Javascript Session Establishment Protocol
draft-ietf-rtcweb-jsep-07
The information below is for an old version of the document.
| Document | Type | Active Internet-Draft (rtcweb WG) | |
|---|---|---|---|
| Authors | Justin Uberti , Cullen Jennings , Eric Rescorla | ||
| Last updated | 2014-07-04 | ||
| Replaces | draft-uberti-rtcweb-jsep | ||
| Stream | Internet Engineering Task Force (IETF) | ||
| Formats | plain text xml htmlized pdfized bibtex | ||
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| Stream | WG state | WG Document | |
| Document shepherd | Ted Hardie | ||
| IESG | IESG state | I-D Exists | |
| Consensus boilerplate | Unknown | ||
| Telechat date | (None) | ||
| Responsible AD | (None) | ||
| Send notices to | (None) |
draft-ietf-rtcweb-jsep-07
Network Working Group J. Uberti
Internet-Draft Google
Intended status: Standards Track C. Jennings
Expires: January 5, 2015 Cisco
E. Rescorla, Ed.
Mozilla
July 4, 2014
Javascript Session Establishment Protocol
draft-ietf-rtcweb-jsep-07
Abstract
This document describes the mechanisms for allowing a Javascript
application to control the signaling plane of a multimedia session
via the interface specified in the W3C RTCPeerConnection API, and
discusses how this relates to existing signaling protocols.
Status of this Memo
This Internet-Draft is submitted in full conformance with the
provisions of BCP 78 and BCP 79.
Internet-Drafts are working documents of the Internet Engineering
Task Force (IETF). Note that other groups may also distribute
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Internet-Drafts are draft documents valid for a maximum of six months
and may be updated, replaced, or obsoleted by other documents at any
time. It is inappropriate to use Internet-Drafts as reference
material or to cite them other than as "work in progress."
This Internet-Draft will expire on January 5, 2015.
Copyright Notice
Copyright (c) 2014 IETF Trust and the persons identified as the
document authors. All rights reserved.
This document is subject to BCP 78 and the IETF Trust's Legal
Provisions Relating to IETF Documents
(http://trustee.ietf.org/license-info) in effect on the date of
publication of this document. Please review these documents
carefully, as they describe your rights and restrictions with respect
to this document. Code Components extracted from this document must
include Simplified BSD License text as described in Section 4.e of
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the Trust Legal Provisions and are provided without warranty as
described in the Simplified BSD License.
Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . . 4
1.1. General Design of JSEP . . . . . . . . . . . . . . . . . . 4
1.2. Other Approaches Considered . . . . . . . . . . . . . . . 5
2. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 6
3. Semantics and Syntax . . . . . . . . . . . . . . . . . . . . . 6
3.1. Signaling Model . . . . . . . . . . . . . . . . . . . . . 6
3.2. Session Descriptions and State Machine . . . . . . . . . . 7
3.3. Session Description Format . . . . . . . . . . . . . . . . 9
3.4. ICE . . . . . . . . . . . . . . . . . . . . . . . . . . . 10
3.4.1. ICE Candidate Trickling . . . . . . . . . . . . . . . 10
3.4.1.1. ICE Candidate Format . . . . . . . . . . . . . . . 11
3.4.2. ICE Candidate Pool . . . . . . . . . . . . . . . . . . 11
3.5. Interactions With Forking . . . . . . . . . . . . . . . . 12
3.5.1. Sequential Forking . . . . . . . . . . . . . . . . . . 12
3.5.2. Parallel Forking . . . . . . . . . . . . . . . . . . . 13
4. Interface . . . . . . . . . . . . . . . . . . . . . . . . . . 13
4.1. Methods . . . . . . . . . . . . . . . . . . . . . . . . . 14
4.1.1. Constructor . . . . . . . . . . . . . . . . . . . . . 14
4.1.2. createOffer . . . . . . . . . . . . . . . . . . . . . 14
4.1.3. createAnswer . . . . . . . . . . . . . . . . . . . . . 15
4.1.4. SessionDescriptionType . . . . . . . . . . . . . . . . 16
4.1.4.1. Use of Provisional Answers . . . . . . . . . . . . 17
4.1.4.2. Rollback . . . . . . . . . . . . . . . . . . . . . 18
4.1.5. setLocalDescription . . . . . . . . . . . . . . . . . 18
4.1.6. setRemoteDescription . . . . . . . . . . . . . . . . . 19
4.1.7. localDescription . . . . . . . . . . . . . . . . . . . 19
4.1.8. remoteDescription . . . . . . . . . . . . . . . . . . 20
4.1.9. updateIce . . . . . . . . . . . . . . . . . . . . . . 20
4.1.10. addIceCandidate . . . . . . . . . . . . . . . . . . . 20
5. SDP Interaction Procedures . . . . . . . . . . . . . . . . . . 20
5.1. Requirements Overview . . . . . . . . . . . . . . . . . . 21
5.1.1. Implementation Requirements . . . . . . . . . . . . . 21
5.1.2. Usage Requirements . . . . . . . . . . . . . . . . . . 22
5.2. Constructing an Offer . . . . . . . . . . . . . . . . . . 22
5.2.1. Initial Offers . . . . . . . . . . . . . . . . . . . . 22
5.2.2. Subsequent Offers . . . . . . . . . . . . . . . . . . 26
5.2.3. Options Handling . . . . . . . . . . . . . . . . . . . 28
5.2.3.1. OfferToReceiveAudio . . . . . . . . . . . . . . . 28
5.2.3.2. OfferToReceiveVideo . . . . . . . . . . . . . . . 29
5.2.3.3. VoiceActivityDetection . . . . . . . . . . . . . . 29
5.2.3.4. IceRestart . . . . . . . . . . . . . . . . . . . . 29
5.3. Generating an Answer . . . . . . . . . . . . . . . . . . . 29
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5.3.1. Initial Answers . . . . . . . . . . . . . . . . . . . 30
5.3.2. Subsequent Answers . . . . . . . . . . . . . . . . . . 33
5.3.3. Options Handling . . . . . . . . . . . . . . . . . . . 33
5.4. Parsing an Offer . . . . . . . . . . . . . . . . . . . . . 33
5.5. Parsing an Answer . . . . . . . . . . . . . . . . . . . . 33
5.6. Applying a Local Description . . . . . . . . . . . . . . . 33
5.7. Applying a Remote Description . . . . . . . . . . . . . . 33
6. Configurable SDP Parameters . . . . . . . . . . . . . . . . . 33
7. Security Considerations . . . . . . . . . . . . . . . . . . . 34
8. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 35
9. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 35
10. References . . . . . . . . . . . . . . . . . . . . . . . . . . 35
10.1. Normative References . . . . . . . . . . . . . . . . . . . 35
10.2. Informative References . . . . . . . . . . . . . . . . . . 38
Appendix A. JSEP Implementation Examples . . . . . . . . . . . . 39
A.1. Example API Flows . . . . . . . . . . . . . . . . . . . . 39
A.1.1. Call using ROAP . . . . . . . . . . . . . . . . . . . 39
A.1.2. Call using XMPP . . . . . . . . . . . . . . . . . . . 40
A.1.3. Adding video to a call, using XMPP . . . . . . . . . . 42
A.1.4. Simultaneous add of video streams, using XMPP . . . . 42
A.1.5. Call using SIP . . . . . . . . . . . . . . . . . . . . 43
A.1.6. Handling early media (e.g. 1-800-GO FEDEX), using
SIP . . . . . . . . . . . . . . . . . . . . . . . . . 44
A.2. Example Session Descriptions . . . . . . . . . . . . . . . 45
A.2.1. createOffer . . . . . . . . . . . . . . . . . . . . . 45
A.2.2. createAnswer . . . . . . . . . . . . . . . . . . . . . 47
A.2.3. Call Flows . . . . . . . . . . . . . . . . . . . . . . 49
Appendix B. Change log . . . . . . . . . . . . . . . . . . . . . 49
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 50
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1. Introduction
This document describes how the W3C WEBRTC RTCPeerConnection
interface[W3C.WD-webrtc-20140617] is used to control the setup,
management and teardown of a multimedia session.
1.1. General Design of JSEP
The thinking behind WebRTC call setup has been to fully specify and
control the media plane, but to leave the signaling plane up to the
application as much as possible. The rationale is that different
applications may prefer to use different protocols, such as the
existing SIP or Jingle call signaling protocols, or something custom
to the particular application, perhaps for a novel use case. In this
approach, the key information that needs to be exchanged is the
multimedia session description, which specifies the necessary
transport and media configuration information necessary to establish
the media plane.
With these considerations in mind, this document describes the
Javascript Session Establishment Protocol (JSEP) that allows for full
control of the signaling state machine from Javascript. JSEP removes
the browser almost entirely from the core signaling flow, which is
instead handled by the Javascript making use of two interfaces: (1)
passing in local and remote session descriptions and (2) interacting
with the ICE state machine.
In this document, the use of JSEP is described as if it always occurs
between two browsers. Note though in many cases it will actually be
between a browser and some kind of server, such as a gateway or MCU.
This distinction is invisible to the browser; 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 optionally modifies that offer, and then uses it to set
up its local config via the setLocalDescription() API. The offer is
then 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.
When the call is accepted, the callee uses the createAnswer() API to
generate an appropriate answer, applies it using
setLocalDescription(), and sends the answer back to the initiator
over the signaling channel. When the offerer gets that answer, it
installs it using setRemoteDescription(), and initial setup is
complete. This process can be repeated for additional offer/answer
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exchanges.
Regarding ICE [RFC5245], JSEP decouples the ICE state machine from
the overall signaling state machine, as the ICE state machine must
remain in the browser, because only the browser has the necessary
knowledge of candidates and other transport info. Performing this
separation also provides additional flexibility; in protocols that
decouple session descriptions from transport, such as Jingle, the
session description can be sent immediately and the transport
information can be sent when available. In protocols that don't,
such as SIP, the information can be used in the aggregated form.
Sending transport information separately can allow for faster ICE and
DTLS startup, since ICE checks can start as soon as any transport
information is available rather than waiting for all of it.
Through its abstraction of signaling, the JSEP approach does require
the application to be aware of the signaling process. While the
application does not need to understand the contents of session
descriptions to set up a call, the application must call the right
APIs at the right times, convert the session descriptions and ICE
information into the defined messages of its chosen signaling
protocol, and perform the reverse conversion on the messages it
receives from the other side.
One way to mitigate this is to provide a Javascript library that
hides this complexity from the developer; said library would
implement a given signaling protocol along with its state machine and
serialization code, presenting a higher level call-oriented interface
to the application developer. For example, libraries exist to adapt
the JSEP API into an API suitable for a SIP or XMPP. Thus, JSEP
provides greater control for the experienced developer without
forcing any additional complexity on the novice developer.
1.2. Other Approaches Considered
One approach that was considered instead of JSEP was to include a
lightweight signaling protocol. Instead of providing session
descriptions to the API, the API would produce and consume messages
from this protocol. While providing a more high-level API, this put
more control of signaling within the browser, forcing the browser to
have to understand and handle concepts like signaling glare. In
addition, it prevented the application from driving the state machine
to a desired state, as is needed in the page reload case.
A second approach that was considered but not chosen was to decouple
the management of the media control objects from session
descriptions, instead offering APIs that would control each component
directly. This was rejected based on a feeling that requiring
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exposure of this level of complexity to the application programmer
would not be beneficial; it would result in an API where even a
simple example would require a significant amount of code to
orchestrate all the needed interactions, as well as creating a large
API surface that needed to be agreed upon and documented. In
addition, these API points could be called in any order, resulting in
a more complex set of interactions with the media subsystem than the
JSEP approach, which specifies how session descriptions are to be
evaluated and applied.
One variation on JSEP that was considered was to keep the basic
session description-oriented API, but to move the mechanism for
generating offers and answers out of the browser. Instead of
providing createOffer/createAnswer methods within the browser, 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, but still allows
applications to generate their own offers/answers (to a large extent)
if they choose, using the description generated by createOffer as an
indication of the browser's capabilities.
2. Terminology
The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
"SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
document are to be interpreted as described in [RFC2119].
3. Semantics and Syntax
3.1. Signaling Model
JSEP does not specify a particular signaling model or state machine,
other than the generic need to exchange SDP media 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 browser 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
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handling. These issues are left entirely up to the application; the
application has complete control over which offers and answers get
handed to the browser, and when.
+-----------+ +-----------+
| Web App |<--- App-Specific Signaling -->| Web App |
+-----------+ +-----------+
^ ^
| SDP | SDP
V V
+-----------+ +-----------+
| Browser |<----------- Media ------------>| Browser |
+-----------+ +-----------+
Figure 1: JSEP Signaling Model
3.2. Session Descriptions and State Machine
In order to establish the media plane, the user agent needs specific
parameters to indicate what to transmit to the remote side, as well
as how to handle the media that is received. These parameters are
determined by the exchange of session descriptions in offers and
answers, and there are certain details to this process that must be
handled in the JSEP APIs.
Whether a session description applies to the local side or the remote
side affects the meaning of that description. For example, the list
of codecs sent to a remote party indicates what the local side is
willing to receive, which, when intersected with the set of codecs
the remote side supports, specifies what the remote side should send.
However, not all parameters follow this rule; for example, the SRTP
parameters [RFC4568] sent to a remote party indicate what the local
side will use to encrypt, and thereby what the remote party should
expect to receive; the remote party will have to accept these
parameters, with no option to choose a different value. [[OPEN
ISSUE: This is not correct because we removed SDES
(https://github.com/rtcweb-wg/jsep/issues/10)]]
In addition, various RFCs put different conditions on the format of
offers versus answers. For example, a offer may propose multiple
SRTP configurations, but an answer may only contain a single SRTP
configuration. [[OPEN ISSUE: See issue 10 above.]]
Lastly, while the exact media parameters are only known only after an
offer and an answer have been exchanged, it is possible for the
offerer to receive media after they have sent an offer and before
they have received an answer. To properly process incoming media in
this case, the offerer's media handler must be aware of the details
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of the offer before the answer arrives.
Therefore, in order to handle session descriptions properly, the user
agent needs:
1. To know if a session description pertains to the local or remote
side.
2. To know if a session description is an offer or an answer.
3. To allow the offer to be specified independently of the answer.
JSEP addresses this by adding both setLocalDescription and
setRemoteDescription methods and having session description objects
contain a type field indicating the type of session description being
supplied. This satisfies the requirements listed above for both the
offerer, who first calls setLocalDescription(sdp [offer]) and then
later setRemoteDescription(sdp [answer]), as well as for the
answerer, who first calls setRemoteDescription(sdp [offer]) and then
later setLocalDescription(sdp [answer]).
JSEP also allows for an answer to be treated as provisional by the
application. Provisional answers provide a way for an answerer to
communicate initial session parameters back to the offerer, in order
to allow the session to begin, while allowing a final answer to be
specified later. This concept of a final answer is important to the
offer/answer model; when such an answer is received, any extra
resources allocated by the caller can be released, now that the exact
session configuration is known. These "resources" can include things
like extra ICE components, TURN candidates, or video decoders.
Provisional answers, on the other hand, do no such deallocation
results; as a result, multiple dissimilar provisional answers can be
received and applied during call setup.
In [RFC3264], the constraint at the signaling level is that only one
offer can be outstanding for a given session, but at the media stack
level, a new offer can be generated at any point. For example, when
using SIP for signaling, if one offer is sent, then cancelled using a
SIP CANCEL, another offer can be generated even though no answer was
received for the first offer. To support this, the JSEP media layer
can provide an offer via the createOffer() method whenever the
Javascript application needs one for the signaling. The answerer can
send back zero or more provisional answers, and finally end the
offer-answer exchange by sending a final answer. The state machine
for this is as follows:
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setRemote(OFFER) setLocal(PRANSWER)
/-----\ /-----\
| | | |
v | v |
+---------------+ | +---------------+ |
| |----/ | |----/
| | setLocal(PRANSWER) | |
| Remote-Offer |------------------- >| Local-Pranswer|
| | | |
| | | |
+---------------+ +---------------+
^ | |
| | setLocal(ANSWER) |
setRemote(OFFER) | |
| V setLocal(ANSWER) |
+---------------+ |
| | |
| |<---------------------------+
| Stable |
| |<---------------------------+
| | |
+---------------+ setRemote(ANSWER) |
^ | |
| | setLocal(OFFER) |
setRemote(ANSWER) | |
| V |
+---------------+ +---------------+
| | | |
| | setRemote(PRANSWER) | |
| Local-Offer |------------------- >|Remote-Pranswer|
| | | |
| |----\ | |----\
+---------------+ | +---------------+ |
^ | ^ |
| | | |
\-----/ \-----/
setLocal(OFFER) setRemote(PRANSWER)
Figure 2: JSEP State Machine
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
In the WebRTC specification, session descriptions are formatted as
SDP messages. While this format is not optimal for manipulation from
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Javascript, it is widely accepted, and frequently updated with new
features. Any alternate encoding of session descriptions would have
to keep pace with the changes to SDP, at least until the time that
this new encoding eclipsed SDP in popularity. As a result, JSEP
currently uses SDP as the internal representation for its session
descriptions.
However, to simplify Javascript processing, and provide for future
flexibility, the SDP syntax is encapsulated within a
SessionDescription object, which can be constructed from SDP, and be
serialized out to SDP. If future specifications agree on a JSON
format for session descriptions, we could easily enable this object
to generate and consume that JSON.
Other methods may be added to SessionDescription in the future to
simplify handling of SessionDescriptions from Javascript. In the
meantime, Javascript libraries can be used to perform these
manipulations.
Note that most applications should be able to treat the
SessionDescriptions produced and consumed by these various API calls
as opaque blobs; that is, the application will not need to read or
change them. The W3C WebRTC API specification will provide
appropriate APIs to allow the application to control various session
parameters, which will provide the necessary information to the
browser about what sort of SessionDescription to produce.
3.4. ICE
When a new ICE candidate is available, the ICE Agent will notify the
application via a callback; these candidates will automatically be
added to the local session description. When all candidates have
been gathered, the callback will also be invoked to signal that the
gathering process is complete.
3.4.1. ICE Candidate Trickling
Candidate trickling is a technique through which a caller may
incrementally provide candidates to the callee after the initial
offer has been dispatched; the semantics of "Trickle ICE" are defined
in [I-D.ietf-mmusic-trickle-ice]. 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 that
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provide control and feedback on the ICE candidate gathering process.
Applications that support candidate trickling can send the initial
offer immediately and send individual candidates when they get the
notified of a new candidate; applications that do not support this
feature can simply wait for the indication that gathering is
complete, and then create and send their offer, with all the
candidates, at this time.
Upon receipt of trickled candidates, the receiving application will
supply them to its ICE Agent. This triggers the ICE Agent to start
using the new remote candidates for connectivity checks.
3.4.1.1. ICE Candidate Format
As with session descriptions, the syntax of the IceCandidate object
provides some abstraction, but can be easily converted to and from
the SDP candidate lines.
The candidate lines are the only SDP information that is contained
within IceCandidate, as they represent the only information needed
that is not present in the initial offer (i.e., for trickle
candidates). This information is carried with the same syntax as the
"candidate-attribute" field defined for ICE. For example:
candidate:1 1 UDP 1694498815 192.0.2.33 10000 typ host
The IceCandidate object also contains fields to indicate which m=
line it should be associated with. The m line can be identified in
one of two ways; either by a m-line index, or a MID. The m-line
index is a zero-based index, with index N referring to the N+1th
m-line in the SDP sent by the entity which sent the IceCandidate.
The MID uses the "media stream identification", as defined in
[RFC5888], to identify the m-line. WebRTC implementations creating
an ICE Candidate object MUST populate both of these fields.
Implementations receiving an ICE Candidate object SHOULD use the MID
if they implement that functionality, or the m-line index, if not.
3.4.2. ICE Candidate Pool
JSEP applications typically inform the browser to begin ICE gathering
via the information supplied to setLocalDescription, as this is where
the app specifies the number of media streams for which to gather
candidates. However, to accelerate cases where the application knows
the number of media streams to use ahead of time, it MAY ask the
browser to gather a pool of potential ICE candidates to help ensure
rapid media setup. When setLocalDescription is eventually called,
and the browser goes to gather the needed ICE candidates, it SHOULD
start by checking if any candidates are available in the pool. If
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there are candidates in the pool, they SHOULD be handed to the
application immediately via the ICE candidate callback. If the pool
becomes depleted, either because a larger-than-expected number of
media streams is used, or because the pool has not had enough time to
gather candidates, the remaining candidates are gathered as usual.
3.5. Interactions With Forking
Some call signaling systems allow various types of forking where an
SDP Offer may be provided to more than one device. For example, SIP
[RFC3261] defines both a "Parallel Search" and "Sequential Search".
Although these are primarily signaling level issues that are outside
the scope of JSEP, they do have some impact on the configuration of
the media plane that is relevant. When forking happens at the
signaling layer, the Javascript application responsible for the
signaling needs to make the decisions about what media should be sent
or received at any point of time, as well as which remote endpoint it
should communicate with; JSEP is used to make sure the media engine
can make the RTP and media perform as required by the application.
The basic operations that the applications can have the media engine
do are:
o Start exchanging media to a given remote peer, but keep all the
resources reserved in the offer.
o Start exchanging media with a given remote peer, and free any
resources in the offer that are not being used.
3.5.1. Sequential Forking
Sequential forking involves a call being dispatched to multiple
remote callees, where each callee can accept the call, but only one
active session ever exists at a time; no mixing of received media is
performed.
JSEP handles sequential forking well, allowing the application to
easily control the policy for selecting the desired remote endpoint.
When an answer arrives from one of the callees, the application can
choose to apply it either as a provisional answer, leaving open the
possibility of using a different answer in the future, or apply it as
a final answer, ending the setup flow.
In a "first-one-wins" situation, the first answer will be applied as
a final answer, and the application will reject any subsequent
answers. In SIP parlance, this would be ACK + BYE.
In a "last-one-wins" situation, all answers would be applied as
provisional answers, and any previous call leg will be terminated.
At some point, the application will end the setup process, perhaps
with a timer; at this point, the application could reapply the
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existing remote description as a final answer.
3.5.2. Parallel Forking
Parallel forking involves a call being dispatched to multiple remote
callees, where each callee can accept the call, and multiple
simultaneous active signaling sessions can be established as a
result. If multiple callees send media at the same time, the
possibilities for handling this are described in Section 3.1 of
[RFC3960]. Most SIP devices today only support exchanging media with
a single device at a time, and do not try to mix multiple early media
audio sources, as that could result in a confusing situation. For
example, consider having a European ringback tone mixed together with
the North American ringback tone - the resulting sound would not be
like either tone, and would confuse the user. If the signaling
application wishes to only exchange media with one of the remote
endpoints at a time, then from a media engine point of view, this is
exactly like the sequential forking case.
In the parallel forking case where the Javascript application wishes
to simultaneously exchange media with multiple peers, the flow is
slightly more complex, but the Javascript application can follow the
strategy that [RFC3960] describes using UPDATE. The UPDATE approach
allows the signaling to set up a separate media flow for each peer
that it wishes to exchange media with. In JSEP, this offer used in
the UPDATE would be formed by simply creating a new PeerConnection
and making sure that the same local media streams have been added
into this new PeerConnection. Then the new PeerConnection object
would produce a SDP offer that could be used by the signaling to
perform the UPDATE strategy discussed in [RFC3960].
As a result of sharing the media streams, the application will end up
with N parallel PeerConnection sessions, each with a local and remote
description and their own local and remote addresses. The media flow
from these sessions can be managed by specifying SDP direction
attributes in the descriptions, 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.
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4.1. Methods
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. The size
of the ICE candidate pool can also be set, if desired; this indicates
the number of ICE components to pre-gather candidates for. If the
application does not indicate a candidate pool size, the browser may
select any default candidate pool size.
In addition, the application can specify its preferred policy
regarding use of BUNDLE, the multiplexing mechanism defined in
[I-D.ietf-mmusic-sdp-bundle-negotiation]. By specifying a policy
from the list below, the application can control how aggressively it
will try to BUNDLE media streams together. The set of available
policies is as follows:
balanced: The application will BUNDLE all media streams of the same
type together. That is, if there are multiple audio and multiple
video MediaStreamTracks attached to a PeerConnection, all but the
first audio and video tracks will be marked as bundle-only, and
candidates will only be gathered for N media streams, where N is
the number of distinct media types. When talking to a non-BUNDLE-
aware endpoint, only the non-bundle-only streams will be
negotiated. This policy balances desire to multiplex with the
need to ensure basic audio and video still works in legacy cases.
Data channels will be in a separate bundle group.
max-bundle: The application will BUNDLE all of its media streams,
including data channels, on a single transport. 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.
max-compat: The application will offer BUNDLE, but mark none of its
streams as bundle-only. This policy will allow all streams to be
received by non-BUNDLE-aware endpoints, but require separate
candidates to be gathered for each media stream.
4.1.2. createOffer
The createOffer method generates a blob of SDP that contains a
[RFC3264] offer with the supported configurations for the session,
including descriptions of the local MediaStreams attached to this
PeerConnection, the codec/RTP/RTCP options supported by this
implementation, and any candidates that have been gathered by the ICE
Agent. An options parameter may be supplied to provide additional
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control over the generated offer. This options parameter should
allow for the following manipulations to be performed:
o To indicate support for a media type even if no MediaStreamTracks
of that type have been added to the session (e.g., an audio call
that wants to receive video.)
o To trigger an ICE restart, for the purpose of reestablishing
connectivity.
In the initial offer, the generated SDP will contain all desired
functionality for the session (functionality that is supported but
not desired by default may be omitted); for each SDP line, the
generation of the SDP will follow the process defined for generating
an initial offer from the document that specifies the given SDP line.
The exact handling of initial offer generation is detailed in
Section 5.2.1 below.
In the event createOffer is called after the session is established,
createOffer will generate an offer to modify the current session
based on any changes that have been made to the session, e.g. adding
or removing MediaStreams, 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. Because this method may need to inspect the system state
to determine the currently available resources, it may be implemented
as an async operation.
Calling this method may do things such as generate new ICE
credentials, but does not result in candidate gathering, or cause
media to start or stop flowing.
4.1.3. createAnswer
The createAnswer method generates a blob of SDP that contains a
[RFC3264] SDP answer with the supported configuration for the session
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that is compatible with the parameters supplied in the most recent
call to setRemoteDescription, which MUST have been called prior to
calling createAnswer. Like createOffer, the returned blob contains
descriptions of the local MediaStreams attached 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 document that specifies the given
SDP line. The exact handling of answer generation is detailed in
Section 5.3. below.
Session descriptions generated by createAnswer must be immediately
usable by setLocalDescription; like createOffer, the returned
description should reflect the current state of the system. Because
this method may need to inspect the system state to determine the
currently available resources, it may need to be implemented as an
async operation.
Calling this method may do things such as generate new ICE
credentials, but does not trigger candidate gathering or change media
state.
4.1.4. SessionDescriptionType
Session description objects (RTCSessionDescription) may be of type
"offer", "pranswer", and "answer". These types provide information
as to how the description parameter should be parsed, and how the
media state should be changed.
"offer" indicates that a description should be parsed as an offer;
said description may include many possible media configurations. A
description used as an "offer" may be applied anytime the
PeerConnection is in a stable state, or as an update to a previously
supplied but unanswered "offer".
"pranswer" indicates that a description should be parsed as an
answer, but not a final answer, and so should not result in the
freeing of allocated resources. It may result in the start of media
transmission, if the answer does not specify an inactive media
direction. A description used as a "pranswer" may be applied as a
response to an "offer", or an update to a previously sent "pranswer".
"answer" indicates that a description should be parsed as an answer,
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the offer-answer exchange should be considered complete, and any
resources (decoders, candidates) that are no longer needed can be
released. A description used as an "answer" may be applied as a
response to a "offer", or an update to a previously sent "pranswer".
The only difference between a provisional and final answer is that
the final answer results in the freeing of any unused resources that
were allocated as a result of the offer. As such, the application
can use some discretion on whether an answer should be applied as
provisional or final, and can change the type of the session
description as needed. For example, in a serial forking scenario, an
application may receive multiple "final" answers, one from each
remote endpoint. The application could choose to accept the initial
answers as provisional answers, and only apply an answer as final
when it receives one that meets its criteria (e.g. a live user
instead of voicemail).
"rollback" is a special session description type implying that the
state machine should be rolled back to the previous state, as
described in Section 4.1.4.2. The contents MUST be empty.
4.1.4.1. Use of Provisional Answers
Most web 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 web application
would be to create and send an "inactive" final answer immediately
after receiving the offer. Later, when the called user actually
accepts the call, the application can create a new "sendrecv" offer
to update the previous offer/answer pair and start the media flow.
While this could also be done with an inactive "pranswer", followed
by a sendrecv "answer", the initial "pranswer" leaves the offer-
answer exchange open, which means that neither side can send an
updated offer during this time.
As an example, consider a typical web application that will set up a
data channel, an audio channel, and a video channel. When an
endpoint receives an offer with these channels, it could send an
answer accepting the data channel for two-way data, and accepting the
audio and video tracks as inactive or receive-only. It could then
ask the user to accept the call, acquire the local media streams, and
send a new offer to the remote side moving the audio and video to be
two-way media. By the time the human has accepted the call and
triggered the new offer, it is likely that the ICE and DTLS
handshaking for all the channels will already have finished.
Of course, some applications may not be able to perform this double
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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.
4.1.4.2. Rollback
In certain situations it may be desirable to "undo" a change made to
setLocalDescription or setRemoteDescription. Consider a case where a
call is ongoing, and one side wants to change some of the session
parameters; that side generates an updated offer and then calls
setLocalDescription. However, the remote side, either before or
after setRemoteDescription, decides it does not want to accept the
new parameters, and sends a reject message back to the offerer. Now,
the offerer, and possibly the answerer as well, need to return to a
stable state and the previous local/remote description. To support
this, we introduce the concept of "rollback".
A rollback discards any proposed changes to the session, returning
the state machine to the stable state, and setting the modified local
and/or remote description back to their previous values. Any
resources or candidates that were allocated by the abandoned local
description are discarded; any media that is received will be
processed according to the previous local and remote descriptions.
Rollback can only be used to cancel proposed changes; there is no
support for rolling back from a stable state to a previous stable
state. Note that this implies that once the answerer has performed
setLocalDescription with his answer, this cannot be rolled back.
A rollback is performed by supplying a session description of type
"rollback" with empty contents to either setLocalDescription or
setRemoteDescription, depending on which was most recently used (i.e.
if the new offer was supplied to setLocalDescription, the rollback
should be done using setLocalDescription as well).
4.1.5. setLocalDescription
The setLocalDescription method instructs the PeerConnection to apply
the supplied SDP blob as its local configuration. The type field
indicates whether the blob should be processed as an offer,
provisional answer, or final answer; offers and answers are checked
differently, using the various rules that exist for each SDP line.
This API changes the local media state; among other things, it sets
up local resources for receiving and decoding media. In order to
successfully handle scenarios where the application wants to offer to
change from one media format to a different, incompatible format, the
PeerConnection must be able to simultaneously support use of both the
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old and new local descriptions (e.g. support codecs that exist in
both descriptions) until a final answer is received, at which point
the PeerConnection can fully adopt the new local description, or roll
back to the old description if the remote side denied the change.
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 them.
If setRemoteDescription was previous called with an offer, and
setLocalDescription is called with an answer (provisional or final),
and the media directions are compatible, and media are available to
send, this will result in the starting of media transmission.
4.1.6. setRemoteDescription
The setRemoteDescription method instructs the PeerConnection to apply
the supplied SDP blob as the desired remote configuration. As in
setLocalDescription, the type field of the indicates how the blob
should be processed.
This API changes the local media state; among other things, it sets
up local resources for sending and encoding media.
If setRemoteDescription was previously called with an offer, and
setLocalDescription is called with an answer (provisional or final),
and the media directions are compatible, and media are available to
send, this will result in the starting of media transmission.
4.1.7. localDescription
The localDescription method returns a copy of the current local
configuration, i.e. what was most recently passed to
setLocalDescription, plus any local candidates that have been
generated by the ICE Agent.
[[OPEN ISSUE: Do we need to expose accessors for both the current
and proposed local description?
https://github.com/rtcweb-wg/jsep/issues/16]]
A null object will be returned if the local description has not yet
been established, or if the PeerConnection has been closed.
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4.1.8. remoteDescription
The remoteDescription method returns a copy of the current remote
configuration, i.e. what was most recently passed to
setRemoteDescription, plus any remote candidates that have been
supplied via processIceMessage.
[[OPEN ISSUE: Do we need to expose accessors for both the current
and proposed remote description?
https://github.com/rtcweb-wg/jsep/issues/16]]
A null object will be returned if the remote description has not yet
been established, or if the PeerConnection has been closed.
4.1.9. updateIce
The updateIce method allows the configuration of the ICE Agent to be
changed during the session, primarily for changing which types of
local candidates are provided to the application and used for
connectivity checks. A callee may initially configure the ICE Agent
to use only relay candidates, to avoid leaking location information,
but update this configuration to use all candidates once the call is
accepted.
Regardless of the configuration, the gathering process collects all
available candidates, but excluded candidates will not be surfaced in
onicecandidate callback or used for connectivity checks.
This call may result in a change to the state of the ICE Agent, and
may result in a change to media state if it results in connectivity
being established.
4.1.10. addIceCandidate
The addIceCandidate method provides a remote candidate to the ICE
Agent, which, if parsed successfully, will be added to the remote
description according to the rules defined for Trickle ICE.
Connectivity checks will be sent to the new candidate.
This call will result in a change to the state of the ICE Agent, and
may result in a change to media state if it results in connectivity
being established.
5. SDP Interaction Procedures
This section describes the specific procedures to be followed when
creating and parsing SDP objects.
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5.1. Requirements Overview
JSEP implementations must comply with the specifications listed below
that govern the creation and processing of offers and answers.
The first set of specifications is the "mandatory-to-implement" set.
All implementations must support these behaviors, but may not use all
of them if the remote side, which may not be a JSEP endpoint, does
not support them.
The second set of specifications is the "mandatory-to-use" set. The
local JSEP endpoint and any remote endpoint must indicate support for
these specifications in their session descriptions.
5.1.1. Implementation Requirements
This list of mandatory-to-implement specifications is derived from
the requirements outlined in [I-D.ietf-rtcweb-rtp-usage].
R-1 [RFC4566] is the base SDP specification and MUST be
implemented.
R-2 [RFC5764] MUST be supported for signaling the UDP/TLS/RTP/SAVPF
RTP profile.
R-3 [RFC5245] MUST be implemented for signaling the ICE credentials
and candidate lines corresponding to each media stream. The
ICE implementation MUST be a Full implementation, not a Lite
implementation.
R-4 [RFC5763] MUST be implemented to signal DTLS certificate
fingerprints.
R-5 [RFC4568] MUST NOT be implemented to signal SDES SRTP keying
information.
R-6 The [RFC5888] grouping framework MUST be implemented for
signaling grouping information, and MUST be used to identify m=
lines via the a=mid attribute.
R-7 [I-D.ietf-mmusic-msid] MUST be supported, in order to signal
associations between RTP objects and W3C MediaStreams and
MediaStreamTracks in a standard way.
R-8 The bundle mechanism in
[I-D.ietf-mmusic-sdp-bundle-negotiation] MUST be supported to
signal the ability to multiplex RTP streams on a single UDP
port, in order to avoid excessive use of port number resources.
R-9 The SDP attributes of "sendonly", "recvonly", "inactive", and
"sendrecv" from [RFC4566] MUST be implemented to signal
information about media direction.
R-10 [RFC5576] MUST be implemented to signal RTP SSRC values.
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R-11 [RFC4585] MUST be implemented to signal RTCP based feedback.
R-12 [RFC5761] MUST be implemented to signal multiplexing of RTP and
RTCP.
R-13 [RFC5506] MUST be implemented to signal reduced-size RTCP
messages.
R-14 [RFC3556] with bandwidth modifiers MAY be supported for
specifying RTCP bandwidth as a fraction of the media bandwidth,
RTCP fraction allocated to the senders and setting maximum
media bit-rate boundaries.
As required by [RFC4566], Section 5.13, JSEP implementations MUST
ignore unknown attribute (a=) lines.
5.1.2. Usage Requirements
All session descriptions handled by JSEP endpoints, 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.
R-1 Either the UDP/TLS/RTP/SAVP or the UDP/TLS/RTP/SAVPF RTP
profile, as specified in [RFC5764], MUST be used.
R-2 ICE, as specified in [RFC5245], MUST be used. Note that the
remote endpoint may use a Lite implementation; implementations
MUST properly handle remote endpoints which do ICE-Lite.
R-3 DTLS-SRTP, as specified in [RFC5763], MUST be used.
5.2. Constructing an Offer
When createOffer is called, a new SDP description must be created
that includes the functionality specified in
[I-D.ietf-rtcweb-rtp-usage]. The exact details of this process are
explained below.
5.2.1. Initial Offers
When createOffer is called for the first time, the result is known as
the initial offer.
The first step in generating an initial offer is to generate session-
level attributes, as specified in [RFC4566], Section 5.
Specifically:
o The first SDP line MUST be "v=0", as specified in [RFC4566],
Section 5.1
o The second SDP line MUST be an "o=" line, as specified in
[RFC4566], Section 5.2. The value of the <username> field SHOULD
be "-". The value of the <sess-id> field SHOULD be a
cryptographically random number. To ensure uniqueness, this
number SHOULD be at least 64 bits long. The value of the <sess-
version> field SHOULD be zero. The value of the <nettype>
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<addrtype> <unicast-address> tuple SHOULD be set to a non-
meaningful address, such as IN IP4 0.0.0.0, to prevent leaking the
local address in this field. As mentioned in [RFC4566], the
entire o= line needs to be unique, but selecting a random number
for <sess-id> is sufficient to accomplish this.
o The third SDP line MUST be a "s=" line, as specified in [RFC4566],
Section 5.3; to match the "o=" line, a single dash SHOULD be used
as the session name, e.g. "s=-". Note that this differs from the
advice in [RFC4566] which proposes a single space, but as both
"o=" and "s=" are meaningless, having the same meaningless value
seems clearer.
o Session Information ("i="), URI ("u="), Email Address ("e="),
Phone Number ("p="), Bandwidth ("b="), Repeat Times ("r="), and
Time Zones ("z=") lines are not useful in this context and SHOULD
NOT be included.
o Encryption Keys ("k=") lines do not provide sufficient security
and MUST NOT be included.
o A "t=" line MUST be added, as specified in [RFC4566], Section 5.9;
both <start-time> and <stop-time> SHOULD be set to zero, e.g. "t=0
0".
o An "a=msid-semantic:WMS" line MUST be added, as specified in
[I-D.ietf-mmusic-msid], Section 4.
The next step is to generate m= sections, as specified in [RFC4566]
Section 5.14, for each MediaStreamTrack that has been added to the
PeerConnection via the addStream method. (Note that this method
takes a MediaStream, which can contain multiple MediaStreamTracks,
and therefore multiple m= sections can be generated even if addStream
is only called once.) m=sections MUST be sorted first by the order in
which the MediaStreams were added to the PeerConnection, and then by
the alphabetical ordering of the media type for the MediaStreamTrack.
For example, if a MediaStream containing both an audio and a video
MediaStreamTrack is added to a PeerConnection, the resultant m=audio
section will precede the m=video section. If a second MediaStream
containing an audio MediaStreamTrack was added, it would follow the
m=video section.
Each m= section, provided it is not being bundled into another m=
section, MUST generate a unique set of ICE credentials and gather its
own unique set of ICE candidates. Otherwise, it MUST use the same
ICE credentials and candidates as the m= section into which it is
being bundled. Note that this means that for offers, any m= sections
which are not bundle-only MUST have unique ICE credentials and
candidates, since it is possible that the answerer will accept them
without bundling them.
For DTLS, all m= sections MUST use the certificate for the identity
that has been specified for the PeerConnection; as a result, they
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MUST all have the same [RFC4572] fingerprint value, or this value
MUST be a session-level attribute.
Each m= section should be generated as specified in [RFC4566],
Section 5.14. For the m= line itself, the following rules MUST be
followed:
o The port value is set to the port of the default ICE candidate for
this m= section; if this m= section is not being bundled into
another m= section, the port value MUST be unique. If no
candidates have yet been gathered, and a 'null' port value is
being used, as indicated in [I-D.ietf-mmusic-trickle-ice], Section
5.1., this port MUST still be unique.
o To properly indicate use of DTLS, the <proto> field MUST be set to
"UDP/TLS/RTP/SAVPF", as specified in [RFC5764], Section 8.
Each m= section MUST include the following attribute lines:
o An "a=mid" line, as specified in [RFC5888], Section 4.
o An "a=msid" line, as specified in [I-D.ietf-mmusic-msid], Section
2.
o [OPEN ISSUE: Use of AppID]
o An "a=sendrecv" line, as specified in [RFC3264], Section 5.1.
o For each supported codec, "a=rtpmap" and "a=fmtp" lines, as
specified in [RFC4566], Section 6. For audio, the codecs
specified in [I-D.ietf-rtcweb-audio], Section 3, MUST be be
supported.
o For each primary codec where RTP retransmission should be used, a
corresponding "a=rtpmap" line indicating "rtx" with the clock rate
of the primary codec and an "a=fmtp" line that references the
payload type of the primary codec, as specified in [RFC4588],
Section 8.1.
o For each supported FEC mechanism, a corresponding "a=rtpmap" line
indicating the desired FEC codec.
o "a=ice-ufrag" and "a=ice-passwd" lines, as specified in [RFC5245],
Section 15.4.
o An "a=ice-options" line, with the "trickle" option, as specified
in [I-D.ietf-mmusic-trickle-ice], Section 4.
o For each candidate that has been gathered during the most recent
gathering phase, an "a=candidate" line, as specified in [RFC5245],
Section 4.3., paragraph 3.
o For the current default candidate, a "c=" line, as specified in
[RFC5245], Section 4.3., paragraph 6. If no candidates have been
gathered yet, the default candidate should be set to the 'null'
value defined in [I-D.ietf-mmusic-trickle-ice], Section 5.1.
o An "a=fingerprint" line, as specified in [RFC4572], Section 5; the
algorithm used for the fingerprint MUST match that used in the
certificate signature.
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o An "a=setup" line, as specified in [RFC4145], Section 4, and
clarified for use in DTLS-SRTP scenarios in [RFC5763], Section 5.
The role value in the offer MUST be "actpass".
o An "a=rtcp-mux" line, as specified in [RFC5761], Section 5.1.1.
o An "a=rtcp-rsize" line, as specified in [RFC5506], Section 5.
o For each supported RTP header extension, an "a=extmap" line, as
specified in [RFC5285], Section 5. The list of header extensions
that SHOULD/MUST be supported is specified in
[I-D.ietf-rtcweb-rtp-usage], Section 5.2. Any header extensions
that require encryption MUST be specified as indicated in
[RFC6904], Section 4.
o For each supported RTCP feedback mechanism, an "a=rtcp-fb"
mechanism, as specified in [RFC4585], Section 4.2. The list of
RTCP feedback mechanisms that SHOULD/MUST be supported is
specified in [I-D.ietf-rtcweb-rtp-usage], Section 5.1.
o An "a=ssrc" line, as specified in [RFC5576], Section 4.1,
indicating the SSRC to be used for sending media, along with the
mandatory "cname" source attribute, as specified in Section 6.1,
indicating the CNAME for the source. The CNAME must be generated
in accordance with [RFC7022]. [OPEN ISSUE: How are CNAMEs
specified for MSTs? Are they randomly generated for each
MediaStream? If so, can two MediaStreams be synced? See:
https://github.com/rtcweb-wg/jsep/issues/4]
o If RTX is supported for this media type, another "a=ssrc" line
with the RTX SSRC, and an "a=ssrc-group" line, as specified in
[RFC5576], section 4.2, with semantics set to "FID" and including
the primary and RTX SSRCs.
o If FEC is supported for this media type, another "a=ssrc" line
with the FEC SSRC, and an "a=ssrc-group" line, as specified in
[RFC5576], section 4.2, with semantics set to "FEC" and including
the primary and FEC SSRCs.
o [OPEN ISSUE: Handling of a=imageattr]
o If the BUNDLE policy for this PeerConnection is set to "max-
bundle", and this is not the first m= section, or the BUNDLE
policy is set to "default", and this is not the first m= section
for this media type, an "a=bundle-only" line.
Lastly, if a data channel has been created, a m= section MUST be
generated for data. The <media> field MUST be set to "application"
and the <proto> field MUST be set to "DTLS/SCTP", as specified in
[I-D.ietf-mmusic-sctp-sdp], Section 3; the "fmt" value MUST be set to
the SCTP port number, as specified in Section 4.1.
Within the data m= section, the "a=mid", "a=ice-ufrag", "a=ice-
passwd", "a=ice-options", "a=candidate", "a=fingerprint", and
"a=setup" lines MUST be included as mentioned above, along with an
"a=sctpmap" line referencing the SCTP port number and specifying the
application protocol indicated in [I-D.ietf-rtcweb-data-protocol].
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[OPEN ISSUE: the -01 of this document is missing this information.]
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 browser offers all
m= sections as one BUNDLE group. However, whether the m= sections
are bundle-only or not depends on the BUNDLE policy.
Attributes which SDP permits to either be at the session level or the
media level SHOULD generally be at the media level even if they are
identical. This promotes readability, especially if one of a set of
initially identical attributes is subsequently changed.
Attributes other than the ones specified above MAY be included,
except for the following attributes which are specifically
incompatible with the requirements of [I-D.ietf-rtcweb-rtp-usage],
and MUST NOT be included:
o "a=crypto"
o "a=key-mgmt"
o "a=ice-lite"
Note that when BUNDLE is used, any additional attributes that are
added MUST follow the advice in [I-D.ietf-mmusic-sdp-mux-attributes]
on how those attributes interact with BUNDLE.
Note that these requirements are in some cases stricter than those of
SDP. Implementations MUST be prepared to accept compliant SDP even
if it would not conform to the requirements for generating SDP in
this specification.
5.2.2. Subsequent Offers
When createOffer is called a second (or later) time, or is called
after a local description has already been installed, the processing
is somewhat different than for an initial offer.
If the initial offer was not applied using setLocalDescription,
meaning the PeerConnection is still in the "stable" state, the steps
for generating an initial offer should be followed, subject to the
following restriction:
o The fields of the "o=" line MUST stay the same except for the
<session-version> field, which MUST increment if the session
description changes in any way, including the addition of ICE
candidates.
If the initial offer was applied using setLocalDescription, but an
answer from the remote side has not yet been applied, meaning the
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PeerConnection is still in the "local-offer" state, an offer is
generated by following the steps in the "stable" state above, along
with these exceptions:
o The "s=" and "t=" lines MUST stay the same.
o Each "a=mid" line MUST stay the same.
o Each "a=ice-ufrag" and "a=ice-pwd" line MUST stay the same unless
the "IceRestart" option (Section 5.2.3 was specified. Note that
it's not clear why you would actually want to do this, since at
this point ICE has not yet started and is thus unlikely to need a
restart.
o For MediaStreamTracks that are still present, the "a=msid",
"a=ssrc", and "a=ssrc-group" lines MUST stay the same.
o If any MediaStreamTracks have been removed, either through the
removeStream method or by removing them from an added MediaStream,
their m= sections MUST be marked as recvonly by changing the value
of the [RFC3264] directional attribute to "a=recvonly". The
"a=msid", "a=ssrc", and "a=ssrc-group" lines MUST be removed from
the associated m= sections.
o If any MediaStreamTracks have been added, and there exist m=
sections of the appropriate media type with no associated
MediaStreamTracks (i.e. as described in the preceding paragraph),
those m= sections MUST be recycled by adding the new
MediaStreamTrack to the m= section. This is done by adding the
necessary "a=msid", "a=ssrc", and "a=ssrc-group" lines to the
recycled m= section, and removing the "a=recvonly" attribute.
If the initial offer was applied using setLocalDescription, and an
answer from the remote side has been applied using
setRemoteDescription, meaning the PeerConnection is in the "remote-
pranswer" or "stable" states, an offer is generated based on the
negotiated session descriptions by following the steps mentioned for
the "local-offer" state above, along with these exceptions: [OPEN
ISSUE: should this be permitted in the remote-pranswer state?]
o If a m= section exists in the current local description, but does
not have an associated local MediaStreamTrack (possibly because
said MediaStreamTrack was removed since the last exchange), a m=
section MUST still be generated in the new offer, as indicated in
[RFC3264], Section 8. The disposition of this section will depend
on the state of the remote MediaStreamTrack associated with this
m= section. If one exists, and it is still in the "live" state,
the new m= section MUST be marked as "a=recvonly", with no
"a=msid" or related attributes present. If no remote
MediaStreamTrack exists, or it is in the "ended" state, the m=
section MUST be marked as rejected, by setting the port to zero,
as indicated in [RFC3264], Section 8.2.
o If any MediaStreamTracks have been added, and there exist recvonly
m= sections of the appropriate media type with no associated
MediaStreamTracks, or rejected m= sections of any media type,
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those m= sections MUST be recycled, and a local MediaStreamTrack
associated with these recycled m= sections until all such existing
m= sections have been used. This includes any recvonly or
rejected m= sections created by the preceding paragraph.
In addition, for each non-recycled, non-rejected m= section in the
new offer, the following adjustments are made based on the contents
of the corresponding m= section in the current remote description:
o The m= line and corresponding "a=rtpmap" and "a=fmtp" lines MUST
only include codecs present in the remote description.
o The RTP header extensions MUST only include those that are present
in the remote description.
o The RTCP feedback extensions MUST only include those that are
present in the remote description.
o The "a=rtcp-mux" line MUST only be added if present in the remote
description.
o The "a=rtcp-rsize" line MUST only be added if present in the
remote description.
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.
5.2.3. Options Handling
The createOffer method takes as a parameter an RTCOfferOptions
object. Special processing is performed when generating a SDP
description if the following constraints are present.
5.2.3.1. OfferToReceiveAudio
If the "OfferToReceiveAudio" option is specified, with an integer
value of N, the offer MUST include N non-rejected m= sections with
media type "audio", even if fewer than N audio MediaStreamTracks have
been added to the PeerConnection. This allows the offerer to receive
audio, including multiple independent streams, even when not sending
it; accordingly, the directional attribute on the audio m= sections
without associated MediaStreamTracks MUST be set to recvonly. If
this option is specified in the case where at least N audio
MediaStreamTracks have already been added to the PeerConnection, or N
non-rejected m= sections with media type "audio" would otherwise be
generated, it has no effect. For backwards compatibility, a value of
"true" is interpreted as equivalent to N=1.
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5.2.3.2. OfferToReceiveVideo
If the "OfferToReceiveVideo" option is specified, with an integer
value of N, the offer MUST include N non-rejected m= sections with
media type "video", even if fewer than N video MediaStreamTracks have
been added to the PeerConnection. This allows the offerer to receive
video, including multiple independent streams, even when not sending
it; accordingly, the directional attribute on the video m= sections
without associated MediaStreamTracks MUST be set to recvonly. If
this option is specified in the case where at least N video
MediaStreamTracks have already been added to the PeerConnection, or N
non-rejected m= sections with media type "video" would otherwise be
generated, it has no effect. For backwards compatibility, a value of
"true" is interpreted as equivalent to N=1.
5.2.3.3. VoiceActivityDetection
If the "VoiceActivityDetection" option is specified, with a value of
"true", the offer MUST indicate support for silence suppression in
the audio it receives by including comfort noise ("CN") codecs for
each offered audio codec, as specified in [RFC3389], Section 5.1,
except for codecs that have their own internal silence suppression
support. For codecs that have their own internal silence suppression
support, the appropriate fmtp parameters for that codec MUST be
specified to indicate that silence suppression for received audio is
desired. For example, when using the Opus codec, the "usedtx=1"
parameter would be specified in the offer. This option allows the
endpoint to significantly reduce the amount of audio bandwidth it
receives, at the cost of some fidelity, depending on the quality of
the remote VAD algorithm.
5.2.3.4. IceRestart
If the "IceRestart" option is specified, with a value of "true", the
offer MUST indicate an ICE restart by generating new ICE ufrag and
pwd attributes, as specified in RFC5245, Section 9.1.1.1. If this
option is specified on an initial offer, it has no effect (since a
new ICE ufrag and pwd are already generated). This option is useful
for reestablishing connectivity in cases where failures are detected.
5.3. Generating an Answer
When createAnswer is called, a new SDP description must be created
that is compatible with the supplied remote description as well as
the requirements specified in [I-D.ietf-rtcweb-rtp-usage]. The exact
details of this process are explained below.
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5.3.1. Initial Answers
When createAnswer is called for the first time after a remote
description has been provided, the result is known as the initial
answer. If no remote description has been installed, an answer
cannot be generated, and an error MUST be returned.
Note that the remote description SDP may not have been created by a
JSEP endpoint and may not conform to all the requirements listed in
Section 5.2. For many cases, this is not a problem. However, if any
mandatory SDP attributes are missing, or functionality listed as
mandatory-to-use above is not present, this MUST be treated as an
error, and MUST cause the affected m= sections to be marked as
rejected.
The first step in generating an initial answer is to generate
session-level attributes. The process here is identical to that
indicated in the Initial Offers section above.
The next step is to generate m= sections for each m= section that is
present in the remote offer, as specified in [RFC3264], Section 6.
For the purposes of this discussion, any session-level attributes in
the offer that are also valid as media-level attributes SHALL be
considered to be present in each m= section.
The next step is to go through each offered m= section. If there is
a local MediaStreamTrack of the same type which has been added to the
PeerConnection via addStream and not yet associated with a m=
section, and the specific m= section is either sendrecv or recvonly,
the MediaStreamTrack will be associated with the m= section at this
time. MediaStreamTracks are assigned to m= sections using the
canonical order described in Section 5.2.1. If there are more m=
sections of a certain type than MediaStreamTracks, some m= sections
will not have an associated MediaStreamTrack. If there are more
MediaStreamTracks of a certain type than compatible m= sections, only
the first N MediaStreamTracks will be able to be associated in the
constructed answer. The remainder will need to be associated in a
subsequent offer.
For each offered m= section, if the associated remote
MediaStreamTrack has been stopped, and is therefore in state "ended",
and no local MediaStreamTrack has been associated, 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.
Provided that is not the case, each m= section in the answer should
then be generated as specified in [RFC3264], Section 6.1. Because
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use of DTLS is mandatory, the <proto> field MUST be set to "UDP/TLS/
RTP/SAVPF". 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 include the following:
o If present in the offer, an "a=mid" line, as specified in
[RFC5888], Section 9.1. The "mid" value MUST match that specified
in the offer.
o If a local MediaStreamTrack has been associated, an "a=msid" line,
as specified in [I-D.ietf-mmusic-msid], Section 2.
o [OPEN ISSUE: Use of AppID]
o Depending on the directionality of the offer, the disposition of
any associated remote MediaStreamTrack, and the presence of an
associated local MediaStreamTrack, the appropriate directionality
attribute, as specified in [RFC3264], Section 6.1. If the offer
was sendrecv, and the remote MediaStreamTrack is still "live", and
there is a local MediaStreamTrack that has been associated, the
directionality MUST be set as sendrecv. If the offer was
sendonly, and the remote MediaStreamTrack is still "live", the
directionality MUST be set as recvonly. If the offer was
recvonly, and a local MediaStreamTrack has been associated, the
directionality MUST be set as sendonly. If the offer was
inactive, the directionality MUST be set as inactive.
o For each supported codec that is present in the offer, "a=rtpmap"
and "a=fmtp" lines, as specified in [RFC4566], Section 6, and
[RFC3264], Section 6.1. For audio, the codecs specified in
[I-D.ietf-rtcweb-audio], Section 3, MUST be supported. Note that
for simplicity, the answerer MAY use different payload types for
codecs than the offerer, as it is not prohibited by Section 6.1.
o If "rtx" is present in the offer, for each primary codec where RTP
retransmission should be used, a corresponding "a=rtpmap" line
indicating "rtx" with the clock rate of the primary codec and an
"a=fmtp" line that references the payload type of the primary
codec, as specified in [RFC4588], Section 8.1.
o For each supported FEC mechanism that is present in the offer, a
corresponding "a=rtpmap" line indicating the desired FEC codec.
o "a=ice-ufrag" and "a=ice-passwd" lines, as specified in [RFC5245],
Section 15.4.
o If the "trickle" ICE option is present in the offer, an "a=ice-
options" line, with the "trickle" option, as specified in
[I-D.ietf-mmusic-trickle-ice], Section 4.
o For each candidate that has been gathered during the most recent
gathering phase, an "a=candidate" line, as specified in [RFC5245],
Section 4.3., paragraph 3.
o For the current default candidate, a "c=" line, as specified in
[RFC5245], Section 4.3., paragraph 6. If no candidates have been
gathered yet, the default candidate should be set to the 'null'
value defined in [I-D.ietf-mmusic-trickle-ice], Section 5.1.
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o An "a=fingerprint" line, as specified in [RFC4572], Section 5; the
algorithm used for the fingerprint MUST match that used in the
certificate signature.
o An "a=setup" line, as specified in [RFC4145], Section 4, and
clarified for use in DTLS-SRTP scenarios in [RFC5763], Section 5.
The role value in the answer MUST be "active" or "passive"; the
"active" role is RECOMMENDED.
o If present in the offer, an "a=rtcp-mux" line, as specified in
[RFC5761], Section 5.1.1.
o If present in the offer, an "a=rtcp-rsize" line, as specified in
[RFC5506], Section 5.
o For each supported RTP header extension that is present in the
offer, an "a=extmap" line, as specified in [RFC5285], Section 5.
The list of header extensions that SHOULD/MUST be supported is
specified in [I-D.ietf-rtcweb-rtp-usage], Section 5.2. Any header
extensions that require encryption MUST be specified as indicated
in [RFC6904], Section 4.
o For each supported RTCP feedback mechanism that is present in the
offer, an "a=rtcp-fb" mechanism, as specified in [RFC4585],
Section 4.2. The list of RTCP feedback mechanisms that SHOULD/
MUST be supported is specified in [I-D.ietf-rtcweb-rtp-usage],
Section 5.1.
o If a local MediaStreamTrack has been associated, an "a=ssrc" line,
as specified in [RFC5576], Section 4.1, indicating the SSRC to be
used for sending media.
o If a local MediaStreamTrack has been associated, and RTX has been
negotiated for this m= section, another "a=ssrc" line with the RTX
SSRC, and an "a=ssrc-group" line, as specified in [RFC5576],
section 4.2, with semantics set to "FID" and including the primary
and RTX SSRCs.
o If a local MediaStreamTrack has been associated, and FEC has been
negotiated for this m= section, another "a=ssrc" line with the FEC
SSRC, and an "a=ssrc-group" line, as specified in [RFC5576],
section 4.2, with semantics set to "FEC" and including the primary
and FEC SSRCs.
o [OPEN ISSUE: Handling of a=imageattr]
If a data channel m= section has been offered, a m= section MUST also
be generated for data. The <media> field MUST be set to
"application" and the <proto> field MUST be set to "DTLS/SCTP", as
specified in [I-D.ietf-mmusic-sctp-sdp], Section 3; the "fmt" value
MUST be set to the SCTP port number, as specified in Section 4.1.
Within the data m= section, the "a=mid", "a=ice-ufrag", "a=ice-
passwd", "a=ice-options", "a=candidate", "a=fingerprint", and
"a=setup" lines MUST be included as mentioned above, along with an
"a=sctpmap" line referencing the SCTP port number and specifying the
application protocol indicated in [I-D.ietf-rtcweb-data-protocol].
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[OPEN ISSUE: the -01 of this document is missing this information.]
If "a=group" attributes with semantics of "BUNDLE" are offered,
corresponding session-level "a=group" attributes MUST be added as
specified in [RFC5888]. These attributes MUST have semantics
"BUNDLE", and MUST include the all mid identifiers from the offered
BUNDLE groups that have not been rejected. Note that regardless of
the presence of "a=bundle-only" in the offer, no m= sections in the
answer should have an "a=bundle-only" line.
Attributes that are common between all m= sections MAY be moved to
session-level, if explicitly defined to be valid at session-level.
The attributes prohibited in the creation of offers are also
prohibited in the creation of answers.
5.3.2. Subsequent Answers
5.3.3. Options Handling
5.4. Parsing an Offer
5.5. Parsing an Answer
5.6. Applying a Local Description
5.7. Applying a Remote Description
6. Configurable SDP Parameters
It is possible to change elements in the SDP returned from
createOffer before passing it to setLocalDescription. When an
implementation receives modified SDP it MUST either:
o Accept the changes and adjust its behavior to match the SDP.
o Reject the changes and return an error via the error callback.
Changes MUST NOT be silently ignored.
The following elements of the SDP media description MUST NOT be
changed between the createOffer and the setLocalDescription, since
they reflect transport attributes that are solely under browser
control, and the browser MUST NOT honor an attempt to change them:
o The number, type and port number of m-lines.
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o The generated ICE credentials (a=ice-ufrag and a=ice-pwd).
o The set of ICE candidates and their parameters (a=candidate).
The following modifications, if done by the browser to a description
between createOffer/createAnswer and the setLocalDescription, MUST be
honored by the browser:
o Remove or reorder codecs (m=)
The following parameters may be controlled by constraints passed into
createOffer/createAnswer. As an open issue, these changes may also
be be performed by manipulating the SDP returned from createOffer/
createAnswer, as indicated above, as long as the capabilities of the
endpoint are not exceeded (e.g. asking for a resolution greater than
what the endpoint can encode):
o [[OPEN ISSUE: This is a placeholder for other modifications,
which we may continue adding as use cases appear.]]
Implementations MAY choose to either honor or reject any elements not
listed in the above two categories, but must do so explicitly as
described at the beginning of this section. Note that future
standards may add new SDP elements to the list of elements which must
be accepted or rejected, but due to version skew, applications must
be prepared for implementations to accept changes which must be
rejected and vice versa.
The application can also modify the SDP to reduce the capabilities in
the offer it sends to the far side or the offer that it installs from
the far side in any way the application sees fit, as long as it is a
valid SDP offer and specifies a subset of what was in the original
offer. This is safe because the answer is not permitted to expand
capabilities and therefore will just respond to what is actually in
the offer.
As always, the application is solely responsible for what it sends to
the other party, and all incoming SDP will be processed by the
browser 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.
7. Security Considerations
The IETF has published separate documents
[I-D.ietf-rtcweb-security-arch] [I-D.ietf-rtcweb-security] describing
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the security architecture for WebRTC as a whole. The remainder of
this section describes security considerations for this document.
While formally the JSEP interface is an API, it is better to think of
it is an Internet protocol, with the JS being untrustworthy from the
perspective of the browser. Thus, the threat model of [RFC3552]
applies. In particular, JS can call the API in any order and with
any inputs, including malicious ones. This is particularly relevant
when we consider the SDP which is passed to setLocalDescription().
While correct API usage requires that the application pass in SDP
which was derived from createOffer() or createAnswer() (perhaps
suitably modified as described in Section 6, there is no guarantee
that applications do so. The browser MUST be prepared for the JS to
pass in bogus data instead.
Conversely, the application programmer MUST recognize that the JS
does not have complete control of browser 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 which wish to conceal their public IP address should
instead configure the ICE agent to use only relay candidates.
8. IANA Considerations
This document requires no actions from IANA.
9. Acknowledgements
Significant text incorporated in the draft as well and review was
provided by Harald Alvestrand and Suhas Nandakumar. Dan Burnett,
Neil Stratford, Eric Rescorla, Anant Narayanan, Andrew Hutton,
Richard Ejzak, Adam Bergkvist and Matthew Kaufman all provided
valuable feedback on this proposal.
10. References
10.1. Normative References
[I-D.ietf-mmusic-msid]
Alvestrand, H., "Cross Session Stream Identification in
the Session Description Protocol",
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draft-ietf-mmusic-msid-01 (work in progress), August 2013.
[I-D.ietf-mmusic-sctp-sdp]
Loreto, S. and G. Camarillo, "Stream Control Transmission
Protocol (SCTP)-Based Media Transport in the Session
Description Protocol (SDP)", draft-ietf-mmusic-sctp-sdp-04
(work in progress), June 2013.
[I-D.ietf-mmusic-sdp-bundle-negotiation]
Holmberg, C., Alvestrand, H., and C. Jennings,
"Multiplexing Negotiation Using Session Description
Protocol (SDP) Port Numbers",
draft-ietf-mmusic-sdp-bundle-negotiation-04 (work in
progress), June 2013.
[I-D.ietf-mmusic-sdp-mux-attributes]
Nandakumar, S., "A Framework for SDP Attributes when
Multiplexing", draft-ietf-mmusic-sdp-mux-attributes-01
(work in progress), February 2014.
[I-D.ietf-rtcweb-audio]
Valin, J. and C. Bran, "WebRTC Audio Codec and Processing
Requirements", draft-ietf-rtcweb-audio-02 (work in
progress), August 2013.
[I-D.ietf-rtcweb-data-protocol]
Jesup, R., Loreto, S., and M. Tuexen, "WebRTC Data Channel
Protocol", draft-ietf-rtcweb-data-protocol-04 (work in
progress), February 2013.
[I-D.ietf-rtcweb-rtp-usage]
Perkins, C., Westerlund, M., and J. Ott, "Web Real-Time
Communication (WebRTC): Media Transport and Use of RTP",
draft-ietf-rtcweb-rtp-usage-09 (work in progress),
September 2013.
[I-D.ietf-rtcweb-security]
Rescorla, E., "Security Considerations for WebRTC",
draft-ietf-rtcweb-security-06 (work in progress),
January 2014.
[I-D.ietf-rtcweb-security-arch]
Rescorla, E., "WebRTC Security Architecture",
draft-ietf-rtcweb-security-arch-09 (work in progress),
February 2014.
[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
Requirement Levels", BCP 14, RFC 2119, March 1997.
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[RFC3261] Rosenberg, J., Schulzrinne, H., Camarillo, G., Johnston,
A., Peterson, J., Sparks, R., Handley, M., and E.
Schooler, "SIP: Session Initiation Protocol", RFC 3261,
June 2002.
[RFC3264] Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model
with Session Description Protocol (SDP)", RFC 3264,
June 2002.
[RFC3552] Rescorla, E. and B. Korver, "Guidelines for Writing RFC
Text on Security Considerations", BCP 72, RFC 3552,
July 2003.
[RFC4145] Yon, D. and G. Camarillo, "TCP-Based Media Transport in
the Session Description Protocol (SDP)", RFC 4145,
September 2005.
[RFC4566] Handley, M., Jacobson, V., and C. Perkins, "SDP: Session
Description Protocol", RFC 4566, July 2006.
[RFC4572] Lennox, J., "Connection-Oriented Media Transport over the
Transport Layer Security (TLS) Protocol in the Session
Description Protocol (SDP)", RFC 4572, July 2006.
[RFC4585] Ott, J., Wenger, S., Sato, N., Burmeister, C., and J. Rey,
"Extended RTP Profile for Real-time Transport Control
Protocol (RTCP)-Based Feedback (RTP/AVPF)", RFC 4585,
July 2006.
[RFC5124] Ott, J. and E. Carrara, "Extended Secure RTP Profile for
Real-time Transport Control Protocol (RTCP)-Based Feedback
(RTP/SAVPF)", RFC 5124, February 2008.
[RFC5245] Rosenberg, J., "Interactive Connectivity Establishment
(ICE): A Protocol for Network Address Translator (NAT)
Traversal for Offer/Answer Protocols", RFC 5245,
April 2010.
[RFC5285] Singer, D. and H. Desineni, "A General Mechanism for RTP
Header Extensions", RFC 5285, July 2008.
[RFC5761] Perkins, C. and M. Westerlund, "Multiplexing RTP Data and
Control Packets on a Single Port", RFC 5761, April 2010.
[RFC5888] Camarillo, G. and H. Schulzrinne, "The Session Description
Protocol (SDP) Grouping Framework", RFC 5888, June 2010.
[RFC6904] Lennox, J., "Encryption of Header Extensions in the Secure
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Real-time Transport Protocol (SRTP)", RFC 6904,
April 2013.
[RFC7022] Begen, A., Perkins, C., Wing, D., and E. Rescorla,
"Guidelines for Choosing RTP Control Protocol (RTCP)
Canonical Names (CNAMEs)", RFC 7022, September 2013.
10.2. Informative References
[I-D.ietf-mmusic-trickle-ice]
Ivov, E., Rescorla, E., and J. Uberti, "Trickle ICE:
Incremental Provisioning of Candidates for the Interactive
Connectivity Establishment (ICE) Protocol",
draft-ietf-mmusic-trickle-ice-00 (work in progress),
March 2013.
[I-D.nandakumar-rtcweb-sdp]
Nandakumar, S. and C. Jennings, "SDP for the WebRTC",
draft-nandakumar-rtcweb-sdp-02 (work in progress),
July 2013.
[RFC3389] Zopf, R., "Real-time Transport Protocol (RTP) Payload for
Comfort Noise (CN)", RFC 3389, September 2002.
[RFC3556] Casner, S., "Session Description Protocol (SDP) Bandwidth
Modifiers for RTP Control Protocol (RTCP) Bandwidth",
RFC 3556, July 2003.
[RFC3960] Camarillo, G. and H. Schulzrinne, "Early Media and Ringing
Tone Generation in the Session Initiation Protocol (SIP)",
RFC 3960, December 2004.
[RFC4568] Andreasen, F., Baugher, M., and D. Wing, "Session
Description Protocol (SDP) Security Descriptions for Media
Streams", RFC 4568, July 2006.
[RFC4588] Rey, J., Leon, D., Miyazaki, A., Varsa, V., and R.
Hakenberg, "RTP Retransmission Payload Format", RFC 4588,
July 2006.
[RFC5506] Johansson, I. and M. Westerlund, "Support for Reduced-Size
Real-Time Transport Control Protocol (RTCP): Opportunities
and Consequences", RFC 5506, April 2009.
[RFC5576] Lennox, J., Ott, J., and T. Schierl, "Source-Specific
Media Attributes in the Session Description Protocol
(SDP)", RFC 5576, June 2009.
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[RFC5763] Fischl, J., Tschofenig, H., and E. Rescorla, "Framework
for Establishing a Secure Real-time Transport Protocol
(SRTP) Security Context Using Datagram Transport Layer
Security (DTLS)", RFC 5763, May 2010.
[RFC5764] McGrew, D. and E. Rescorla, "Datagram Transport Layer
Security (DTLS) Extension to Establish Keys for the Secure
Real-time Transport Protocol (SRTP)", RFC 5764, May 2010.
[W3C.WD-webrtc-20140617]
Bergkvist, A., Burnett, D., Narayanan, A., and C.
Jennings, "WebRTC 1.0: Real-time Communication Between
Browsers", World Wide Web Consortium WD WD-webrtc-
20140617, June 2014,
<http://www.w3.org/TR/2011/WD-webrtc-20140617>.
Appendix A. JSEP Implementation Examples
A.1. Example API Flows
Below are several sample flows for the new PeerConnection and library
APIs, demonstrating when the various APIs are called in different
situations and with various transport protocols. For clarity and
simplicity, the createOffer/createAnswer calls are assumed to be
synchronous in these examples, whereas the actual APIs are async.
A.1.1. Call using ROAP
This example demonstrates a ROAP call, without the use of trickle
candidates.
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// Call is initiated toward Answerer
OffererJS->OffererUA: pc = new PeerConnection();
OffererJS->OffererUA: pc.addStream(localStream, null);
OffererUA->OffererJS: iceCallback(candidate);
OffererJS->OffererUA: offer = pc.createOffer(null);
OffererJS->OffererUA: pc.setLocalDescription("offer", offer);
OffererJS->AnswererJS: {"type":"OFFER", "sdp":offer }
// OFFER arrives at Answerer
AnswererJS->AnswererUA: pc = new PeerConnection();
AnswererJS->AnswererUA: pc.setRemoteDescription("offer", msg.sdp);
AnswererUA->AnswererJS: onaddstream(remoteStream);
AnswererUA->OffererUA: iceCallback(candidate);
// Answerer accepts call
AnswererJS->AnswererUA: pc.addStream(localStream, null);
AnswererJS->AnswererUA: answer = pc.createAnswer(msg.sdp, null);
AnswererJS->AnswererUA: pc.setLocalDescription("answer", answer);
AnswererJS->OffererJS: {"type":"ANSWER","sdp":answer }
// ANSWER arrives at Offerer
OffererJS->OffererUA: pc.setRemoteDescription("answer", answer);
OffererUA->OffererJS: onaddstream(remoteStream);
// ICE Completes (at Answerer)
AnswererUA->OffererUA: Media
// ICE Completes (at Offerer)
OffererJS->AnswererJS: {"type":"OK" }
OffererUA->AnswererUA: Media
A.1.2. Call using XMPP
This example demonstrates an XMPP call, making use of trickle
candidates.
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// Call is initiated toward Answerer
OffererJS->OffererUA: pc = new PeerConnection();
OffererJS->OffererUA: pc.addStream(localStream, null);
OffererJS->OffererUA: offer = pc.createOffer(null);
OffererJS->OffererUA: pc.setLocalDescription("offer", offer);
OffererJS: xmpp = createSessionInitiate(offer);
OffererJS->AnswererJS: <jingle action="session-initiate"/>
OffererJS->OffererUA: pc.startIce();
OffererUA->OffererJS: onicecandidate(cand);
OffererJS: createTransportInfo(cand);
OffererJS->AnswererJS: <jingle action="transport-info"/>
// session-initiate arrives at Answerer
AnswererJS->AnswererUA: pc = new PeerConnection();
AnswererJS: offer = parseSessionInitiate(xmpp);
AnswererJS->AnswererUA: pc.setRemoteDescription("offer", offer);
AnswererUA->AnswererJS: onaddstream(remoteStream);
// transport-infos arrive at Answerer
AnswererJS->AnswererUA: candidate = parseTransportInfo(xmpp);
AnswererJS->AnswererUA: pc.addIceCandidate(candidate);
AnswererUA->AnswererJS: onicecandidate(cand)
AnswererJS: createTransportInfo(cand);
AnswererJS->OffererJS: <jingle action="transport-info"/>
// transport-infos arrive at Offerer
OffererJS->OffererUA: candidates = parseTransportInfo(xmpp);
OffererJS->OffererUA: pc.addIceCandidate(candidates);
// Answerer accepts call
AnswererJS->AnswererUA: pc.addStream(localStream, null);
AnswererJS->AnswererUA: answer = pc.createAnswer(offer, null);
AnswererJS: xmpp = createSessionAccept(answer);
AnswererJS->AnswererUA: pc.setLocalDescription("answer", answer);
AnswererJS->OffererJS: <jingle action="session-accept"/>
// session-accept arrives at Offerer
OffererJS: answer = parseSessionAccept(xmpp);
OffererJS->OffererUA: pc.setRemoteDescription("answer", answer);
OffererUA->OffererJS: onaddstream(remoteStream);
// ICE Completes (at Answerer)
AnswererUA->OffererUA: Media
// ICE Completes (at Offerer)
OffererUA->AnswererUA: Media
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A.1.3. Adding video to a call, using XMPP
This example demonstrates an XMPP call, where the XMPP content-add
mechanism is used to add video media to an existing session. For
simplicity, candidate exchange is not shown.
Note that the offerer for the change to the session may be different
than the original call offerer.
// Offerer adds video stream
OffererJS->OffererUA: pc.addStream(videoStream)
OffererJS->OffererUA: offer = pc.createOffer(null);
OffererJS: xmpp = createContentAdd(offer);
OffererJS->OffererUA: pc.setLocalDescription("offer", offer);
OffererJS->AnswererJS: <jingle action="content-add"/>
// content-add arrives at Answerer
AnswererJS: offer = parseContentAdd(xmpp);
AnswererJS->AnswererUA: pc.setRemoteDescription("offer", offer);
AnswererJS->AnswererUA: answer = pc.createAnswer(offer, null);
AnswererJS->AnswererUA: pc.setLocalDescription("answer", answer);
AnswererJS: xmpp = createContentAccept(answer);
AnswererJS->OffererJS: <jingle action="content-accept"/>
// content-accept arrives at Offerer
OffererJS: answer = parseContentAccept(xmpp);
OffererJS->OffererUA: pc.setRemoteDescription("answer", answer);
A.1.4. Simultaneous add of video streams, using XMPP
This example demonstrates an XMPP call, where new video sources are
added at the same time to a call that already has video; since adding
these sources only affects one side of the call, there is no
conflict. The XMPP description-info mechanism is used to indicate
the new sources to the remote side.
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// Offerer and "Answerer" add video streams at the same time
OffererJS->OffererUA: pc.addStream(offererVideoStream2)
OffererJS->OffererUA: offer = pc.createOffer(null);
OffererJS: xmpp = createDescriptionInfo(offer);
OffererJS->OffererUA: pc.setLocalDescription("offer", offer);
OffererJS->AnswererJS: <jingle action="description-info"/>
AnswererJS->AnswererUA: pc.addStream(answererVideoStream2)
AnswererJS->AnswererUA: offer = pc.createOffer(null);
AnswererJS: xmpp = createDescriptionInfo(offer);
AnswererJS->AnswererUA: pc.setLocalDescription("offer", offer);
AnswererJS->OffererJS: <jingle action="description-info"/>
// description-info arrives at "Answerer", and is acked
AnswererJS: offer = parseDescriptionInfo(xmpp);
AnswererJS->OffererJS: <iq type="result"/> // ack
// description-info arrives at Offerer, and is acked
OffererJS: offer = parseDescriptionInfo(xmpp);
OffererJS->AnswererJS: <iq type="result"/> // ack
// ack arrives at Offerer; remote offer is used as an answer
OffererJS->OffererUA: pc.setRemoteDescription("answer", offer);
// ack arrives at "Answerer"; remote offer is used as an answer
AnswererJS->AnswererUA: pc.setRemoteDescription("answer", offer);
A.1.5. Call using SIP
This example demonstrates a simple SIP call (e.g. where the client
talks to a SIP proxy over WebSockets).
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// Call is initiated toward Answerer
OffererJS->OffererUA: pc = new PeerConnection();
OffererJS->OffererUA: pc.addStream(localStream, null);
OffererUA->OffererJS: onicecandidate(candidate);
OffererJS->OffererUA: offer = pc.createOffer(null);
OffererJS->OffererUA: pc.setLocalDescription("offer", offer);
OffererJS: sip = createInvite(offer);
OffererJS->AnswererJS: SIP INVITE w/ SDP
// INVITE arrives at Answerer
AnswererJS->AnswererUA: pc = new PeerConnection();
AnswererJS: offer = parseInvite(sip);
AnswererJS->AnswererUA: pc.setRemoteDescription("offer", offer);
AnswererUA->AnswererJS: onaddstream(remoteStream);
AnswererUA->OffererUA: onicecandidate(candidate);
// Answerer accepts call
AnswererJS->AnswererUA: pc.addStream(localStream, null);
AnswererJS->AnswererUA: answer = pc.createAnswer(offer, null);
AnswererJS: sip = createResponse(200, answer);
AnswererJS->AnswererUA: pc.setLocalDescription("answer", answer);
AnswererJS->OffererJS: 200 OK w/ SDP
// 200 OK arrives at Offerer
OffererJS: answer = parseResponse(sip);
OffererJS->OffererUA: pc.setRemoteDescription("answer", answer);
OffererUA->OffererJS: onaddstream(remoteStream);
OffererJS->AnswererJS: ACK
// ICE Completes (at Answerer)
AnswererUA->OffererUA: Media
// ICE Completes (at Offerer)
OffererUA->AnswererUA: Media
A.1.6. Handling early media (e.g. 1-800-GO FEDEX), using SIP
This example demonstrates how early media could be handled; for
simplicity, only the offerer side of the call is shown.
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// Call is initiated toward Answerer
OffererJS->OffererUA: pc = new PeerConnection();
OffererJS->OffererUA: pc.addStream(localStream, null);
OffererUA->OffererJS: onicecandidate(candidate);
OffererJS->OffererUA: offer = pc.createOffer(null);
OffererJS->OffererUA: pc.setLocalDescription("offer", offer);
OffererJS: sip = createInvite(offer);
OffererJS->AnswererJS: SIP INVITE w/ SDP
// 180 Ringing is received by offerer, w/ SDP
OffererJS: answer = parseResponse(sip);
OffererJS->OffererUA: pc.setRemoteDescription("pranswer", answer);
OffererUA->OffererJS: onaddstream(remoteStream);
// ICE Completes (at Offerer)
OffererUA->AnswererUA: Media
// 200 OK arrives at Offerer
OffererJS: answer = parseResponse(sip);
OffererJS->OffererUA: pc.setRemoteDescription("answer", answer);
OffererJS->AnswererJS: ACK
A.2. Example Session Descriptions
A.2.1. createOffer
This SDP shows a typical initial offer, created by createOffer for a
PeerConnection with a single audio MediaStreamTrack, a single video
MediaStreamTrack, and a single data channel. Host candidates have
also already been gathered. Note some lines have been broken into
two lines for formatting reasons.
v=0
o=- 4962303333179871722 1 IN IP4 0.0.0.0
s=-
t=0 0
a=msid-semantic:WMS
a=group:BUNDLE audio video data
m=audio 56500 UDP/TLS/RTP/SAVPF 111 0 8 126
c=IN IP4 192.0.2.1
a=rtcp:56501 IN IP4 192.0.2.1
a=candidate:3348148302 1 udp 2113937151 192.0.2.1 56500
typ host generation 0
a=candidate:3348148302 2 udp 2113937151 192.0.2.1 56501
typ host generation 0
a=ice-ufrag:ETEn1v9DoTMB9J4r
a=ice-pwd:OtSK0WpNtpUjkY4+86js7ZQl
a=ice-options:trickle
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a=mid:audio
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=sendrecv
a=rtcp-mux
a=rtcp-rsize
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=rtpmap:111 opus/48000/2
a=fmtp:111 minptime=10
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:126 telephone-event/8000
a=maxptime:60
a=ssrc:1732846380 cname:EocUG1f0fcg/yvY7
a=msid:47017fee-b6c1-4162-929c-a25110252400
f83006c5-a0ff-4e0a-9ed9-d3e6747be7d9
m=video 56502 UDP/TLS/RTP/SAVPF 100 115 116 117
c=IN IP4 192.0.2.1
a=rtcp:56503 IN IP4 192.0.2.1
a=candidate:3348148302 1 udp 2113937151 192.0.2.1 56502
typ host generation 0
a=candidate:3348148302 2 udp 2113937151 192.0.2.1 56503
typ host generation 0
a=ice-ufrag:BGKkWnG5GmiUpdIV
a=ice-pwd:mqyWsAjvtKwTGnvhPztQ9mIf
a=ice-options:trickle
a=mid:video
a=extmap:2 urn:ietf:params:rtp-hdrext:toffset
a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
a=sendrecv
a=rtcp-mux
a=rtcp-rsize
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=rtpmap:100 VP8/90000
a=rtcp-fb:100 ccm fir
a=rtcp-fb:100 nack
a=rtcp-fb:100 goog-remb
a=rtpmap:115 rtx/90000
a=fmtp:115 apt=100
a=rtpmap:116 red/90000
a=rtpmap:117 ulpfec/90000
a=ssrc:1366781083 cname:EocUG1f0fcg/yvY7
a=ssrc:1366781084 cname:EocUG1f0fcg/yvY7
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a=ssrc:1366781085 cname:EocUG1f0fcg/yvY7
a=ssrc-group:FID 1366781083 1366781084
a=ssrc-group:FEC 1366781083 1366781085
a=msid:61317484-2ed4-49d7-9eb7-1414322a7aae
f30bdb4a-5db8-49b5-bcdc-e0c9a23172e0
m=application 56504 DTLS/SCTP 5000
c=IN IP4 192.0.2.1
a=candidate:3348148302 1 udp 2113937151 192.0.2.1 56504
typ host generation 0
a=ice-ufrag:VD5v2BnbZm3mgP3d
a=ice-pwd:+Jlkuox+VVIUDqxcfIDuTZMH
a=ice-options:trickle
a=mid:data
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=sctpmap:5000 webrtc-datachannel 16
A.2.2. createAnswer
This SDP shows a typical initial answer to the above offer, created
by createAnswer for a PeerConnection with a single audio
MediaStreamTrack, a single video MediaStreamTrack, and a single data
channel. Host candidates have also already been gathered. Note some
lines have been broken into two lines for formatting reasons.
v=0
o=- 6729291447651054566 1 IN IP4 0.0.0.0
s=-
t=0 0
a=msid-semantic:WMS
a=group:BUNDLE audio video data
m=audio 20000 UDP/TLS/RTP/SAVPF 111 0 8 126
c=IN IP4 192.0.2.2
a=candidate:2299743422 1 udp 2113937151 192.0.2.2 20000
typ host generation 0
a=ice-ufrag:6sFvz2gdLkEwjZEr
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=mid:audio
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=sendrecv
a=rtcp-mux
a=rtpmap:111 opus/48000/2
a=fmtp:111 minptime=10
a=rtpmap:0 PCMU/8000
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a=rtpmap:8 PCMA/8000
a=rtpmap:126 telephone-event/8000
a=maxptime:60
a=ssrc:3429951804 cname:Q/NWs1ao1HmN4Xa5
a=msid:PI39StLS8W7ZbQl1sJsWUXkr3Zf12fJUvzQ1
PI39StLS8W7ZbQl1sJsWUXkr3Zf12fJUvzQ1a0
m=video 20000 UDP/TLS/RTP/SAVPF 100 115 116 117
c=IN IP4 192.0.2.2
a=candidate:2299743422 1 udp 2113937151 192.0.2.2 20000
typ host generation 0
a=ice-ufrag:6sFvz2gdLkEwjZEr
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=mid:video
a=extmap:2 urn:ietf:params:rtp-hdrext:toffset
a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
a=sendrecv
a=rtcp-mux
a=rtpmap:100 VP8/90000
a=rtcp-fb:100 ccm fir
a=rtcp-fb:100 nack
a=rtcp-fb:100 goog-remb
a=rtpmap:115 rtx/90000
a=fmtp:115 apt=100
a=rtpmap:116 red/90000
a=rtpmap:117 ulpfec/90000
a=ssrc:3229706345 cname:Q/NWs1ao1HmN4Xa5
a=ssrc:3229706346 cname:Q/NWs1ao1HmN4Xa5
a=ssrc:3229706347 cname:Q/NWs1ao1HmN4Xa5
a=ssrc-group:FID 3229706345 3229706346
a=ssrc-group:FEC 3229706345 3229706347
a=msid:PI39StLS8W7ZbQl1sJsWUXkr3Zf12fJUvzQ1
PI39StLS8W7ZbQl1sJsWUXkr3Zf12fJUvzQ1v0
m=application 20000 DTLS/SCTP 5000
c=IN IP4 192.0.2.2
a=candidate:2299743422 1 udp 2113937151 192.0.2.2 20000
typ host generation 0
a=ice-ufrag:6sFvz2gdLkEwjZEr
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=mid:data
a=sctpmap:5000 webrtc-datachannel 16
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A.2.3. Call Flows
Example SDP for WebRTC call flows can be found in
[I-D.nandakumar-rtcweb-sdp]. [TODO: should these call flows be
merged into this section?]
Appendix B. Change log
Changes in draft-06:
o Reworked handling of m= line recycling.
o Added handling of BUNDLE and bundle-only.
o Clarified handling of rollback.
o Added text describing the ICE Candidate Pool and its behavior.
o Allowed OfferToReceiveX to create multiple recvonly m= sections.
Changes in draft-05:
o Fixed several issues identified in the createOffer/Answer sections
during document review.
o Updated references.
Changes in draft-04:
o Filled in sections on createOffer and createAnswer.
o Added SDP examples.
o Fixed references.
Changes in draft-03:
o Added text describing relationship to W3C specification
Changes in draft-02:
o Converted from nroff
o Removed comparisons to old approaches abandoned by the working
group
o Removed stuff that has moved to W3C specification
o Align SDP handling with W3C draft
o Clarified section on forking.
Changes in draft-01:
o Added diagrams for architecture and state machine.
o Added sections on forking and rehydration.
o Clarified meaning of "pranswer" and "answer".
o Reworked how ICE restarts and media directions are controlled.
o Added list of parameters that can be changed in a description.
o Updated suggested API and examples to match latest thinking.
o Suggested API and examples have been moved to an appendix.
Changes in draft -00:
Uberti, et al. Expires January 5, 2015 [Page 49]
Internet-Draft JSEP July 2014
o Migrated from draft-uberti-rtcweb-jsep-02.
Authors' Addresses
Justin Uberti
Google
747 6th Ave S
Kirkland, WA 98033
USA
Email: justin@uberti.name
Cullen Jennings
Cisco
170 West Tasman Drive
San Jose, CA 95134
USA
Email: fluffy@iii.ca
Eric Rescorla (editor)
Mozilla
331 Evelyn Ave
Mountain View, CA 94041
USA
Email: ekr@rtfm.com
Uberti, et al. Expires January 5, 2015 [Page 50]