setRemoteDescription is called with an answer (provisional or final),
(3) the media directions are compatible, and (4) media is available
to send, this will result in the starting of media transmission.
4.1.13. 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.
4.1.14. 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".
4.1.15. 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.
4.1.16. 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.
A null object will be returned if the state of the PeerConnection is
"stable" or "have-local-offer".
4.1.17. 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 that 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.
4.1.18. 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 calling this method
depend on when it is invoked, and they will differ, depending on
which specific parameters are changed:
* 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
existing candidates will be discarded, and new candidates will be
gathered from the new servers.
* 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 made available to the
application until a gathering phase occurs, and so any necessary
filtering can still be done on any pooled candidates.
* 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.
* The bundle and RTCP-multiplexing policies MUST NOT be changed
after the construction of the PeerConnection.
Calling this method may result in a change to the state of the ICE
agent.
4.1.19. addIceCandidate
The addIceCandidate method provides an update to the ICE agent via an
IceCandidate object (Section 3.5.2.1). If the IceCandidate's
candidate field is non-null, the IceCandidate is treated as a new
remote ICE candidate, which will be added to the current and/or
pending remote description according to the rules defined for Trickle
ICE. Otherwise, the IceCandidate is treated as an end-of-candidates
indication, as defined in [RFC8838], Section 14.
In either case, the "m=" section index, MID, and ufrag fields from
the supplied IceCandidate are used to determine which "m=" section
and ICE candidate generation the IceCandidate belongs to, as
described in Section 3.5.2.1 above. In the case of an end-of-
candidates indication, null values for the "m=" section index and MID
fields are interpreted to mean that the indication applies to all
"m=" sections in the specified ICE candidate generation. However, if
both fields are null for a new remote candidate, this MUST be treated
as an invalid condition, as specified below.
If any IceCandidate fields contain invalid values or an error occurs
during the processing of the IceCandidate object, the supplied
IceCandidate MUST be ignored and an error MUST be returned.
Otherwise, the new remote candidate or end-of-candidates indication
is supplied to the ICE agent. In the case of a new remote candidate,
connectivity checks will be sent to the new candidate, assuming
setLocalDescription has already been called to initialize the ICE
gathering process.
4.1.20. onicecandidate Event
The onicecandidate event is dispatched to the application in two
situations: (1) when the ICE agent has discovered a new allowed local
ICE candidate during ICE gathering, as outlined in Section 3.5.1 and
subject to the restrictions discussed in Section 3.5.3, or (2) when
an ICE gathering phase completes. The event contains a single
IceCandidate object, as defined in Section 3.5.2.1.
In the first case, the newly discovered candidate is reflected in the
IceCandidate object, and all of its fields MUST be non-null. This
candidate will also be added to the current and/or pending local
description according to the rules defined for Trickle ICE.
In the second case, the event's IceCandidate object MUST have its
candidate field set to null to indicate that the current gathering
phase is complete, i.e., there will be no further onicecandidate
events in this phase. However, the IceCandidate's ufrag field MUST
be specified to indicate which ICE candidate generation is ending.
The IceCandidate's "m=" section index and MID fields MAY be specified
to indicate that the event applies to a specific "m=" section, or set
to null to indicate it applies to all "m=" sections in the current
ICE candidate generation. This event can be used by the application
to generate an end-of-candidates indication, as defined in [RFC8838],
Section 13.
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 stop 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. The permitted values
for direction are "recvonly", "sendrecv", "sendonly", and "inactive",
mirroring the identically named direction attributes defined in
[RFC4566], Section 6.
When creating offers, the transceiver direction is directly reflected
in the output, even for re-offers. When creating answers, the
transceiver direction is intersected with the offered direction, as
explained in Section 5.3 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.
4.2.5. currentDirection
The currentDirection property indicates the last negotiated direction
for the transceiver's associated "m=" section. More specifically, it
indicates the direction attribute [RFC3264] of the associated "m="
section in the last applied answer (including provisional answers),
with "send" and "recv" directions reversed if it was a remote answer.
For example, if the direction 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.
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.
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.
* ICE, as specified in [RFC8445], MUST be used. Note that the
remote endpoint may use a lite implementation; implementations
MUST properly handle remote endpoints that use ICE-lite. The
remote endpoint may also use an older version of ICE;
implementations MUST properly handle remote endpoints that use ICE
as specified in [RFC5245].
* DTLS [RFC6347] or DTLS-SRTP [RFC5763] MUST be used, as appropriate
for the media type, as specified in [RFC8827].
The SDP security descriptions mechanism for SRTP keying [RFC4568]
MUST NOT be used, as discussed in [RFC8827].
5.1.2. Profile Names and Interoperability
For media "m=" sections, JSEP implementations MUST support the
"UDP/TLS/RTP/SAVPF" profile specified in [RFC5764] as well as the
"TCP/DTLS/RTP/SAVPF" profile specified in [RFC7850] and MUST indicate
one of these profiles for each media "m=" line they produce in an
offer. For data "m=" sections, implementations MUST support the
"UDP/DTLS/SCTP" profile as well as the "TCP/DTLS/SCTP" profile and
MUST indicate one of these profiles for each data "m=" line they
produce in an offer. The exact profile to use is determined by the
protocol associated with the current default or selected ICE
candidate, as described in [RFC8839], Section 4.2.1.2.
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/TLS/RTP/SAVPF" or "TCP/DTLS/RTP/SAVPF".
In order to simplify compatibility with such endpoints, JSEP
implementations MUST follow the following rules when processing the
media "m=" sections in a received offer:
* Any profile in the offer matching one of the following MUST be
accepted:
- "RTP/AVP" (defined in [RFC4566], Section 8.2.2)
- "RTP/AVPF" (defined in [RFC4585], Section 9)
- "RTP/SAVP" (defined in [RFC3711], Section 12)
- "RTP/SAVPF" (defined in [RFC5124], Section 6)
- "TCP/DTLS/RTP/SAVP" (defined in [RFC7850], Section 3.4)
- "TCP/DTLS/RTP/SAVPF" (defined in [RFC7850], Section 3.5)
- "UDP/TLS/RTP/SAVP" (defined in [RFC5764], Section 9)
- "UDP/TLS/RTP/SAVPF" (defined in [RFC5764], Section 9)
* The profile in any "m=" line in any generated answer MUST exactly
match the profile provided in the offer.
* 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" attributes. Note that lack of an
"a=fingerprint" attribute will lead to negotiation failure.
* 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".
* 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. This simplifies JSEP behavior,
with minimal downside, as any remote endpoint that fails to handle
such a re-offer will also fail to handle a JSEP endpoint's initial
offer.
5.2. Constructing an Offer
When createOffer is called, a new SDP description MUST be created
that includes the functionality specified in [RFC8834]. 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:
* The first SDP line MUST be "v=0" as defined in [RFC4566],
Section 5.1.
* The second SDP line MUST be an "o=" line as defined in [RFC4566],
Section 5.2. The value of the <username> field SHOULD be "-".
The <sess-id> MUST be representable by a 64-bit signed integer,
and the value MUST be less than 2^(63)-1. It is RECOMMENDED that
the <sess-id> be constructed by generating a 64-bit quantity with
the highest bit set to zero and the remaining 63 bits being
cryptographically random. The value of the <nettype> <addrtype>
<unicast-address> tuple SHOULD be set to a non-meaningful address,
such as IN IP4 0.0.0.0, to prevent leaking a local IP address in
this field; this problem is discussed in [RFC8828]. 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.
* The third SDP line MUST be a "s=" line as defined 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 in JSEP, having the same meaningless
value seems clearer.
* 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.
* Encryption Keys ("k=") lines do not provide sufficient security
and MUST NOT be included.
* 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".
* An "a=ice-options" line with the "trickle" and "ice2" options MUST
be added, as specified in [RFC8840], Section 4.1.1 and [RFC8445],
Section 10.
* If WebRTC identity is being used, an "a=identity" line MUST be
added, as described in [RFC8827], 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. If there are no
such RtpTransceivers, no "m=" sections are generated; more can be
added later, as discussed in [RFC3264], Section 5.
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
contain a unique set of ICE credentials and a 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 any and all certificates that
have been specified for the PeerConnection; as a result, they MUST
all have the same fingerprint value or values [RFC8122], or these
values MUST be session-level attributes.
Each "m=" section MUST be generated as specified in [RFC4566],
Section 5.14. For the "m=" line itself, the following rules MUST be
followed:
* If the "m=" section is marked as bundle-only, then the <port>
value MUST be set to zero. Otherwise, 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 default port value
of 9 (Discard) MUST be used, as indicated in [RFC8840],
Section 4.1.1.
* To properly indicate use of DTLS, the <proto> field MUST be set to
"UDP/TLS/RTP/SAVPF", as specified in [RFC5764], Section 8.
* 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.
* Unless excluded by the above restrictions, the media formats MUST
include the mandatory audio/video codecs as specified in
[RFC7874], Section 3 and [RFC7742], 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 default value "IN IP4
0.0.0.0", as defined in [RFC8840], Section 4.1.1.
[RFC8859] groups SDP attributes into different categories. To avoid
unnecessary duplication when bundling, attributes of category
IDENTICAL or TRANSPORT MUST NOT be repeated in bundled "m=" sections,
repeating the guidance from [RFC8843], Section 7.1.3. This includes
"m=" sections for which bundling has been negotiated and is still
desired, as well as "m=" sections marked as bundle-only.
The following attributes, which are of a category other than
IDENTICAL or TRANSPORT, MUST be included in each "m=" section:
* 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 [RFC8843], Section 15.2.
Note that this does not set the RtpTransceiver mid property, as
that only occurs when the description is applied. The generated
MID value can be considered a "proposed" MID at this point.
* A direction attribute that is the same as that of the associated
transceiver.
* 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.
* 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.
* For each supported Forward Error Correction (FEC) mechanism,
"a=rtpmap" and "a=fmtp" lines, as specified in [RFC4566],
Section 6. The FEC mechanisms that MUST be supported are
specified in [RFC8854], Section 7, and specific usage for each
media type is outlined in Sections 4 and 5.
* 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.
* If this "m=" section is for video media and there are known
limitations on the size of images that can be decoded, an
"a=imageattr" line, as specified in Section 3.6.
* 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 [RFC8834],
Section 5.2. Any header extensions that require encryption MUST
be specified as indicated in [RFC6904], Section 4.
* For each supported RTCP feedback mechanism, an "a=rtcp-fb" line,
as specified in [RFC4585], Section 4.2. The list of RTCP feedback
mechanisms that SHOULD/MUST be supported is specified in
[RFC8834], Section 5.1.
* 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 [RFC8830], Section 2, but omitting the
"appdata" field.
* If the RtpTransceiver has a sendrecv or sendonly direction, and
the application has specified a rid-id for an encoding, 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 [RFC8851], and its direction MUST be "send". If the
application has chosen a rid-id, it MUST be used; otherwise, a
rid-id MUST be generated by the implementation. rid-ids MUST be
generated in a fashion that does not leak user information, e.g.,
randomly or using a per-PeerConnection counter (see guidance at
the end of [RFC8852], Section 3.3), and SHOULD be 3 bytes or less,
to allow them to efficiently fit into the RTP header extensions
defined in [RFC8852], Section 3.3. If no encodings have been
specified, or only one encoding is specified but without a rid-id,
then no "a=rid" lines are generated.
* 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 [RFC8853], Section 5.1.
The associated set of rid-ids MUST include all of the rid-ids used
in the "a=rid" lines for this "m=" section.
* If (1) the bundle policy for this PeerConnection is set to "max-
bundle" and this is not the first "m=" section or (2) 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 that either have a
unique address or are associated with the BUNDLE-tag. (In initial
offers, this means those "m=" sections that do not contain an
"a=bundle-only" attribute.)
* "a=ice-ufrag" and "a=ice-pwd" lines, as specified in [RFC8839],
Section 5.4.
* For each desired digest algorithm, one or more "a=fingerprint"
lines for each of the endpoint's certificates, as specified in
[RFC8122], Section 5.
* 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".
* An "a=tls-id" line, as specified in [RFC8842], Section 5.2.
* An "a=rtcp" line, as specified in [RFC3605], Section 2.1,
containing the default value "9 IN IP4 0.0.0.0", because no
candidates have yet been gathered.
* An "a=rtcp-mux" line, as specified in [RFC5761], Section 5.1.3.
* If the RTP/RTCP multiplexing policy is "require", an "a=rtcp-mux-
only" line, as specified in [RFC8858], Section 4.
* An "a=rtcp-rsize" line, as specified in [RFC5506], Section 5.
Lastly, if a data channel has been created, an "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" [RFC8841]. The
<fmt> value MUST be set to "webrtc-datachannel" as specified in
[RFC8841], Section 4.2.2.
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 [RFC8841],
Section 5.1; and, if appropriate, an "a=max-message-size" line, as
defined in [RFC8841], 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):
* "a=ice-ufrag"
* "a=ice-pwd"
* "a=fingerprint"
* "a=setup"
* "a=tls-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 that SDP permits to be at either the session level or the
media level SHOULD generally be at the media level even if they are
identical. This assists development and debugging by making it
easier to understand individual media sections, especially if one of
a set of initially identical attributes is subsequently changed.
However, implementations MAY choose to aggregate attributes at the
session level, and JSEP implementations MUST be prepared to receive
attributes in either location.
Attributes other than the ones specified above MAY be included,
except for the following attributes, which are specifically
incompatible with the requirements of [RFC8834] and MUST NOT be
included:
* "a=crypto"
* "a=key-mgmt"
* "a=ice-lite"
Note that when bundle is used, any additional attributes that are
added MUST follow the advice in [RFC8859] 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.
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 previous offer was not applied using setLocalDescription,
meaning the PeerConnection is still in the "stable" state, the steps
for generating an initial offer MUST be followed, subject to the
following restriction:
* 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 previous offer was applied using setLocalDescription, but a
corresponding answer from the remote side has not yet been applied,
meaning the PeerConnection is still in the "have-local-offer" state,
an offer is generated by following the steps in the "stable" state
above, along with these exceptions:
* The "s=" and "t=" lines MUST stay the same.
* 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
(including a new MID value) and placing it at the same index as
the "m=" section with a zero port.
* 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.
* 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.
* For RtpTransceivers that are not stopped, the "a=msid" line or
lines 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.
* Each "m=" and "c=" line MUST be filled in with the port, relevant
RTP profile, and address of the default candidate for the "m="
section, as described in [RFC8839], Section 4.2.1.2 and clarified
in Section 5.1.2. If no RTP candidates have yet been gathered,
default values MUST still be used, as described above.
* Each "a=mid" line MUST stay the same.
* Each "a=ice-ufrag" and "a=ice-pwd" line MUST stay the same, unless
the ICE configuration has changed (e.g., changes to either 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.
* 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,
default values MUST be used, as described in Section 5.2.1 above.
* 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 [RFC8839], Section 5.1. If candidate
gathering for the section has completed, an "a=end-of-candidates"
attribute MUST be added, as described in [RFC8840], Section 8.2.
If the "m=" section is bundled into another "m=" section, both
"a=candidate" and "a=end-of-candidates" MUST be omitted.
* For RtpTransceivers that are still present, the "a=rid" lines MUST
stay the same.
* For RtpTransceivers that are still present, any "a=simulcast" line
MUST stay the same.
If the previous offer was applied using setLocalDescription, and a
corresponding answer from the remote side has been applied using
setRemoteDescription, meaning the PeerConnection is in the "have-
remote-pranswer" state or the "stable" state, an offer is generated
based on the negotiated session descriptions by following the steps
mentioned for the "have-local-offer" state above.
In addition, for each existing, 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 local
or remote description, as appropriate:
* The "m=" line and corresponding "a=rtpmap" and "a=fmtp" lines MUST
only include media formats that have not been excluded by the
codec preferences of the associated transceiver and also MUST
include all currently available formats. Media formats that were
previously offered but are no longer available (e.g., a shared
hardware codec) MAY be excluded.
* 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 most recent answer. Any media formats
that were not present in the most recent answer MUST be added
after all existing formats.
* The RTP header extensions MUST only include those that are present
in the most recent answer.
* The RTCP feedback mechanisms MUST only include those that are
present in the most recent answer, except for the case of format-
specific mechanisms that are referencing a newly added media
format.
* The "a=rtcp" line MUST NOT be added if the most recent answer
included an "a=rtcp-mux" line.
* The "a=rtcp-mux" line MUST be the same as that in the most recent
answer.
* The "a=rtcp-mux-only" line MUST NOT be added.
* The "a=rtcp-rsize" line MUST NOT be added unless present in the
most recent answer.
* An "a=bundle-only" line, as defined in [RFC8843], Section 6, MUST
NOT be added. Instead, JSEP implementations MUST simply omit
parameters in the IDENTICAL and TRANSPORT categories for bundled
"m=" sections, as described in [RFC8843], Section 7.1.3.
* 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
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 an 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 [RFC8839], Section 4.4.3.1.1. 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
Silence suppression, also known as discontinuous transmission
("DTX"), can reduce the bandwidth used for audio by switching to a
special encoding when voice activity is not detected, at the cost of
some fidelity.
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.
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. In addition, the implementation
MUST NOT use silence suppression for media it generates, regardless
of whether the "CN" codecs or related fmtp parameters appear in the
peer's description. The impact of these rules is that silence
suppression in JSEP depends on mutual agreement of both sides, which
ensures consistent handling regardless of which codec is used.
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 [RFC8834]. 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 Section 5.2.1 above, except that the "a=ice-options"
line, with the "trickle" option as specified in [RFC8840],
Section 4.1.3 and the "ice2" option as specified in [RFC8445],
Section 10, 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.
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
m=video 10001 UDP/TLS/RTP/SAVPF 96
a=mid:v1
a=msid:ms1
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
m=video 20001 UDP/TLS/RTP/SAVPF 96
a=mid:v1
a=msid:ms2
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
m=video 20001 UDP/TLS/RTP/SAVPF 96
a=mid:v1
a=msid:ms2b
Finally, if the answerer does not add any tracks, its transceivers
will not reference any MediaStreams, causing the preferences of the
offerer to be maintained, and so the subsequent answer will contain
an identical "LS" group.
a=group:LS a1 v1
m=audio 20000 UDP/TLS/RTP/SAVPF 0
a=mid:a1
a=recvonly
m=video 20001 UDP/TLS/RTP/SAVPF 96
a=mid:v1
a=recvonly
The example in Section 7.2 shows a more involved case of "LS" group
generation.
The next step is to generate an "m=" section 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. Each offered "m="
section will have an associated RtpTransceiver, as described in
Section 5.10. 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:
* The associated RtpTransceiver has been stopped.
* There is no offered media format that is both supported and, if
applicable, allowed by codec preferences.
* 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.
* 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.
* This "m=" section is in a bundle group, and the group's offerer
tagged "m=" section is being rejected due to one of the above
reasons. This requires all "m=" sections in the bundle group to
be rejected, as specified in [RFC8843], Section 7.3.3.
Otherwise, each "m=" section in the answer MUST then be generated as
specified in [RFC3264], Section 6.1. For the "m=" line itself, the
following rules MUST be followed:
* The <port> value would normally be set to the port of the default
ICE candidate for this "m=" section, but given that no candidates
are available yet, the default <port> value of 9 (Discard) MUST be
used, as indicated in [RFC8840], Section 4.1.1.
* The <proto> field MUST be set to exactly match the <proto> field
for the corresponding "m=" line in the offer.
* If codec preferences have been set for the associated transceiver,
media formats MUST be generated in the corresponding order,
regardless of what was offered, and MUST exclude any codecs not
present in the codec preferences.
* Otherwise, the media formats on the "m=" line MUST be generated in
the same order as those offered in the current remote description,
excluding any currently unsupported formats. Any currently
available media formats that are not present in the current remote
description MUST be added after all existing formats.
* In either case, the media formats in the answer MUST include at
least one format that is present in the offer but MAY include
formats that are locally supported but not present in the offer,
as mentioned in [RFC3264], Section 6.1. If no common format
exists, the "m=" section is rejected as described above.
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 default value "IN IP4
0.0.0.0", as defined in [RFC8840], Section 4.1.3.
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 or TRANSPORT):
* 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.
* 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.
* 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.
* If "rtx" is present in the offer, 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.
* 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 [RFC8854], Section 7, and
specific usage for each media type is outlined in Sections 4 and
5.
* 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.
* If this "m=" section is for video media and there are known
limitations on the size of images that can be decoded, an
"a=imageattr" line, as specified in Section 3.6.
* 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 [RFC8834], Section 5.2. Any header extensions that
require encryption MUST be specified as indicated in [RFC6904],
Section 4.
* For each supported RTCP feedback mechanism that is present in the
offer, an "a=rtcp-fb" line, as specified in [RFC4585],
Section 4.2. The list of RTCP feedback mechanisms that SHOULD/
MUST be supported is specified in [RFC8834], Section 5.1.
* 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 [RFC8830], Section 2, but omitting the
"appdata" field.
Each "m=" section that is not bundled into another "m=" section MUST
contain the following attributes (which are of category IDENTICAL or
TRANSPORT):
* "a=ice-ufrag" and "a=ice-pwd" lines, as specified in [RFC8839],
Section 5.4.
* For each desired digest algorithm, one or more "a=fingerprint"
lines for each of the endpoint's certificates, as specified in
[RFC8122], Section 5.
* 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". When
the offer contains the "actpass" value, as will always be the case
with JSEP endpoints, the answerer SHOULD use the "active" role.
Offers from non-JSEP endpoints MAY send other values for
"a=setup", in which case the answer MUST use a value consistent
with the value in the offer.
* An "a=tls-id" line, as specified in [RFC8842], Section 5.3.
* 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 default value
"9 IN IP4 0.0.0.0" (because no candidates have yet been gathered).
* 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, an "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 [RFC8841],
Section 5.1; and, if appropriate, an "a=max-message-size" line, as
defined in [RFC8841], 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:
* "a=ice-ufrag"
* "a=ice-pwd"
* "a=fingerprint"
* "a=setup"
* "a=tls-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 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, all "m=" sections in the
answer MUST NOT have an "a=bundle-only" line.
Attributes that are common between all "m=" sections MAY be moved to
the session level if explicitly defined to be valid at the 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 previous answer was not applied using setLocalDescription,
meaning the PeerConnection is still in the "have-remote-offer" state,
the steps for generating an initial answer MUST be followed, subject
to the following restriction:
* 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:
* The "s=" and "t=" lines MUST stay the same.
* 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 [RFC8839], Section 4.2.1.2. Note that in certain
cases, the "m=" line protocol may not match that of the default
candidate, because the "m=" line protocol value MUST match what
was supplied in the offer, as described above.
* 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 [RFC8839], Section 4.4.1.1.1. If
the "m=" section is bundled into another "m=" section, it still
MUST NOT contain any ICE credentials.
* Each "a=tls-id" line MUST stay the same, unless the offerer's
"a=tls-id" line changed, in which case a new tls-id value MUST be
created, as described in [RFC8842], Section 5.2.
* 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.
* RTCP multiplexing MUST be used, and an "a=rtcp-mux" line inserted
if and only if the "m=" section previously used RTCP multiplexing.
* 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, default
values MUST be used, as described in Section 5.3.1 above.
* 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 [RFC8839], Section 5.1. If candidate
gathering for the section has completed, an "a=end-of-candidates"
attribute MUST be added, as described in [RFC8840], Section 8.2.
If the "m=" section is bundled into another "m=" section, both
"a=candidate" and "a=end-of-candidates" MUST be omitted.
* 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
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 MUST
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"). The impact of this rule is that an answerer
will not try to use silence suppression with any endpoint that does
not offer it, making silence suppression support bilateral even with
non-JSEP endpoints.
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.
After calling setLocalDescription with an offer or answer, the
application MAY modify the SDP to reduce its capabilities before
sending it to the far side, as long as it follows the rules above
that define a valid JSEP offer or answer. Likewise, an application
that has received an offer or answer from a peer MAY modify the
received SDP, subject to the same constraints, before calling
setRemoteDescription.
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.
5.5. Processing a Local Description
When a SessionDescription is supplied to setLocalDescription, the
following steps MUST be performed:
* If the description is of type "rollback", follow the processing
defined in Section 5.7 and skip the processing described in the
rest of this section.
* Otherwise, 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".
* If the type is not correct for the current state, processing MUST
stop and an error MUST be returned.
* 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.
* Next, the SessionDescription is parsed into a data structure, as
described in Section 5.8 below.
* Finally, the parsed SessionDescription is applied as described in
Section 5.9 below.
5.6. Processing a Remote Description
When a SessionDescription is supplied to setRemoteDescription, the
following steps MUST be performed:
* If the description is of type "rollback", follow the processing
defined in Section 5.7 and skip the processing described in the
rest of this section.
* Otherwise, 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".
* If the type is not correct for the current state, processing MUST
stop and an error MUST be returned.
* Next, the SessionDescription is parsed into a data structure, as
described in Section 5.8 below. If parsing fails for any reason,
processing MUST stop and an error MUST be returned.
* Finally, the parsed SessionDescription is applied as described in
Section 5.10 below.
5.7. Processing a Rollback
A rollback may be performed if the PeerConnection is in any state
except for "stable". This means that both offers and provisional
answers can be rolled back. 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. If a rollback is
attempted in the "stable" state, processing MUST stop and an error
MUST be returned. Note that this implies that once the answerer has
performed setLocalDescription with its answer, this cannot be rolled
back.
The effect of rollback MUST be the same regardless of whether
setLocalDescription or setRemoteDescription is called.
In order to process rollback, a JSEP implementation abandons the
current offer/answer transaction, sets the signaling state to
"stable", and sets the pending local and/or remote description (see
Sections 4.1.14 and 4.1.16) to "null". Any resources or candidates
that were allocated by the abandoned local description are discarded;
any media that is received is processed according to the previous
local and remote descriptions.
A rollback disassociates any RtpTransceivers that were associated
with "m=" sections by the application of the rolled-back session
description (see Sections 5.10 and 5.9). 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, an 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 an "m=" section for the added track appear
in the generated offer.
5.8. Parsing a Session Description
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 that it is syntactically correct
according to its defining ABNF, check that it conforms to the
semantics used in [RFC4566] and [RFC3264], 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.
5.8.1. 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.
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 only
used for ICE mismatch detection, as defined in [RFC8445],
Section 5.4. Note that JSEP implementations should never
encounter this condition because ICE is required for WebRTC.
* The "i=", "u=", "e=", "p=", "t=", "r=", "z=", and "k=" lines MUST
be checked for syntax, but their values are not otherwise 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:
* Any "a=group" lines are parsed as specified in [RFC5888],
Section 5, and the group's semantics and mids are stored.
* If present, a single "a=ice-lite" line is parsed as specified in
[RFC8839], Section 5.3, and a value indicating the presence of
ice-lite is stored.
* If present, a single "a=ice-ufrag" line is parsed as specified in
[RFC8839], Section 5.4, and the ufrag value is stored.
* If present, a single "a=ice-pwd" line is parsed as specified in
[RFC8839], Section 5.4, and the password value is stored.
* If present, a single "a=ice-options" line is parsed as specified
in [RFC8839], Section 5.6, and the set of specified options is
stored.
* Any "a=fingerprint" lines are parsed as specified in [RFC8122],
Section 5, and the set of fingerprint and algorithm values is
stored.
* If present, a single "a=setup" line is parsed as specified in
[RFC4145], Section 4, and the setup value is stored.
* If present, a single "a=tls-id" line is parsed as specified in
[RFC8842], Section 5, and the attribute value is stored.
* Any "a=identity" lines are parsed and the identity values stored
for subsequent verification, as specified in [RFC8827], Section 5.
* Any "a=extmap" lines are parsed as specified in [RFC5285],
Section 5, and their values are stored.
Other attributes that are not relevant to JSEP may also be present,
and implementations SHOULD process any that they recognize. 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.
5.8.2. 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:
* 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.
* 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:
* If present, a single "a=ice-ufrag" line is parsed as specified in
[RFC8839], Section 5.4, and the ufrag value is stored.
* If present, a single "a=ice-pwd" line is parsed as specified in
[RFC8839], Section 5.4, and the password value is stored.
* If present, a single "a=ice-options" line is parsed as specified
in [RFC8839], Section 5.6, and the set of specified options is
stored.
* Any "a=candidate" attributes MUST be parsed as specified in
[RFC8839], Section 5.1, and their values stored.
* Any "a=remote-candidates" attributes MUST be parsed as specified
in [RFC8839], Section 5.2, but their values are ignored.
* If present, a single "a=end-of-candidates" attribute MUST be
parsed as specified in [RFC8840], Section 8.1, and its presence or
absence flagged and stored.
* Any "a=fingerprint" lines are parsed as specified in [RFC8122],
Section 5, and the set of fingerprint and algorithm values is
stored.
If the "m=" <proto> value indicates use of RTP, as described in
Section 5.1.2 above, the following attribute lines MUST be processed:
* The "m=" <fmt> value MUST be parsed as specified in [RFC4566],
Section 5.14, and the individual values stored.
* Any "a=rtpmap" or "a=fmtp" lines MUST be parsed as specified in
[RFC4566], Section 6, and their values stored.
* If present, a single "a=ptime" line MUST be parsed as described in
[RFC4566], Section 6, and its value stored.
* If present, a single "a=maxptime" line MUST be parsed as described
in [RFC4566], Section 6, and its value stored.
* If present, a single direction attribute line (e.g., "a=sendrecv")
MUST be parsed as described in [RFC4566], Section 6, and its value
stored.
* Any "a=ssrc" attributes MUST be parsed as specified in [RFC5576],
Section 4.1, and their values stored.
* Any "a=extmap" attributes MUST be parsed as specified in
[RFC5285], Section 5, and their values stored.
* Any "a=rtcp-fb" attributes MUST be parsed as specified in
[RFC4585], Section 4.2, and their values stored.
* 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.
* If present, a single "a=rtcp-mux-only" attribute MUST be parsed as
specified in [RFC8858], Section 3, and its presence or absence
flagged and stored.
* 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.
* 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.
* If present, "a=msid" attributes MUST be parsed as specified in
[RFC8830], Section 3.2, and their values stored, ignoring any
"appdata" field. If no "a=msid" attributes are present, a random
msid-id value is generated for a "default" MediaStream for the
session, if not already present, and this value is stored.
* Any "a=imageattr" attributes MUST be parsed as specified in
[RFC6236], Section 3, and their values stored.
* Any "a=rid" lines MUST be parsed as specified in [RFC8851],
Section 10, and their values stored.
* If present, a single "a=simulcast" line MUST be parsed as
specified in [RFC8853], and its values stored.
Otherwise, if the "m=" <proto> value indicates use of SCTP, the
following attribute lines MUST be processed:
* The "m=" <fmt> value MUST be parsed as specified in [RFC8841],
Section 4.3, and the application protocol value stored.
* An "a=sctp-port" attribute MUST be present, and it MUST be parsed
as specified in [RFC8841], Section 5.2, and the value stored.
* If present, a single "a=max-message-size" attribute MUST be parsed
as specified in [RFC8841], Section 6, and the value stored.
Otherwise, use the specified default.
Other attributes that are not relevant to JSEP may also be present,
and implementations SHOULD process any that they recognize. As
required by [RFC4566], Section 5.13, unknown attribute lines MUST be
ignored.
5.8.3. Semantics Verification
Assuming that 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:
* For each "m=" section, valid values for each of the mandatory-to-
use features enumerated in Section 5.1.1 MUST be present. These
values MAY be either 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 [RFC8839], Section 5.4.
- A tls-id value, which MUST be set according to [RFC8842],
Section 5. If this is a re-offer or a response to a re-offer
and the tls-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.
- A 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.
* All rid-ids referenced in an "a=simulcast" line MUST exist as
"a=rid" lines.
* Each "m=" section is also checked to ensure that prohibited
features are not used.
* 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, that section MUST
also contain an "a=rtcp-mux" attribute.
* If an "m=" section was present in the previous answer, the state
of RTP/RTCP multiplexing MUST match what was previously
negotiated.
If this session description is of type "pranswer" or "answer", the
following additional checks are applied:
* 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.
* 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.
5.9. 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.
First, "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.
* If this "m=" section is new, begin gathering candidates for it, as
defined in [RFC8445], Section 5.1.1, unless it is definitively
being bundled (either (1) this is an offer and the "m=" section is
marked bundle-only or (2) it is an answer and the "m=" section is
bundled into another "m=" section).
* Or, if the ICE ufrag and password values have changed, trigger the
ICE agent to start an ICE restart as described in [RFC8445],
Section 9, and begin gathering new candidates for the "m="
section. If this description is an answer, also start checks on
that media section.
* If the "m=" section <proto> value indicates use of RTP:
- If there is no RtpTransceiver associated with this "m="
section, find one and associate it with this "m=" section
according to the following steps. Note that this situation
will only occur when applying an offer.
o Find the RtpTransceiver that corresponds to this "m="
section, using the mapping between transceivers and "m="
section indices established when creating the offer.
o 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.
- For each specified RTP header extension, establish a mapping
between the extension ID and URI, as described in [RFC5285],
Section 6.
- 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 [RFC8843], Section 15.
- For each specified media format, establish a mapping between
the payload type and the actual media format, as described in
[RFC3264], Section 6.1. In addition, prepare to demux RTP
streams intended for this "m=" section based on the media
formats supported by this "m=" section, as described in
[RFC8843], Section 9.2.
- For each specified "rtx" media format, establish a mapping
between the RTX payload type and its associated primary payload
type, as described in Sections 8.6 and 8.7 of [RFC4588].
- If the direction 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 Section 5.11 below.
5.10. 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
canTrickleIceCandidates 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.
* For any specified "CT" bandwidth value, set this value as the
limit for the maximum total bitrate for all "m=" sections, as
specified in [RFC4566], Section 5.8. 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.
* For any specified "RR" or "RS" bandwidth values, handle as
specified in [RFC3556], Section 2.
* Any "AS" bandwidth value ([RFC4566], Section 5.8) MUST be ignored,
as the meaning of this construct at the session level is not well
defined.
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.
* If the ICE ufrag or password changed from the previous remote
description:
- If the description is of type "offer", the implementation MUST
note that an ICE restart is needed, as described in [RFC8839],
Section 4.4.1.1.1.
- If the description is of type "answer" or "pranswer", then
check to see if the current local description is an ICE
restart, and if not, generate an error. 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 checklist state for the
"m=" section. Finally, signal the ICE agent to begin checks.
* If the current local description indicates an ICE restart but
neither the ICE ufrag nor the password has changed from the
previous remote description (as prescribed by [RFC8445],
Section 9), generate an error.
* Configure the ICE components associated with this media section to
use the supplied ICE remote ufrag and password for their
connectivity checks.
* Pair any supplied ICE candidates with any gathered local
candidates, as described in [RFC8445], Section 6.1.2, and start
connectivity checks with the appropriate credentials.
* If an "a=end-of-candidates" attribute is present, process the end-
of-candidates indication as described in [RFC8838], Section 14.
* If the "m=" section <proto> value indicates use of RTP:
- If the "m=" section is being recycled (see Section 5.2.2),
disassociate 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 disassociated in the previous step),
either find an RtpTransceiver or create one according to the
following steps:
o 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.
o If no RtpTransceiver was found in the previous step, create
one with a recvonly direction.
o 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. Note that RTX payload types may refer to primary
payload types that are not supported by the local media
implementation, in which case the RTX payload type MUST also be
ignored.
- For each specified fmtp parameter that is supported by the
local implementation, enable them on the associated media
formats.
- For each specified Synchronization Source (SSRC) that is
signaled in the "m=" section, prepare to demux RTP streams
intended for this "m=" section using that SSRC, as described in
[RFC8843], Section 9.2.
- 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.
- For each specified RTCP feedback mechanism that is supported by
the local implementation, enable them on the associated media
formats.
- For any specified "TIAS" ("Transport Independent Application
Specific Maximum") 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 1000 changes the unit from kbps to bps (as required by
TIAS), and the 0.95 is to allocate 5% to RTCP. An estimate of
header overhead is then subtracted out, in which the 50 is
based on 50 packets per second, the 40 is based on typical
header size (in bytes), and the 8 converts bytes to bits. Note
that "TIAS" is preferred over "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":
o For each specified "CN" media format, configure silence
suppression for all supported media formats with the same
clock rate, as described in [RFC3389], Section 5, except for
formats that have their own internal silence suppression
mechanisms. Silence suppression for such formats (e.g.,
Opus) is controlled via fmtp parameters, as discussed in
Section 5.2.3.2.
o For each specified "telephone-event" media format, enable
dual-tone multifrequency (DTMF) transmission for all
supported media formats with the same clock rate, as
described in [RFC4733], Section 2.5.1.2. If there are any
supported media formats that do not have a corresponding
telephone-event format, disable DTMF transmission for those
formats.
o For any specified "ptime" value, configure the available
media formats to use the specified packet size when sending.
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 Section 5.11 below.
5.11. 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:
* If the "m=" section has been rejected (i.e., the <port> value 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 [RFC8839], Section 4.4.3.1.
* If the remote DTLS fingerprint has been changed or the value of
the "a=tls-id" attribute 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
the tls-id value or fingerprint, then the same DTLS connection is
continued over the new ICE channel. Note that although JSEP
requires that answerers change the tls-id value if and only if the
offerer does, non-JSEP answerers are permitted to change the tls-
id value as long as the offer contained an ICE restart. Thus,
JSEP implementations that process DTLS data prior to receiving an
answer MUST be prepared to receive either a ClientHello or data
from the previous DTLS connection.
* 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.
* If the "m=" section <proto> value indicates use of RTP:
- If the "m=" section references 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. However, new media formats and new RTP header
extension values are permitted in the answer, as described in
[RFC3264], Section 7 and [RFC5285], Section 6.
- 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 direction attribute in the answer indicates that the
JSEP implementation should be sending media ("sendonly" for
local answers, "recvonly" for remote answers, or "sendrecv" for
either type of answer), choose the media format to send as the
most preferred media format from the remote description that is
also locally supported, as discussed in Sections 6.1 and 7 of
[RFC3264], and start transmitting RTP media using that format
once the underlying transport layers have been established. If
an SSRC has not already been chosen for this outgoing RTP
stream, choose a unique random one. If media is already being
transmitted, the same SSRC SHOULD be used unless the clock rate
of the new codec is different, in which case a new SSRC MUST be
chosen, as specified in [RFC7160], Section 4.1.
- 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 MID header extension has been
negotiated, include it in the outgoing RTP streams, as
indicated in [RFC8843], Section 15. If the RtpStreamId or
RepairedRtpStreamId header extensions have been negotiated and
rid-ids have been established, include these header extensions
in the outgoing RTP streams, as indicated in [RFC8851],
Section 4.
- If the "m=" section is of type "audio", and silence suppression
was (1) configured for the send media format as a result of
processing the remote description and (2) also enabled for that
format in the local description, use silence suppression for
outgoing media, in accordance with the guidance in
Section 5.2.3.2. If these conditions are not met, silence
suppression MUST NOT be used for outgoing media.
- If simulcast has been negotiated, send the appropriate number
of Source RTP Streams as specified in [RFC8853], Section 5.3.3.
- 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 unique 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 clock rate, allow redundant encoding
using the specified format for resiliency purposes, as
discussed in [RFC8854], 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 [RFC8108], Section 5.4.1 to
select which SSRC to use.
- If the direction attribute in the answer indicates that the
JSEP implementation should not be sending media ("recvonly" for
local answers, "sendonly" for remote answers, or "inactive" for
either type of answer), stop transmitting all RTP media, but
continue sending RTCP, as described in [RFC3264], Section 5.1.
* 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 an
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 [RFC8841], 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 [RFC8843], Section 7.4.
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 [RFC8843], Section 9.2. 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 or sections 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 the RID mechanism
[RFC8851] and its associated RtpStreamId and RepairedRtpStreamId
identifiers to distinguish between multiple encoded streams and
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
accommodate RFC line-length restrictions, some of the SDP lines 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.
More examples of SDP for WebRTC call flows, including examples with
IPv6 addresses, can be found in [SDP4WebRTC].
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 the JavaScript application in
Alice's browser to the JavaScript in Bob's browser, abbreviated as
"AliceJS" and "BobJS", respectively, are assumed to flow over some
signaling protocol via a web server. The JavaScript 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. The user agents (JSEP implementations) in Alice's and Bob's
browsers, abbreviated as "AliceUA" and "BobUA", respectively, are
both using the default bundle policy of "balanced" and the default
RTCP mux policy of "require".
// 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
s=-
t=0 0
a=ice-options:trickle ice2
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=fmtp:97 0-15
a=fmtp:98 0-15
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
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=tls-id:91bbf309c0990a6bec11e38ba2933cee
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 100 101 102 103
c=IN IP4 203.0.113.100
a=mid:v1
a=sendrecv
a=rtpmap:100 VP8/90000
a=rtpmap:101 H264/90000
a=fmtp:101 packetization-mode=1;profile-level-id=42e01f
a=rtpmap:102 rtx/90000
a=fmtp:102 apt=100
a=rtpmap:103 rtx/90000
a=fmtp:103 apt=101
a=extmap:1 urn:ietf:params:rtp-hdrext:sdes:mid
a=extmap:3 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id
a=rtcp-fb:100 ccm fir
a=rtcp-fb:100 nack
a=rtcp-fb:100 nack pli
a=msid:47017fee-b6c1-4162-929c-a25110252400
a=ice-ufrag:BGKk
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=tls-id:91bbf309c0990a6bec11e38ba2933cee
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:
v=0
o=- 6729291447651054566 1 IN IP4 0.0.0.0
s=-
t=0 0
a=ice-options:trickle ice2
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=fmtp:97 0-15
a=fmtp:98 0-15
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
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=tls-id:eec3392ab83e11ceb6a0990c903fbb19
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 100 101 102 103
c=IN IP4 203.0.113.200
a=mid:v1
a=sendrecv
a=rtpmap:100 VP8/90000
a=rtpmap:101 H264/90000
a=fmtp:101 packetization-mode=1;profile-level-id=42e01f
a=rtpmap:102 rtx/90000
a=fmtp:102 apt=100
a=rtpmap:103 rtx/90000
a=fmtp:103 apt=101
a=extmap:1 urn:ietf:params:rtp-hdrext:sdes:mid
a=extmap:3 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id
a=rtcp-fb:100 ccm fir
a=rtcp-fb:100 nack
a=rtcp-fb:100 nack pli
a=msid:61317484-2ed4-49d7-9eb7-1414322a7aae
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 screen sharing, both supporting FEC --
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 event 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
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|
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:
v=0
o=- 4962303333179871723 1 IN IP4 0.0.0.0
s=-
t=0 0
a=ice-options:trickle ice2
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=fmtp:97 0-15
a=fmtp:98 0-15
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
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=tls-id:17f0f4ba8a5f1213faca591b58ba52a7
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
|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:
v=0
o=- 7729291447651054566 1 IN IP4 0.0.0.0
s=-
t=0 0
a=ice-options:trickle ice2
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=fmtp:97 0-15
a=fmtp:98 0-15
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
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=tls-id:7a25ab85b195acaf3121f5a8ab4f0f71
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
|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 ice2
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=fmtp:97 0-15
a=fmtp:98 0-15
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
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=tls-id:7a25ab85b195acaf3121f5a8ab4f0f71
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 100 101 102 103 104
c=IN IP4 192.0.2.200
a=mid:v1
a=sendrecv
a=rtpmap:100 VP8/90000
a=rtpmap:101 H264/90000
a=fmtp:101 packetization-mode=1;profile-level-id=42e01f
a=rtpmap:102 rtx/90000
a=fmtp:102 apt=100
a=rtpmap:103 rtx/90000
a=fmtp:103 apt=101
a=rtpmap:104 flexfec/90000
a=extmap:1 urn:ietf:params:rtp-hdrext:sdes:mid
a=extmap:3 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id
a=rtcp-fb:100 ccm fir
a=rtcp-fb:100 nack
a=rtcp-fb:100 nack pli
a=msid:71317484-2ed4-49d7-9eb7-1414322a7aae
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 100 101 102 103 104
c=IN IP4 192.0.2.200
a=mid:v2
a=sendrecv
a=rtpmap:100 VP8/90000
a=rtpmap:101 H264/90000
a=fmtp:101 packetization-mode=1;profile-level-id=42e01f
a=rtpmap:102 rtx/90000
a=fmtp:102 apt=100
a=rtpmap:103 rtx/90000
a=fmtp:103 apt=101
a=rtpmap:104 flexfec/90000
a=extmap:1 urn:ietf:params:rtp-hdrext:sdes:mid
a=extmap:3 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id
a=rtcp-fb:100 ccm fir
a=rtcp-fb:100 nack
a=rtcp-fb:100 nack pli
a=msid:81317484-2ed4-49d7-9eb7-1414322a7aae
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 ice2
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=fmtp:97 0-15
a=fmtp:98 0-15
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
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=tls-id:17f0f4ba8a5f1213faca591b58ba52a7
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 100 101 102 103
c=IN IP4 192.0.2.100
a=mid:v1
a=recvonly
a=rtpmap:100 VP8/90000
a=rtpmap:101 H264/90000
a=fmtp:101 packetization-mode=1;profile-level-id=42e01f
a=rtpmap:102 rtx/90000
a=fmtp:102 apt=100
a=rtpmap:103 rtx/90000
a=fmtp:103 apt=101
a=imageattr:100 recv [x=[48:1920],y=[48:1080],q=1.0]
a=extmap:1 urn:ietf:params:rtp-hdrext:sdes:mid
a=extmap:3 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id
a=rtcp-fb:100 ccm fir
a=rtcp-fb:100 nack
a=rtcp-fb:100 nack pli
m=video 12100 UDP/TLS/RTP/SAVPF 100 101 102 103
c=IN IP4 192.0.2.100
a=mid:v2
a=recvonly
a=rtpmap:100 VP8/90000
a=rtpmap:101 H264/90000
a=fmtp:101 packetization-mode=1;profile-level-id=42e01f
a=rtpmap:102 rtx/90000
a=fmtp:102 apt=100
a=rtpmap:103 rtx/90000
a=fmtp:103 apt=101
a=imageattr:100 recv [x=[48:1920],y=[48:1080],q=1.0]
a=extmap:1 urn:ietf:params:rtp-hdrext:sdes:mid
a=extmap:3 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id
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 4.1.10.1. Here, Alice's endpoint sends an offer to Bob's
endpoint to start an audio/video call. Bob immediately responds with
an answer that accepts the audio/video "m=" sections but marks them
as sendonly (from his perspective), meaning that Alice will not yet
send media. This allows the JSEP implementation to start negotiating
ICE and DTLS immediately. Bob's endpoint then prompts him to answer
the call, and when he does, his endpoint sends a second offer, which
enables the audio and video "m=" sections, and thereby bidirectional
media transmission. The advantage of such a flow is that as soon as
the first answer is received, the implementation can proceed with ICE
and DTLS negotiation and establish the session transport. If the
transport setup completes before the second offer is sent, then media
can be transmitted by the callee immediately upon answering the call,
minimizing perceived post-dial delay. The second offer/answer
exchange can also change the preferred codecs or other session
parameters.
This example also makes use of the "relay" ICE candidate policy
described in Section 3.5.3 to minimize the ICE gathering and checking
needed.
// set up local media state
AliceJS->AliceUA: create new PeerConnection with "relay" ICE policy
AliceJS->AliceUA: addTrack with two tracks: audio and video
AliceJS->AliceUA: createOffer to get |offer-C1|
AliceJS->AliceUA: setLocalDescription with |offer-C1|
// |offer-C1| is sent over signaling protocol to Bob
AliceJS->WebServer: signaling with |offer-C1|
WebServer->BobJS: signaling with |offer-C1|
// |offer-C1| arrives at Bob
BobJS->BobUA: create new PeerConnection with "relay" ICE policy
BobJS->BobUA: setRemoteDescription with |offer-C1|
BobUA->BobJS: ontrack events for audio and video
// a relay candidate is sent to Bob
AliceUA->AliceJS: onicecandidate (relay) |offer-C1-candidate-1|
AliceJS->WebServer: signaling with |offer-C1-candidate-1|
WebServer->BobJS: signaling with |offer-C1-candidate-1|
BobJS->BobUA: addIceCandidate with |offer-C1-candidate-1|
// Bob prepares an early answer to warm up the
// transport
BobJS->BobUA: addTransceiver with null audio and video tracks
BobJS->BobUA: transceiver.setDirection(sendonly) for both
BobJS->BobUA: createAnswer
BobJS->BobUA: setLocalDescription with answer
// |answer-C1| is sent over signaling protocol
// to Alice
BobJS->WebServer: signaling with |answer-C1|
WebServer->AliceJS: signaling with |answer-C1|
// |answer-C1| (sendonly) arrives at Alice
AliceJS->AliceUA: setRemoteDescription with |answer-C1|
AliceUA->AliceJS: ontrack events for audio and video
// a relay candidate is sent to Alice
BobUA->BobJS: onicecandidate (relay) |answer-B1-candidate-1|
BobJS->WebServer: signaling with |answer-B1-candidate-1|
WebServer->AliceJS: signaling with |answer-B1-candidate-1|
AliceJS->AliceUA: addIceCandidate with |answer-B1-candidate-1|
// ICE and DTLS establish while call is ringing
// Bob accepts call, starts media, and sends
// new offer
BobJS->BobUA: transceiver.setTrack with audio and video tracks
BobUA->AliceUA: media sent from Bob to Alice
BobJS->BobUA: transceiver.setDirection(sendrecv) for both
transceivers
BobJS->BobUA: createOffer
BobJS->BobUA: setLocalDescription with offer
// |offer-C2| is sent over signaling protocol
// to Alice
BobJS->WebServer: signaling with |offer-C2|
WebServer->AliceJS: signaling with |offer-C2|
// |offer-C2| (sendrecv) arrives at Alice
AliceJS->AliceUA: setRemoteDescription with |offer-C2|
AliceJS->AliceUA: createAnswer
AliceJS->AliceUA: setLocalDescription with |answer-C2|
AliceUA->BobUA: media sent from Alice to Bob
// |answer-C2| is sent over signaling protocol
// to Bob
AliceJS->WebServer: signaling with |answer-C2|
WebServer->BobJS: signaling with |answer-C2|
BobJS->BobUA: setRemoteDescription with |answer-C2|
The SDP for |offer-C1| looks like:
v=0
o=- 1070771854436052752 1 IN IP4 0.0.0.0
s=-
t=0 0
a=ice-options:trickle ice2
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=fmtp:97 0-15
a=fmtp:98 0-15
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
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=tls-id:9e5b948ade9c3d41de6617b68f769e55
a=rtcp-mux
a=rtcp-mux-only
a=rtcp-rsize
m=video 0 UDP/TLS/RTP/SAVPF 100 101 102 103
c=IN IP4 0.0.0.0
a=mid:v1
a=sendrecv
a=rtpmap:100 VP8/90000
a=rtpmap:101 H264/90000
a=fmtp:101 packetization-mode=1;profile-level-id=42e01f
a=rtpmap:102 rtx/90000
a=fmtp:102 apt=100
a=rtpmap:103 rtx/90000
a=fmtp:103 apt=101
a=extmap:1 urn:ietf:params:rtp-hdrext:sdes:mid
a=extmap:3 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id
a=rtcp-fb:100 ccm fir
a=rtcp-fb:100 nack
a=rtcp-fb:100 nack pli
a=msid:bbce3ba6-abfc-ac63-d00a-e15b286f8fce
a=bundle-only
|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:
v=0
o=- 6386516489780559513 1 IN IP4 0.0.0.0
s=-
t=0 0
a=ice-options:trickle ice2
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=fmtp:97 0-15
a=fmtp:98 0-15
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
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=tls-id:55e967f86b7166ed14d3c9eda849b5e9
a=rtcp-mux
a=rtcp-mux-only
a=rtcp-rsize
m=video 9 UDP/TLS/RTP/SAVPF 100 101 102 103
c=IN IP4 0.0.0.0
a=mid:v1
a=sendonly
a=rtpmap:100 VP8/90000
a=rtpmap:101 H264/90000
a=fmtp:101 packetization-mode=1;profile-level-id=42e01f
a=rtpmap:102 rtx/90000
a=fmtp:102 apt=100
a=rtpmap:103 rtx/90000
a=fmtp:103 apt=101
a=extmap:1 urn:ietf:params:rtp-hdrext:sdes:mid
a=extmap:3 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id
a=rtcp-fb:100 ccm fir
a=rtcp-fb:100 nack
a=rtcp-fb:100 nack pli
a=msid:751f239e-4ae0-c549-aa3d-890de772998b
|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 ice2
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=fmtp:97 0-15
a=fmtp:98 0-15
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
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=tls-id:55e967f86b7166ed14d3c9eda849b5e9
a=rtcp-mux
a=rtcp-mux-only
a=rtcp-rsize
a=candidate:1 1 udp 255 192.0.2.200 12200 typ relay
raddr 0.0.0.0 rport 0
a=end-of-candidates
m=video 12200 UDP/TLS/RTP/SAVPF 100 101 102 103
c=IN IP4 192.0.2.200
a=mid:v1
a=sendrecv
a=rtpmap:100 VP8/90000
a=rtpmap:101 H264/90000
a=fmtp:101 packetization-mode=1;profile-level-id=42e01f
a=rtpmap:102 rtx/90000
a=fmtp:102 apt=100
a=rtpmap:103 rtx/90000
a=fmtp:103 apt=101
a=extmap:1 urn:ietf:params:rtp-hdrext:sdes:mid
a=extmap:3 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id
a=rtcp-fb:100 ccm fir
a=rtcp-fb:100 nack
a=rtcp-fb:100 nack pli
a=msid:751f239e-4ae0-c549-aa3d-890de772998b
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 ice2
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=fmtp:97 0-15
a=fmtp:98 0-15
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
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=tls-id:9e5b948ade9c3d41de6617b68f769e55
a=rtcp-mux
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 100 101 102 103
c=IN IP4 192.0.2.100
a=mid:v1
a=sendrecv
a=rtpmap:100 VP8/90000
a=rtpmap:101 H264/90000
a=fmtp:101 packetization-mode=1;profile-level-id=42e01f
a=rtpmap:102 rtx/90000
a=fmtp:102 apt=100
a=rtpmap:103 rtx/90000
a=fmtp:103 apt=101
a=extmap:1 urn:ietf:params:rtp-hdrext:sdes:mid
a=extmap:3 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id
a=rtcp-fb:100 ccm fir
a=rtcp-fb:100 nack
a=rtcp-fb:100 nack pli
a=msid:bbce3ba6-abfc-ac63-d00a-e15b286f8fce
8. Security Considerations
The IETF has published separate documents [RFC8827] [RFC8826]
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 as an Internet protocol, with the application JavaScript being
untrustworthy from the perspective of the JSEP implementation. Thus,
the threat model of [RFC3552] applies. In particular, JavaScript can
call the API in any order and with any inputs, including malicious
ones. This is particularly relevant when we consider the SDP that is
passed to setLocalDescription. While correct API usage requires that
the application pass in SDP that was derived from createOffer or
createAnswer, there is no guarantee that applications do so. The
JSEP implementation MUST be prepared for the JavaScript to pass in
bogus data instead.
Conversely, the application programmer needs to be aware that the
JavaScript 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. Applications that wish to conceal their public IP
address MUST instead configure the ICE agent to use only relay
candidates.
9. IANA Considerations
This document has no IANA actions.
10. References
10.1. Normative References
[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
Requirement Levels", BCP 14, RFC 2119,
DOI 10.17487/RFC2119, March 1997,
<https://www.rfc-editor.org/info/rfc2119>.
[RFC3261] Rosenberg, J., Schulzrinne, H., Camarillo, G., Johnston,
A., Peterson, J., Sparks, R., Handley, M., and E.
Schooler, "SIP: Session Initiation Protocol", RFC 3261,
DOI 10.17487/RFC3261, June 2002,
<https://www.rfc-editor.org/info/rfc3261>.
[RFC3264] Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model
with Session Description Protocol (SDP)", RFC 3264,
DOI 10.17487/RFC3264, June 2002,
<https://www.rfc-editor.org/info/rfc3264>.
[RFC3552] Rescorla, E. and B. Korver, "Guidelines for Writing RFC
Text on Security Considerations", BCP 72, RFC 3552,
DOI 10.17487/RFC3552, July 2003,
<https://www.rfc-editor.org/info/rfc3552>.
[RFC3605] Huitema, C., "Real Time Control Protocol (RTCP) attribute
in Session Description Protocol (SDP)", RFC 3605,
DOI 10.17487/RFC3605, October 2003,
<https://www.rfc-editor.org/info/rfc3605>.
[RFC3711] Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K.
Norrman, "The Secure Real-time Transport Protocol (SRTP)",
RFC 3711, DOI 10.17487/RFC3711, March 2004,
<https://www.rfc-editor.org/info/rfc3711>.
[RFC3890] Westerlund, M., "A Transport Independent Bandwidth
Modifier for the Session Description Protocol (SDP)",
RFC 3890, DOI 10.17487/RFC3890, September 2004,
<https://www.rfc-editor.org/info/rfc3890>.
[RFC4145] Yon, D. and G. Camarillo, "TCP-Based Media Transport in
the Session Description Protocol (SDP)", RFC 4145,
DOI 10.17487/RFC4145, September 2005,
<https://www.rfc-editor.org/info/rfc4145>.
[RFC4566] Handley, M., Jacobson, V., and C. Perkins, "SDP: Session
Description Protocol", RFC 4566, DOI 10.17487/RFC4566,
July 2006, <https://www.rfc-editor.org/info/rfc4566>.
[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,
DOI 10.17487/RFC4585, July 2006,
<https://www.rfc-editor.org/info/rfc4585>.
[RFC5124] Ott, J. and E. Carrara, "Extended Secure RTP Profile for
Real-time Transport Control Protocol (RTCP)-Based Feedback
(RTP/SAVPF)", RFC 5124, DOI 10.17487/RFC5124, February
2008, <https://www.rfc-editor.org/info/rfc5124>.
[RFC5285] Singer, D. and H. Desineni, "A General Mechanism for RTP
Header Extensions", RFC 5285, DOI 10.17487/RFC5285, July
2008, <https://www.rfc-editor.org/info/rfc5285>.
[RFC5761] Perkins, C. and M. Westerlund, "Multiplexing RTP Data and
Control Packets on a Single Port", RFC 5761,
DOI 10.17487/RFC5761, April 2010,
<https://www.rfc-editor.org/info/rfc5761>.
[RFC5888] Camarillo, G. and H. Schulzrinne, "The Session Description
Protocol (SDP) Grouping Framework", RFC 5888,
DOI 10.17487/RFC5888, June 2010,
<https://www.rfc-editor.org/info/rfc5888>.
[RFC6236] Johansson, I. and K. Jung, "Negotiation of Generic Image
Attributes in the Session Description Protocol (SDP)",
RFC 6236, DOI 10.17487/RFC6236, May 2011,
<https://www.rfc-editor.org/info/rfc6236>.
[RFC6347] Rescorla, E. and N. Modadugu, "Datagram Transport Layer
Security Version 1.2", RFC 6347, DOI 10.17487/RFC6347,
January 2012, <https://www.rfc-editor.org/info/rfc6347>.
[RFC6716] Valin, JM., Vos, K., and T. Terriberry, "Definition of the
Opus Audio Codec", RFC 6716, DOI 10.17487/RFC6716,
September 2012, <https://www.rfc-editor.org/info/rfc6716>.
[RFC6904] Lennox, J., "Encryption of Header Extensions in the Secure
Real-time Transport Protocol (SRTP)", RFC 6904,
DOI 10.17487/RFC6904, April 2013,
<https://www.rfc-editor.org/info/rfc6904>.
[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,
<https://www.rfc-editor.org/info/rfc7160>.
[RFC7587] Spittka, J., Vos, K., and JM. Valin, "RTP Payload Format
for the Opus Speech and Audio Codec", RFC 7587,
DOI 10.17487/RFC7587, June 2015,
<https://www.rfc-editor.org/info/rfc7587>.
[RFC7742] Roach, A.B., "WebRTC Video Processing and Codec
Requirements", RFC 7742, DOI 10.17487/RFC7742, March 2016,
<https://www.rfc-editor.org/info/rfc7742>.
[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,
<https://www.rfc-editor.org/info/rfc7850>.
[RFC7874] Valin, JM. and C. Bran, "WebRTC Audio Codec and Processing
Requirements", RFC 7874, DOI 10.17487/RFC7874, May 2016,
<https://www.rfc-editor.org/info/rfc7874>.
[RFC8108] Lennox, J., Westerlund, M., Wu, Q., and C. Perkins,
"Sending Multiple RTP Streams in a Single RTP Session",
RFC 8108, DOI 10.17487/RFC8108, March 2017,
<https://www.rfc-editor.org/info/rfc8108>.
[RFC8122] Lennox, J. and C. Holmberg, "Connection-Oriented Media
Transport over the Transport Layer Security (TLS) Protocol
in the Session Description Protocol (SDP)", RFC 8122,
DOI 10.17487/RFC8122, March 2017,
<https://www.rfc-editor.org/info/rfc8122>.
[RFC8174] Leiba, B., "Ambiguity of Uppercase vs Lowercase in RFC
2119 Key Words", BCP 14, RFC 8174, DOI 10.17487/RFC8174,
May 2017, <https://www.rfc-editor.org/info/rfc8174>.
[RFC8445] Keranen, A., Holmberg, C., and J. Rosenberg, "Interactive
Connectivity Establishment (ICE): A Protocol for Network
Address Translator (NAT) Traversal", RFC 8445,
DOI 10.17487/RFC8445, July 2018,
<https://www.rfc-editor.org/info/rfc8445>.
[RFC8826] Rescorla, E., "Security Considerations for WebRTC",
RFC 8826, DOI 10.17487/RFC8826, January 2021,
<https://www.rfc-editor.org/info/rfc8826>.
[RFC8827] Rescorla, E., "WebRTC Security Architecture", RFC 8827,
DOI 10.17487/RFC8827, January 2021,
<https://www.rfc-editor.org/info/rfc8827>.
[RFC8830] Alvestrand, H., "WebRTC MediaStream Identification in the
Session Description Protocol", RFC 8830,
DOI 10.17487/RFC8830, January 2021,
<https://www.rfc-editor.org/info/rfc8830>.
[RFC8834] Perkins, C., Westerlund, M., and J. Ott, "Media Transport
and Use of RTP in WebRTC", RFC 8834, DOI 10.17487/RFC8834,
January 2021, <https://www.rfc-editor.org/info/rfc8834>.
[RFC8838] Ivov, E., Uberti, J., and P. Saint-Andre, "Trickle ICE:
Incremental Provisioning of Candidates for the Interactive
Connectivity Establishment (ICE) Protocol", RFC 8838,
DOI 10.17487/RFC8838, January 2021,
<https://www.rfc-editor.org/info/rfc8838>.
[RFC8839] Petit-Huguenin, M., Nandakumar, S., Holmberg, C., Keränen,
A., and R. Shpount, "Session Description Protocol (SDP)
Offer/Answer Procedures for Interactive Connectivity
Establishment (ICE)", RFC 8839, DOI 10.17487/RFC8839,
January 2021, <https://www.rfc-editor.org/info/rfc8839>.
[RFC8840] Ivov, E., Stach, T., Marocco, E., and C. Holmberg, "A
Session Initiation Protocol (SIP) Usage for Incremental
Provisioning of Candidates for the Interactive
Connectivity Establishment (Trickle ICE)", RFC 8840,
DOI 10.17487/RFC8840, January 2021,
<https://www.rfc-editor.org/info/rfc8840>.
[RFC8841] Holmberg, C., Shpount, R., Loreto, S., and G. Camarillo,
"Session Description Protocol (SDP) Offer/Answer
Procedures for Stream Control Transmission Protocol (SCTP)
over Datagram Transport Layer Security (DTLS) Transport",
RFC 8841, DOI 10.17487/RFC8841, January 2021,
<https://www.rfc-editor.org/info/rfc8841>.
[RFC8842] Holmberg, C. and R. Shpount, "Session Description Protocol
(SDP) Offer/Answer Considerations for Datagram Transport
Layer Security (DTLS) and Transport Layer Security (TLS)",
RFC 8842, DOI 10.17487/RFC8842, January 2021,
<https://www.rfc-editor.org/info/rfc8842>.
[RFC8843] Holmberg, C., Alvestrand, H., and C. Jennings,
"Negotiating Media Multiplexing Using the Session
Description Protocol (SDP)", RFC 8843,
DOI 10.17487/RFC8843, January 2021,
<https://www.rfc-editor.org/info/rfc8843>.
[RFC8851] Roach, A.B., Ed., "RTP Payload Format Restrictions",
RFC 8851, DOI 10.17487/RFC8851, January 2021,
<https://www.rfc-editor.org/info/rfc8851>.
[RFC8852] Roach, A.B., Nandakumar, S., and P. Thatcher, "RTP Stream
Identifier Source Description (SDES)", RFC 8852,
DOI 10.17487/RFC8852, January 2021,
<https://www.rfc-editor.org/info/rfc8852>.
[RFC8853] Burman, B., Westerlund, M., Nandakumar, S., and M. Zanaty,
"Using Simulcast in Session Description Protocol (SDP) and
RTP Sessions", RFC 8853, DOI 10.17487/RFC8853, January
2021, <https://www.rfc-editor.org/info/rfc8853>.
[RFC8854] Uberti, J., "WebRTC Forward Error Correction
Requirements", RFC 8854, DOI 10.17487/RFC8854, January
2021, <https://www.rfc-editor.org/info/rfc8854>.
[RFC8858] Holmberg, C., "Indicating Exclusive Support of RTP and RTP
Control Protocol (RTCP) Multiplexing Using the Session
Description Protocol (SDP)", RFC 8858,
DOI 10.17487/RFC8858, January 2021,
<https://www.rfc-editor.org/info/rfc8858>.
[RFC8859] Nandakumar, S., "A Framework for Session Description
Protocol (SDP) Attributes When Multiplexing", RFC 8859,
DOI 10.17487/RFC8859, January 2021,
<https://www.rfc-editor.org/info/rfc8859>.
10.2. Informative References
[RFC3389] Zopf, R., "Real-time Transport Protocol (RTP) Payload for
Comfort Noise (CN)", RFC 3389, DOI 10.17487/RFC3389,
September 2002, <https://www.rfc-editor.org/info/rfc3389>.
[RFC3556] Casner, S., "Session Description Protocol (SDP) Bandwidth
Modifiers for RTP Control Protocol (RTCP) Bandwidth",
RFC 3556, DOI 10.17487/RFC3556, July 2003,
<https://www.rfc-editor.org/info/rfc3556>.
[RFC3960] Camarillo, G. and H. Schulzrinne, "Early Media and Ringing
Tone Generation in the Session Initiation Protocol (SIP)",
RFC 3960, DOI 10.17487/RFC3960, December 2004,
<https://www.rfc-editor.org/info/rfc3960>.
[RFC4568] Andreasen, F., Baugher, M., and D. Wing, "Session
Description Protocol (SDP) Security Descriptions for Media
Streams", RFC 4568, DOI 10.17487/RFC4568, July 2006,
<https://www.rfc-editor.org/info/rfc4568>.
[RFC4588] Rey, J., Leon, D., Miyazaki, A., Varsa, V., and R.
Hakenberg, "RTP Retransmission Payload Format", RFC 4588,
DOI 10.17487/RFC4588, July 2006,
<https://www.rfc-editor.org/info/rfc4588>.
[RFC4733] Schulzrinne, H. and T. Taylor, "RTP Payload for DTMF
Digits, Telephony Tones, and Telephony Signals", RFC 4733,
DOI 10.17487/RFC4733, December 2006,
<https://www.rfc-editor.org/info/rfc4733>.
[RFC5245] Rosenberg, J., "Interactive Connectivity Establishment
(ICE): A Protocol for Network Address Translator (NAT)
Traversal for Offer/Answer Protocols", RFC 5245,
DOI 10.17487/RFC5245, April 2010,
<https://www.rfc-editor.org/info/rfc5245>.
[RFC5506] Johansson, I. and M. Westerlund, "Support for Reduced-Size
Real-Time Transport Control Protocol (RTCP): Opportunities
and Consequences", RFC 5506, DOI 10.17487/RFC5506, April
2009, <https://www.rfc-editor.org/info/rfc5506>.
[RFC5576] Lennox, J., Ott, J., and T. Schierl, "Source-Specific
Media Attributes in the Session Description Protocol
(SDP)", RFC 5576, DOI 10.17487/RFC5576, June 2009,
<https://www.rfc-editor.org/info/rfc5576>.
[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, DOI 10.17487/RFC5763, May
2010, <https://www.rfc-editor.org/info/rfc5763>.
[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,
DOI 10.17487/RFC5764, May 2010,
<https://www.rfc-editor.org/info/rfc5764>.
[RFC6120] Saint-Andre, P., "Extensible Messaging and Presence
Protocol (XMPP): Core", RFC 6120, DOI 10.17487/RFC6120,
March 2011, <https://www.rfc-editor.org/info/rfc6120>.
[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,
<https://www.rfc-editor.org/info/rfc6464>.
[RFC8828] Uberti, J. and G. Shieh, "WebRTC IP Address Handling
Requirements", RFC 8828, DOI 10.17487/RFC8828, January
2021, <https://www.rfc-editor.org/info/rfc8828>.
[SDP4WebRTC]
Nandakumar, S. and C. Jennings, "Annotated Example SDP for
WebRTC", Work in Progress, Internet-Draft, draft-ietf-
rtcweb-sdp-14, 17 December 2020,
<https://tools.ietf.org/html/draft-ietf-rtcweb-sdp-14>.
[TS26.114] 3GPP, "3rd Generation Partnership Project; Technical
Specification Group Services and System Aspects; IP
Multimedia Subsystem (IMS); Multimedia Telephony; Media
handling and interaction (Release 16)", 3GPP TS 26.114
V16.3.0, September 2019,
<https://www.3gpp.org/DynaReport/26114.htm>.
[W3C.webrtc]
Jennings, C., Ed., Boström, H., Ed., and J. Bruaroey, Ed.,
"WebRTC 1.0: Real-time Communication Between Browsers",
World Wide Web Consortium PR PR-webrtc-20201215, December
2020, <https://www.w3.org/TR/2020/PR-webrtc-20201215/>.
Appendix A. SDP ABNF Syntax
For the syntax validation performed in Section 5.8, the following
list of ABNF definitions is used:
+=========================+==========================+
| Attribute | Reference |
+=========================+==========================+
| ptime | Section 6 of [RFC4566] |
+-------------------------+--------------------------+
| maxptime | Section 6 of [RFC4566] |
+-------------------------+--------------------------+
| rtpmap | Section 6 of [RFC4566] |
+-------------------------+--------------------------+
| recvonly | Section 9 of [RFC4566] |
+-------------------------+--------------------------+
| sendrecv | Section 9 of [RFC4566] |
+-------------------------+--------------------------+
| sendonly | Section 9 of [RFC4566] |
+-------------------------+--------------------------+
| inactive | Section 9 of [RFC4566] |
+-------------------------+--------------------------+
| fmtp | Section 9 of [RFC4566] |
+-------------------------+--------------------------+
| rtcp | Section 2.1 of [RFC3605] |
+-------------------------+--------------------------+
| setup | Section 4 of [RFC4145] |
+-------------------------+--------------------------+
| fingerprint | Section 5 of [RFC8122] |
+-------------------------+--------------------------+
| rtcp-fb | Section 4.2 of [RFC4585] |
+-------------------------+--------------------------+
| extmap | Section 7 of [RFC5285] |
+-------------------------+--------------------------+
| mid | Section 4 of [RFC5888] |
+-------------------------+--------------------------+
| group | Section 5 of [RFC5888] |
+-------------------------+--------------------------+
| imageattr | Section 3.1 of [RFC6236] |
+-------------------------+--------------------------+
| extmap (encrypt option) | Section 4 of [RFC6904] |
+-------------------------+--------------------------+
| candidate | Section 5.1 of [RFC8839] |
+-------------------------+--------------------------+
| remote-candidates | Section 5.2 of [RFC8839] |
+-------------------------+--------------------------+
| ice-lite | Section 5.3 of [RFC8839] |
+-------------------------+--------------------------+
| ice-ufrag | Section 5.4 of [RFC8839] |
+-------------------------+--------------------------+
| ice-pwd | Section 5.4 of [RFC8839] |
+-------------------------+--------------------------+
| ice-options | Section 5.6 of [RFC8839] |
+-------------------------+--------------------------+
| msid | Section 3 of [RFC8830] |
+-------------------------+--------------------------+
| rid | Section 10 of [RFC8851] |
+-------------------------+--------------------------+
| simulcast | Section 5.1 of [RFC8853] |
+-------------------------+--------------------------+
| tls-id | Section 4 of [RFC8842] |
+-------------------------+--------------------------+
Table 1: SDP ABNF References
Acknowledgements
Harald Alvestrand, Taylor Brandstetter, Suhas Nandakumar, and Peter
Thatcher provided significant text for this document. Bernard Aboba,
Adam Bergkvist, Jan-Ivar Bruaroey, Dan Burnett, Ben Campbell, Alissa
Cooper, Richard Ejzak, Stefan Håkansson, Ted Hardie, Christer
Holmberg, Andrew Hutton, Randell Jesup, Matthew Kaufman, Anant
Narayanan, Adam Roach, Robert Sparks, Neil Stratford, Martin Thomson,
Sean Turner, and Magnus Westerlund all provided valuable feedback on
this document.
Authors' Addresses
Justin Uberti
Google
747 6th Street South
Kirkland, WA 98033
United States of America
Email: justin@uberti.name
Cullen Jennings
Cisco
400 3rd Avenue SW
Calgary AB T2P 4H2
Canada
Email: fluffy@iii.ca
Eric Rescorla (editor)
Mozilla
Email: ekr@rtfm.com