Network Working Group J. Uberti
Internet-Draft Google
Intended status: Standards Track C. Jennings
Expires: September 4, 2012 Cisco Systems, Inc.
March 3, 2012
Javascript Session Establishment Protocol
draft-ietf-rtcweb-jsep-00
Abstract
This document proposes a mechanism for allowing a Javascript
application to fully control the signaling plane of a multimedia
session, and discusses how this would work with existing signaling
protocols.
This document is an input document for discussion. It should be
discussed in the RTCWEB WG list, rtcweb@ietf.org.
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
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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 July 26, 2012.
Copyright Notice
Copyright (c) 2012 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
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publication of this document. Please review these documents
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to this document. Code Components extracted from this document must
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include Simplified BSD License text as described in Section 4.e of
the Trust Legal Provisions and are provided without warranty as
described in the Simplified BSD License.
Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . . 4
Terminology . . . . . . . . . . . . . . . . . . . . . . . . . . . 5
2. JSEP Approach . . . . . . . . . . . . . . . . . . . . . . . . . 5
3. Other Approaches Considered . . . . . . . . . . . . . . . . . . 6
4. Semantics and Syntax . . . . . . . . . . . . . . . . . . . . . 7
4.1. Signaling Model . . . . . . . . . . . . . . . . . . . . . . 7
4.2. Session Descriptions . . . . . . . . . . . . . . . . . . . 7
4.3. Session Description Format . . . . . . . . . . . . . . . . 8
4.4. Separation of Signaling and ICE State Machines . . . . . . 9
4.5. ICE Candidate Trickling . . . . . . . . . . . . . . . . . . 9
4.6. ICE Candidate Format . . . . . . . . . . . . . . . . . . . 10
5. Media Setup Overview . . . . . . . . . . . . . . . . . . . . . 10
5.1. Initiating the Session . . . . . . . . . . . . . . . . . . 10
5.1.1. Generating An Offer . . . . . . . . . . . . . . . . . . 11
5.1.2. Applying the Offer . . . . . . . . . . . . . . . . . . 11
5.1.3. Initiating ICE . . . . . . . . . . . . . . . . . . . . 11
5.1.4. Serializing the Offer and Candidates . . . . . . . . . 11
5.2. Receiving the Session . . . . . . . . . . . . . . . . . . . 12
5.2.1. Receiving the Offer . . . . . . . . . . . . . . . . . . 12
5.2.2. Initiating ICE . . . . . . . . . . . . . . . . . . . . 12
5.2.3. Handling ICE Messages . . . . . . . . . . . . . . . . . 12
5.2.4. Generating the Answer . . . . . . . . . . . . . . . . . 12
5.2.5. Applying the Answer . . . . . . . . . . . . . . . . . . 13
5.2.6. Serializing the Answer . . . . . . . . . . . . . . . . 13
5.3. Completing the Session . . . . . . . . . . . . . . . . . . 13
5.3.1. Receiving the Answer . . . . . . . . . . . . . . . . . 13
5.4. Updates to the Session . . . . . . . . . . . . . . . . . . 13
6. Proposed WebRTC API changes . . . . . . . . . . . . . . . . . . 14
6.1. PeerConnection API . . . . . . . . . . . . . . . . . . . . 14
6.1.1 MediaHints . . . . . . . . . . . . . . . . . . . . . . . 15
6.1.2 createOffer . . . . . . . . . . . . . . . . . . . . . . 16
6.1.3 createAnswer . . . . . . . . . . . . . . . . . . . . . . 16
6.1.4 SDP_OFFER, SDP_PRANSWER, and SDP_ANSWER . . . . . . . . 17
6.1.5 setLocalDescription . . . . . . . . . . . . . . . . . . 17
6.1.6 setRemoteDescription . . . . . . . . . . . . . . . . . . 18
6.1.7 localDescription . . . . . . . . . . . . . . . . . . . . 18
6.1.8 remoteDescription . . . . . . . . . . . . . . . . . . . 18
6.1.9 IceOptions . . . . . . . . . . . . . . . . . . . . . . . 19
6.1.10 startIce . . . . . . . . . . . . . . . . . . . . . . . 19
6.1.11 processIceMessage . . . . . . . . . . . . . . . . . . . 19
7. Example API Flows . . . . . . . . . . . . . . . . . . . . . . . 20
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7.1. Call using ROAP . . . . . . . . . . . . . . . . . . . . . . 20
7.2. Call using XMPP . . . . . . . . . . . . . . . . . . . . . . 20
7.3. Adding video to a call, using XMPP . . . . . . . . . . . . 22
7.4. Simultaneous add of video streams, using XMPP . . . . . . . 22
7.5. Call using SIP . . . . . . . . . . . . . . . . . . . . . . 23
7.6. Handling early media (e.g. 1-800-FEDEX), using SIP . . . . 24
8. Example Application . . . . . . . . . . . . . . . . . . . . . . 24
9. Security Considerations . . . . . . . . . . . . . . . . . . . . 26
10. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 26
11. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 26
12. References . . . . . . . . . . . . . . . . . . . . . . . . . . 26
12.1. Normative References . . . . . . . . . . . . . . . . . . . 26
12.2. Informative References . . . . . . . . . . . . . . . . . . 27
Appendix A. Open Issues . . . . . . . . . . . . . . . . . . . . . 27
Appendix B. Change log . . . . . . . . . . . . . . . . . . . . . . 27
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 27
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1. Introduction
The general 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.
The original spec for WebRTC attempted to implement this protocol-
agnostic signaling by providing a mechanism to exchange session
descriptions in the form of SDP blobs. Upon starting a session, the
browser would generate a SDP blob, which would be passed to the
application for transport over its preferred signaling protocol. On
the remote side, this blob would be passed into the browser from the
application, and the browser would then generate a blob of its own in
response. Upon transmission back to the initiator, this blob would be
plugged into their browser, and the handshake would be complete.
Experimentation with this mechanism turned up several shortcomings,
which generally stemmed from there being insufficient context at the
browser to fully determine the meaning of a SDP blob. For example,
determining whether a blob is an offer or an answer, or
differentiating a new offer from a retransmit.
The ROAP proposal, specified in http://tools.ietf.org/html/draft-
jennings-rtcweb-signaling-01, attempted to resolve these issues by
providing additional structure in the messaging - in essence, to
create a generic signaling protocol that specifies how the browser
signaling state machine should operate. However, even though the
protocol is abstracted, the state machine forces a least-common-
denominator approach on the signaling interactions. For example, in
Jingle, the call initiator can provide additional ICE candidates even
after the initial offer has been sent, which allows the offer to be
sent immediately for quicker call startup. However, in the browser
state machine, there is no notion of sending an updated offer before
the initial offer has been responded to, rendering this functionality
impossible.
While specific concerns like this could be addressed by modifying the
generic protocol, others would likely be discovered later. The main
reason this mechanism is inflexible is because it embeds a signaling
state machine within the browser. Since the browser generates the
session descriptions on its own, and fully controls the possible
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states and advancement of the signaling state machine, modification
of the session descriptions or use of alternate state machines
becomes difficult or impossible.
The browser environment also has its own challenges that cause
problems for an embedded signaling state machine. One of these is
that the user may reload the web page at any time. If this happens,
and the state machine is being run at a server, the server can simply
push the current state back down to the page and resume the call
where it left off. If instead the state machine is run at the browser
end, and is instantiated within, for example, the PeerConnection
object, that state machine will be reinitialized when the page is
reloaded and the JavaScript re-executed. This actually complicates
the design of any interoperability service, as all cases where an
offer or answer has already been generated but is now "forgotten"
must now be handled by trying to move the client state machine
forward to the same state it had been in previously in order to match
what has already been delivered to and/or answered by the far side,
or handled by ensuring that aborts are cleanly handled from every
state and the negotiation rapidly restarted.
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 RFC 2119 [RFC2119].
2. JSEP Approach
To resolve these issues, this document proposes the Javascript
Session Establishment Protocol (JSEP) that pulls the signaling state
machine out of the browser and into Javascript. This mechanism
effectively removes the browser almost completely from the core
signaling flow; the only interface needed is a way for the
application to pass in the local and remote session descriptions
negotiated by whatever signaling mechanism is used, and a way to
interact with the ICE state machine.
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 on
PeerConnection. The application can do massaging of that offer, if it
wants to, and then uses it to set up its local config via a
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 a
setRemoteDescription() API.
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When the call is accepted, the callee uses a 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
exchanges.
Regarding ICE, in this approach we decouple the ICE state machine
from the overall signaling state machine; the ICE state machine must
remain in the browser, given that only the browser has the necessary
knowledge of candidates and other transport info. While transport has
typically been lumped in with session descriptions, performing this
separation it provides additional flexibility. In protocols that
decouple session descriptions from transport, such as Jingle, the
transport information can be sent separately; in protocols that
don't, such as SIP, the information can be easily aggregated and
recombined. Sending transport information separately can allow for
faster ICE and DTLS startup, since the necessary roundtrips can occur
while waiting for the remote side to accept the session.
The JSEP approach does come with a minor downside. As the application
now is responsible for driving the signaling state machine, slightly
more application code is necessary to perform call setup; the
application must call the right APIs at the right times, and convert
the session desciptions and ICE information into the defined messages
of its chosen signaling protocol, instead of simply forwarding the
messages emitted from the browser.
One way to mitigate this is to provide a Javascript library that
hides this complexity from the developer, which would implement the
state machine and serialization of the desired signaling protocol.
For example, this library could convert easily adapt the JSEP API
into the exact ROAP API, thereby implementing the ROAP signaling
protocol. Such a library could of course also implement other popular
signaling protocols, including SIP or Jingle. In this fashion we can
enable greater control for the experienced developer without forcing
any additional complexity on the novice developer.
3. Other Approaches Considered
Another approach that was considered for JSEP was to move the
mechanism for generating offers and answers out of the browser as
well. This approach would add a getCapabilities API which would
provide the application with the information it needed in order to
generate 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
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correct answer from an arbitrary offer and available capabilities.
While this could certainly be addressed by using a library like the
one mentioned above, some experimentation also indicates that coming
up with a sufficiently complete getCapabilities API is a nontrivial
undertaking. Nevertheless, if we wanted to go down this road, JSEP
makes it significantly easier; if a getCapabilities API is added in
the future, the application can generate session descriptions
accordingly and pass those to the
setLocalDescription/setRemoteDescription APIs added by JSEP. (Even
with JSEP, an application could still perform its own browser
fingerprinting and generate approximate session descriptions as a
result.)
Note also that while JSEP transfers more control to Javascript, it is
not intended to be an example of a "low-level" API. The general
argument against a low-level API is that there are too many necessary
API points, and they can be called in any order, leading to something
that is hard to specify and test. In the approach proposed here,
control is performed via session descriptions; this requires only a
few APIs to handle these descriptions, and they are evaluated in a
specific fashion, which reduces the number of possible states and
interactions.
4. Semantics and Syntax
4.1. Signaling Model
JSEP does not specify a particular signaling model or state machine,
other than the generic need to exchange RFC 3264 offers and answers
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 PeerConnection. However, the actual
mechanism by which these offers and answers are communicated to the
remote side, including addressing, retransmission, forking, and glare
handling, is left entirely up to the application.
4.2. Session Descriptions
In order to establish the media plane, PeerConnection 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 was sent or received 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 decode,
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and what the remote party should send. Not all parameters follow this
rule; the SRTP parameters [RFC4568] sent to a remote party indicate
what the local side will use to encrypt, and thereby how the remote
party should expect to receive.
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.
Lastly, while the exact media parameters are only known only after a
offer and an answer have been exchanged, it is possible for the
offerer to receive media after they have sent an offer and before
they have received an answer. To properly process incoming media in
this case, the offerer's media handler must be aware of the details
of the offerer before the answer arrives.
Therefore, in order to handle session descriptions properly,
PeerConnection 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 a setLocalDescription and a
setRemoteDescription method, and both these methods take as a first
parameter either the value SDP_OFFER, SDP_PRANSWER (for a non-final
answer) or SDP_ANSWER (for a final answer). This satisfies the
requirements listed above for both the offerer, who first calls
setLocalDescription(SDP_OFFER, sdp) and then later
setRemoteDescription(SDP_ANSWER, sdp), as well as for the answerer,
who first calls setRemoteDescription(SDP_OFFER, sdp) and then later
setLocalDescription(SDP_ANSWER, sdp).
While it could be possible to implicitly determine the value of the
offer/answer argument inside of PeerConnection, requiring it to be
specified explicitly seems substantially more robust, allowing
invalid combinations (i.e. an answer before an offer) to generate an
appropriate error.
4.3. Session Description Format
In the current WebRTC specification, session descriptions are
formatted as SDP messages. While this format is not optimal for
manipulation from Javascript, it is widely accepted, and frequently
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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 continues to use 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 we were able to agree on a JSON format for
session descriptions, we could easily enable this object to
generate/expect JSON.
Other methods may be added to SessionDescription in the future to
simplify handling of SessionDescriptions from Javascript.
4.4. Separation of Signaling and ICE State Machines
Previously, PeerConnection operated two state machines, referred to
in the spec as an "ICE Agent", which handles the establishment of
peer-to-peer connectivity, and an "SDP Agent", which handles the
state of the offer-answer signaling. The states of these state
machines were exposed through the iceState and sdpState attributes on
PeerConnection, with an additional readyState attribute that
reflected the high-level state of the PeerConnection.
JSEP does away with the SDP Agent within the browser; this
functionality is now controlled directly by the application, which
uses the setLocalDescription and setRemoteDescription APIs to tell
PeerConnection what SDP has been negotiated. The ICE Agent remains in
the browser, as it still needs to perform gathering of candidates,
connectivity checking, and related ICE functionality.
The net effect of this is that sdpState goes away, and
processSignalingMessage becomes processIceMessage, which now
specifically handles incoming ICE candidates. To allow the
application to control exactly when it wants to start ICE negotiation
(e.g. either on receipt of the call, or only after accepting the
call), a startIce method has been added.
4.5. 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. This 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 allocate all
possible candidates, resulting in faster call startup in many cases.
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JSEP supports optional candidate trickling by providing APIs that
provide control and feedback on the ICE candidate gathering process.
Applications that support candidate trickling can send the initial
offer immediately and send individual candidates when they get a
callback with a new candidate; applications that do not support this
feature can simply wait for the callback that indicates gathering is
complete, and simply create and send their offer, with all the
candidates, at this time.
To be clear, aplications that do not make use of candidate tricking
can ignore processIceMessage entirely, and use IceCallback solely to
indicate when candidate gathering is complete.
4.6. ICE Candidate Format
As with session descriptions, we choose to provide an IceCandidate
object that provides some abstraction, but can be easily converted
to/from SDP a=candidate lines.
The IceCandidate object has a field to indicate which m= line it
should be associated with, and a method to convert to a SDP
representation, ex:
a=candidate:1 1 UDP 1694498815 66.77.88.99 10000 typ host
Currently, a=candidate lines are the only thing that are contained
within IceCandidate, as this is the only information that is needed
that is not present in the initial offer (i.e. for trickle
candidates).
5. Media Setup Overview
The example here shows a typical call setup using the JSEP model. We
assume the following architecture in this example, where UA is
synonymous with "browser", and JS is synonymous with "web
application":
OffererUA <-> OffererJS <-> WebServer <-> AnswererJS <-> AnswererUA
5.1. Initiating the Session
The initiator creates a PeerConnection, installs its IceCallback, and
adds the desired MediaStreams (presumably obtained via getUserMedia).
The PeerConnection is in the NEW state.
OffererJS->OffererUA: var pc = new PeerConnection(config, iceCb);
OffererJS->OffererUA: pc.addStream(stream);
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5.1.1. Generating An Offer
The initiator then creates a session description to offer to the
callee. This description includes the codecs and other necessary
session parameters, as well as information about each of the streams
that has been added (e.g. SSRC, CNAME, etc.) The created description
includes all parameters that the offerer's UA supports; if the
initiator wants to influence the created offer, they can pass in a
MediaHints object to createOffer that allows for customization (e.g.
if the initiator wants to receive but not send video). The initiator
can also directly manipulate the created session description as well,
perhaps if it wants to change the priority of the offered codecs.
OffererJS->OffererUA: var offer = pc.createOffer(null);
5.1.2. Applying the Offer
The initiator then instructs the PeerConnection to use this offer as
the local description for this session, i.e. what codecs it will use
for received media, what SRTP keys it will use for sending media (if
using SDES), etc. In order that the UA handle the description
properly, the initiator marks it as an offer when calling
setLocalDescription; this indicates to the UA that multiple
capabilities have been offered, but this set may be pared back later,
when the answer arrives.
Since the local user agent must be prepared to receive media upon
applying the offer, this operation will cause local decoder resources
to be allocated, based on the codecs indicated in the offer.
OffererJS->OffererUA: pc.setLocalDescription(SDP_OFFER, offer);
5.1.3. Initiating ICE
The initiator can now start the ICE process of candidate generation
and connectivity checking. This results in callbacks to the
application's IceCallback. Candidates are provided to the IceCallback
as they are allocated, with the |moreToFollow| argument set to true
if there are still allocations pending; when the last allocation
completes or times out, this callback will be invoked with
|moreToFollow| set to false.
OffererJS->OffererUA: pc.startIce();
OffererUA->OffererJS: iceCallback(candidate, ...);
5.1.4. Serializing the Offer and Candidates
At this point, the offerer is ready to send its offer to the callee
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using its preferred signaling protocol. Depending on the protocol, it
can either send the initial session description first, and then
"trickle" the ICE candidates as they are given to the application, or
it can wait for all the ICE candidates to be collected, and then send
the offer and list of candidates all at once.
5.2. Receiving the Session
Through the chosen signaling protocol, the recipient is notified of
an incoming session request. It creates a PeerConnection, and
installs its own IceCallback.
AnswererJS->AnswererUA: var pc = new PeerConnection(config, iceCb);
5.2.1. Receiving the Offer
The recipient converts the received offer from its signaling protocol
into SDP format, and supplies it to its PeerConnection, again marking
it as an offer. As a remote description, the offer indicates what
codecs the remote side wants to use for receiving, as well as what
SRTP keys it will use for sending. The setting of the remote
description causes callbacks to be issued, informing the application
of what kinds of streams are present in the offer.
This step will also cause encoder resources to be allocated, based on
the codecs specified in |offer|.
AnswererJS->AnswererUA: pc.setRemoteDescription(SDP_OFFER, offer);
AnswererUA->AnswererJS: onAddStream(stream);
5.2.2. Initiating ICE
The recipient then starts its own ICE state machine, to allow
connectivity to be established as quickly as possible.
AnswererJS->AnswererUA: pc.startIce();
AnswererUA->AnswererJS: iceCallback(candidate, ...);
5.2.3. Handling ICE Messages
If ICE candidates from the remote site were included in the offer,
the ICE Agent will automatically start trying to use them. Otherwise,
if ICE candidates are sent separately, they are passed into the
PeerConnection when they arrive.
AnswererJS->AnswererUA: pc.processIceMessage(candidate);
5.2.4. Generating the Answer
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Once the recipient has decided to accept the session, it generates an
answer session description. This process performs the appropriate
intersection of codecs and other parameters to generate the correct
answer. As with the offer, MediaHints can be provided to influence
the answer that is generated, and/or the application can post-process
the answer manually.
AnswererJS->AnswererUA: pc.createAnswer(offer, null);
5.2.5. Applying the Answer
The recipient then instructs the PeerConnection to use the answer as
its local description for this session, i.e. what codecs it will use
to receive media, etc. It also marks the description as an answer,
which tells the UA that these parameters are final. This causes the
PeerConnection to move to the ACTIVE state, and transmission of media
by the answerer to start.
AnswererJS->AnswererUA: pc.setLocalDescription(SDP_ANSWER, answer);
AnswererUA->OffererUA: <media>
5.2.6. Serializing the Answer
As with the offer, the answer (with or without candidates) is now
converted to the desired signaling format and sent to the initiator.
5.3. Completing the Session
5.3.1. Receiving the Answer
The initiator converts the answer from the signaling protocol and
applies it as the remote description, marking it as an answer. This
causes the PeerConnection to move to the ACTIVE state, and
transmission of media by the offerer to start.
OffererJS->OffererUA: pc.setRemoteDescription(SDP_ANSWER, answer);
OffererUA->AnswererUA: <media>
5.4. Updates to the Session
Updates to the session are handled with a new offer/answer exchange.
However, since media will already be flowing at this point, the new
offerer needs to support both its old session description as well as
the new one it has offered, until the change is accepted by the
remote side.
Note also that in an update scenario, the roles may be reversed, i.e.
the update offerer can be different than the original offerer.
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6. Proposed WebRTC API changes
6.1. PeerConnection API
The text below indicates the recommended changes to the
PeerConnection API to implement the JSEP functionality. Methods
marked with a [+] are new/proposed; methods marked with a [-] have
been removed in this proposal.
[Constructor (in DOMString configuration, in IceCallback iceCb)]
interface PeerConnection {
// creates a blob of SDP to be provided as an offer.
[+] SessionDescription createOffer (MediaHints hints);
// creates a blob of SDP to be provided as an answer.
[+] SessionDescription createAnswer (DOMString offer,
MediaHints hints);
// actions, for setLocalDescription/setRemoteDescription
[+] const unsigned short SDP_OFFER = 0x100;
[+] const unsigned short SDP_PRANSWER = 0x200;
[+] const unsigned short SDP_ANSWER = 0x300;
// sets the local session description
[+] void setLocalDescription (unsigned short action,
SessionDescription desc);
// sets the remote session description
[+] void setRemoteDescription (unsigned short action,
SessionDescription desc);
// returns the current local session description
[+] readonly SessionDescription localDescription;
// returns the current remote session description
[+] readonly SessionDescription remoteDescription;
[-] void processSignalingMessage (DOMString message);
const unsigned short NEW = 0; // initial state
[+] const unsigned short OPENING = 1; // local or remote desc set
const unsigned short ACTIVE = 2; // local and remote desc set
const unsigned short CLOSED = 3; // ended state
readonly attribute unsigned short readyState;
// starts ICE connection/handshaking
[+] void startIce (optional IceOptions options);
// processes received ICE information
[+] void processIceMessage (IceCandidate candidate);
const unsigned short ICE_GATHERING = 0x100;
const unsigned short ICE_WAITING = 0x200;
const unsigned short ICE_CHECKING = 0x300;
const unsigned short ICE_CONNECTED = 0x400;
const unsigned short ICE_COMPLETED = 0x500;
const unsigned short ICE_FAILED = 0x600;
const unsigned short ICE_CLOSED = 0x700;
readonly attribute unsigned short iceState;
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[-] const unsigned short SDP_IDLE = 0x1000;
[-] const unsigned short SDP_WAITING = 0x2000;
[-] const unsigned short SDP_GLARE = 0x3000;
[-] readonly attribute unsigned short sdpState;
void addStream (MediaStream stream, MediaStreamHints hints);
void removeStream (MediaStream stream);
readonly attribute MediaStream[] localStreams;
readonly attribute MediaStream[] remoteStreams;
void close ();
[ rest of interface omitted ]
};
[Constructor (in DOMString sdp)]
interface SessionDescription {
// adds the specified candidate to the description
void addCandidate(IceCandidate candidate);
// serializes the description to SDP
DOMString toSdp();
};
[Constructor (in DOMString label, in DOMString candidateLine)]
interface IceCandidate {
// the m= line this candidate is associated with
readonly DOMString label;
// creates a SDP-ized form of this candidate
DOMString toSdp();
};
6.1.1 MediaHints
MediaHints is an object that can be passed into createOffer or
createAnswer to affect the type of offer/answer that is generated.
The following properties can be set on MediaHints:
has_audio: boolean
Indicates whether we want to receive audio; defaults to true if we
have audio streams, else false
has_video: boolean
Indicates whether we want to receive video; defaults to true if we
have video streams, else false
As an example, MediaHints could be used to create a session that
transmits only audio, but is able to receive video from the remote
side, by forcing the inclusion of a m=video line even when no video
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sources are provided.
6.1.2 createOffer
The createOffer method generates a blob of SDP that contains a RFC
3264 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. The |hints| parameter may be supplied to provide additional
control over the generated offer.
As an offer, the generated SDP will contain the full set of
capabilities supported by the session (as opposed to an answer, which
will include only a specific negotiated subset to use); for each SDP
line, the generation of the SDP must follow the appropriate process
for generating an offer. In the event createOffer is called after the
session is established, createOffer will generate an offer that is
compatible with the current session, incorporating any changes that
have been made to the session since the last complete offer-answer
exchange, such as addition or removal of streams. If no changes have
been made, the offer will be identical to the current local
description.
Session descriptions generated by createOffer must be immediately
usable by setLocalDescription; if a system has limited resources
(e.g. a finite number of decoders), createOffer should return an
offer that reflects the current state of the system, so that
setLocalDescription will succeed when it attempts to acquire those
resources.
Calling this method does not change the state of the PeerConnection;
its use is not required.
A TBD exception is thrown if the |hints| parameter is malformed.
6.1.3 createAnswer
The createAnswer method generates a blob of SDP that contains a RFC
3264 SDP answer with the supported configuration for the session that
is compatible with the parameters supplied in |offer|. 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. The |hints| parameter may be supplied
to provide additional control over the generated answer.
As an answer, the generated SDP will contain a specific configuration
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that specifies how the media plane should be established. For each
SDP line, the generation of the SDP must follow the appropriate
process for generating an answer.
Session descriptions generated by createAnswer must be immediately
usable by setLocalDescription; like createOffer, the returned
description should reflect the current state of the system.
Calling this method does not change the state of the PeerConnection;
its use is not required.
A TBD exception is thrown if the |hints| parameter is malformed, or
the |offer| parameter is missing or malformed.
6.1.4 SDP_OFFER, SDP_PRANSWER, and SDP_ANSWER
The SDP_XXXX enums serve as arguments to setLocalDescription and
setRemoteDescription. They provide information as to how the
|description| parameter should be parsed, and how the media state
should be changed.
SDP_OFFER indicates that a description should be parsed as an offer;
said description may include many possible media configurations. A
description used as a SDP_OFFER may be applied anytime the
PeerConnection is in a stable state, or as an update to a previously
sent but unanswered SDP_OFFER.
SDP_PRANSWER indicates that a description should be parsed as an
answer, but not a final answer, and so should not result in the
starting of media transmission. A description used as a SDP_PRANSWER
may be applied as a response to a SDP_OFFER, or an update to a
previously sent SDP_PRANSWER.
SDP_ANSWER indicates that a description should be parsed as an
answer, and the offer-answer exchange should be considered complete.
A description used as a SDP_ANSWER may be applied as a response to a
SDP_OFFER, or an update to a previously send SDP_PRANSWER.
6.1.5 setLocalDescription
The setLocalDescription method instructs the PeerConnection to apply
the supplied SDP blob as its local configuration. The |type|
parameter indicates whether the blob should be processed as an offer
(SDP_OFFER), provisional answer (SDP_PRANSWER), or final answer
(SDP_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
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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
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.
Changes to the state of media transmission will only occur when a
final answer is successfully applied.
A TBD exception is thrown if |description| is invalid. A TBD
exception is thrown if there are insufficient local resources to
apply |description|.
6.1.6 setRemoteDescription
The setRemoteDescription method instructs the PeerConnection to apply
the supplied SDP blob as the desired remote configuration. As in
setLocalDescription, the |type| parameter 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.
Changes to the state of media transmission will only occur when a
final answer is successfully applied.
A TBD exception is thrown if |description| is invalid. A TBD
exception is thrown if there are insufficient local resources to
apply |description|.
6.1.7 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.
A null object will be returned if the local description has not yet
been established.
6.1.8 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
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supplied via processIceMessage.
A null object will be returned if the remote description has not yet
been established.
6.1.9 IceOptions
IceOptions is an object that can be passed into startIce to restrict
the candidates that are provided to the application and used for
connectivity checks. This can be useful if the application wants to
only use TURN candidates for privacy reasons, or only local + STUN
candidates for cost reasons.
The following properties can be set on IceOptions:
use_candidates: "all", "no_relay", "only_relay"
Indicates what types of local candidates should be used; defaults
to "all"
6.1.10 startIce
The startIce method starts or updates the ICE Agent process of
gathering local candidates and pinging remote candidates. The
|options| argument can be used to restrict which types of local
candidates are provided to the application and used for pinging; this
can be used to limit the use of TURN candidates by a callee to avoid
leaking location information prior to the call being accepted.
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.
A TBD exception will be thrown if |options| is malformed.
6.1.11 processIceMessage
The processIceMessage method provides a remote candidate to the ICE
Agent, which will be added to the remote description. If startIce has
been called, connectivity checks will be sent to the new candidates.
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.
A TBD exception will be thrown if |candidate| is missing or
malformed.
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7. 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.
7.1. Call using ROAP
This example demonstrates a ROAP call, without the use of trickle
candidates.
// Call is initiated toward Answerer
OffererJS->OffererUA: pc = new PeerConnection();
OffererJS->OffererUA: pc.addStream(localStream, null);
OffererJS->OffererUA: pc.startIce();
OffererUA->OffererJS: iceCallback(candidate, false);
OffererJS->OffererUA: offer = pc.createOffer(null);
OffererJS->OffererUA: pc.setLocalDescription(SDP_OFFER, offer.toSdp());
OffererJS->AnswererJS: {"type":"OFFER", "sdp":"<offer>"}
// OFFER arrives at Answerer
AnswererJS->AnswererUA: pc = new PeerConnection();
AnswererJS->AnswererUA: pc.setRemoteDescription(SDP_OFFER, msg.sdp);
AnswererUA->AnswererJS: onaddstream(remoteStream);
AnswererJS->AnswererUA: pc.startIce();
AnswererUA->OffererUA: iceCallback(candidate, false);
// Answerer accepts call
AnswererJS->AnswererUA: peer.addStream(localStream, null);
AnswererJS->AnswererUA: answer = peer.createAnswer(msg.offer, null);
AnswererJS->AnswererUA: peer.setLocalDescription(SDP_ANSWER, answer);
AnswererJS->OffererJS: {"type":"ANSWER","sdp":"<answer>"}
// ANSWER arrives at Offerer
OffererJS->OffererUA: peer.setRemoteDescription(ANSWER, answer);
OffererUA->OffererJS: onaddstream(remoteStream);
// ICE Completes (at Answerer)
AnswererUA->AnswererJS: onopen();
AnswererUA->OffererUA: Media
// ICE Completes (at Offerer)
OffererUA->OffererJS: onopen();
OffererJS->AnswererJS: {"type":"OK" }
OffererUA->AnswererUA: Media
7.2. Call using XMPP
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This example demonstrates an XMPP call, making use of trickle
candidates.
// 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(SDP_OFFER, offer);
OffererJS: xmpp = createSessionInitiate(offer);
OffererJS->AnswererJS: <jingle action="session-initiate"/>
OffererJS->OffererUA: pc.startIce();
OffererUA->OffererJS: iceCallback(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(SDP_OFFER, offer);
AnswererUA->AnswererJS: onaddstream(remoteStream);
// transport-infos arrive at Answerer
AnswererJS->AnswererUA: candidates = parseTransportInfo(xmpp);
AnswererJS->AnswererUA: pc.processIceMessage(candidates);
AnswererJS->AnswererUA: pc.startIce();
AnswererUA->AnswererJS: iceCallback(cand, ...)
AnswererJS: createTransportInfo(cand);
AnswererJS->OffererJS: <jingle action="transport-info"/>
// transport-infos arrive at Offerer
OffererJS->OffererUA: candidates = parseTransportInfo(xmpp);
OffererJS->OffererUA: pc.processIceMessage(candidates);
// Answerer accepts call
AnswererJS->AnswererUA: peer.addStream(localStream, null);
AnswererJS->AnswererUA: answer = peer.createAnswer(offer, null);
AnswererJS: xmpp = createSessionAccept(answer);
AnswererJS->AnswererUA: pc.setLocalDescription(SDP_ANSWER, answer);
AnswererJS->OffererJS: <jingle action="session-accept"/>
// session-accept arrives at Offerer
OffererJS: answer = parseSessionAccept(xmpp);
OffererJS->OffererUA: peer.setRemoteDescription(ANSWER, answer);
OffererUA->OffererJS: onaddstream(remoteStream);
// ICE Completes (at Answerer)
AnswererUA->AnswererJS: onopen();
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AnswererUA->OffererUA: Media
// ICE Completes (at Offerer)
OffererUA->OffererJS: onopen();
OffererUA->AnswererUA: Media
7.3. 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(SDP_OFFER, offer);
OffererJS->AnswererJS: <jingle action="content-add"/>
// content-add arrives at Answerer
AnswererJS: offer = parseContentAdd(xmpp);
AnswererJS->AnswererUA: pc.setRemoteDescription(SDP_OFFER, offer);
AnswererJS->AnswererUA: answer = pc.createAnswer(offer, null);
AnswererJS->AnswererUA: pc.setLocalDescription(SDP_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(SDP_ANSWER, answer);
7.4. 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.
// 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(SDP_OFFER, offer);
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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(SDP_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(SDP_ANSWER, offer);
// ack arrives at "Answerer"; remote offer is used as an answer
AnswererJS->AnswererUA: pc.setRemoteDescription(SDP_ANSWER, offer);
7.5. Call using SIP
This example demonstrates a simple SIP call (e.g. where the client
talks to a SIP proxy over WebSockets).
// Call is initiated toward Answerer
OffererJS->OffererUA: pc = new PeerConnection();
OffererJS->OffererUA: pc.addStream(localStream, null);
OffererJS->OffererUA: pc.startIce();
OffererUA->OffererJS: iceCallback(candidate, false);
OffererJS->OffererUA: offer = pc.createOffer(null);
OffererJS->OffererUA: pc.setLocalDescription(SDP_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(SDP_OFFER, offer);
AnswererUA->AnswererJS: onaddstream(remoteStream);
AnswererJS->AnswererUA: pc.startIce();
AnswererUA->OffererUA: iceCallback(candidate, false);
// Answerer accepts call
AnswererJS->AnswererUA: peer.addStream(localStream, null);
AnswererJS->AnswererUA: answer = peer.createAnswer(offer, null);
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AnswererJS: sip = createResponse(200, answer);
AnswererJS->AnswererUA: peer.setLocalDescription(SDP_ANSWER, answer);
AnswererJS->OffererJS: 200 OK w/ SDP
// 200 OK arrives at Offerer
OffererJS: answer = parseResponse(sip);
OffererJS->OffererUA: peer.setRemoteDescription(ANSWER, answer);
OffererUA->OffererJS: onaddstream(remoteStream);
OffererJS->AnswererJS: ACK
// ICE Completes (at Answerer)
AnswererUA->AnswererJS: onopen();
AnswererUA->OffererUA: Media
// ICE Completes (at Offerer)
OffererUA->OffererJS: onopen();
OffererUA->AnswererUA: Media
7.6. Handling early media (e.g. 1-800-FEDEX), using SIP
This example demonstrates how early media could be handled; for
simplicity, only the offerer side of the call is shown.
// Call is initiated toward Answerer
OffererJS->OffererUA: pc = new PeerConnection();
OffererJS->OffererUA: pc.addStream(localStream, null);
OffererJS->OffererUA: pc.startIce();
OffererUA->OffererJS: iceCallback(candidate, false);
OffererJS->OffererUA: offer = pc.createOffer(null);
OffererJS->OffererUA: pc.setLocalDescription(SDP_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(SDP_PRANSWER, answer);
OffererUA->OffererJS: onaddstream(remoteStream);
// ICE Completes (at Offerer)
OffererUA->OffererJS: onopen();
OffererUA->AnswererUA: Media
// 200 OK arrives at Offerer
OffererJS: answer = parseResponse(sip);
OffererJS->OffererUA: pc.setRemoteDescription(SDP_ANSWER, answer);
OffererJS->AnswererJS: ACK
8. Example Application
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The following example demonstrates a simple video calling
application, roughly corresponding to the flow in Example 7.1.
var signalingChannel = createSignalingChannel();
var pc = null;
var hasCandidates = false;
function start(isCaller) {
// create a PeerConnection and hook up the IceCallback
pc = new webkitPeerConnection(
"", function (candidate, moreToFollow) {
if (!moreToFollow) {
hasCandidates = true;
maybeSignal(isCaller);
}
});
// get the local stream and show it in the local video element
navigator.webkitGetUserMedia(
{"audio": true, "video": true}, function (localStream) {
selfView.src = webkitURL.createObjectURL(localStream);
pc.addStream(localStream);
maybeSignal(isCaller);
}
// once remote stream arrives, show it in the remote video element
pc.onaddstream = function(evt) {
remoteView.src = webkitURL.createObjectURL(evt.stream);
};
// if we're the caller, create and install our offer,
// and start candidate generation
if (isCaller) {
offer = pc.createOffer(null);
pc.setLocalDescription(SDP_OFFER, offer);
pc.startIce();
}
}
function maybeSignal(isCaller) {
// only signal once we have a local stream and local candidates
if (localStreams.size() == 0 || !hasCandidates) return;
if (isCaller) {
offer = pc.localDescription;
signalingChannel.send(
JSON.stringify({ "type": "offer", "sdp": offer }));
} else {
// if we're the callee, generate, apply, and send the answer
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answer = pc.createAnswer(pc.remoteDescription, null);
pc.setLocalDescription(SDP_ANSWER, answer);
signalingChannel.send(
JSON.stringify({ "type": "answer", "sdp": answer }));
}
}
signalingChannel.onmessage = function(evt) {
var msg = JSON.parse(evt.data);
if (msg.type == "offer") {
// create the PeerConnection
start(false);
// feed the received offer into the PeerConnection and
// start candidate generation
pc.setRemoteDescription(PeerConnection.SDP_OFFER, msg.sdp);
pc.startIce();
} else if (msg.type == "answer") {
// feed the answer into the PeerConnection to complete setup
pc.setRemoteDescription(PeerConnection.SDP_ANSWER, msg.sdp);
}
9. Security Considerations
TODO
10. IANA Considerations
This document requires no actions from IANA.
11. Acknowledgements
Harald Alvestrand, Dan Burnett, Neil Stratford, Eric Rescorla, and
Anant Narayanan all provided valuable feedback on this proposal.
Matthew Kaufman provided the observation that keeping state out of
the browser allows a call to continue even if the page is reloaded.
Adam Bergvist provided a code example that served as the basis for
the example in Section 8.
12. References
12.1. Normative References
[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
Requirement Levels", BCP 14, RFC 2119, March 1997.
[RFC3264] Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model
with Session Description Protocol (SDP)", RFC 3264, June 2002.
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[RFC4566] Handley, M., Jacobson, V., and C. Perkins, "SDP: Session
Description Protocol", RFC 4566, July 2006.
12.2. Informative References
[RFC4568] Andreasen, F., Baugher, M., and D. Wing, "Session
Description Protocol (SDP) Security Descriptions for Media Streams",
RFC 4568, July 2006.
[RFC5245] Rosenberg, J., "Interactive Connectivity Establishment
(ICE): A Protocol for Network Address Translator (NAT) Traversal for
Offer/Answer Protocols", RFC 5245, April 2010.
[webrtc-api] Bergkvist, Burnett, Jennings, Narayanan, "WebRTC 1.0:
Real-time Communication Between Browsers", October 2011.
Available at http://dev.w3.org/2011/webrtc/editor/webrtc.html
Appendix A. Open Issues
- Determine list of exceptions that can be thrown by each method.
Leaning toward something like a PCException, a la
https://developer.mozilla.org/en/IndexedDB/IDBDatabaseException
- Need callback to indicate that the transport is down, e.g.
ICE_DISCONNECTED or ondisconnected().
Appendix B. Change log
00: Migrated from draft-uberti-rtcweb-jsep-02.
Authors' Addresses
Justin Uberti
Google
5 Cambridge Center
Cambridge, MA 02142
Email: justin@uberti.name
Cullen Jennings
Cisco
170 West Tasman Drive
San Jose, CA 95134
USA
Email: fluffy@cisco.com
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