Network Working Group J. Uberti
Internet-Draft Google
Intended status: Standards Track C. Jennings
Expires: December 6, 2012 Cisco Systems, Inc.
June 4, 2012
Javascript Session Establishment Protocol
draft-ietf-rtcweb-jsep-01
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
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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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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
1.1. 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 and State Machine . . . . . . . . . . 7
4.3. Session Description Format . . . . . . . . . . . . . . . . 9
4.4. Separation of Signaling and ICE State Machines . . . . . . 10
4.5. ICE Candidate Trickling . . . . . . . . . . . . . . . . . . 10
4.6. ICE Candidate Format . . . . . . . . . . . . . . . . . . . 11
4.7. Interactions With Forking . . . . . . . . . . . . . . . . . 11
4.7.1. Serial Forking . . . . . . . . . . . . . . . . . . . . 11
4.7.2. Parallel Forking . . . . . . . . . . . . . . . . . . . 12
4.8. Session Rehydration . . . . . . . . . . . . . . . . . . . . 12
5. Interface . . . . . . . . . . . . . . . . . . . . . . . . . . . 13
5.1. Methods . . . . . . . . . . . . . . . . . . . . . . . . . . 13
5.1.1. createOffer . . . . . . . . . . . . . . . . . . . . . . 13
5.1.2. createAnswer . . . . . . . . . . . . . . . . . . . . . 14
5.1.3. SessionDescriptionType . . . . . . . . . . . . . . . . 14
5.1.4. setLocalDescription . . . . . . . . . . . . . . . . . . 15
5.1.5. setRemoteDescription . . . . . . . . . . . . . . . . . 15
5.1.6. localDescription . . . . . . . . . . . . . . . . . . . 16
5.1.7. remoteDescription . . . . . . . . . . . . . . . . . . . 16
5.1.8. updateIce . . . . . . . . . . . . . . . . . . . . . . . 16
5.1.9. addIceCandidate . . . . . . . . . . . . . . . . . . . . 17
5.2. Configurable SDP Parameters . . . . . . . . . . . . . . . . 17
6. Media Setup Overview . . . . . . . . . . . . . . . . . . . . . 17
6.1. Initiating the Session . . . . . . . . . . . . . . . . . . 18
6.1.1. Generating An Offer . . . . . . . . . . . . . . . . . . 18
6.1.2. Applying the Offer . . . . . . . . . . . . . . . . . . 18
6.1.3. Handling ICE Callbacks . . . . . . . . . . . . . . . . 18
6.1.4. Serializing the Offer and Candidates . . . . . . . . . 19
6.2. Receiving the Session . . . . . . . . . . . . . . . . . . . 19
6.2.1. Receiving the Offer . . . . . . . . . . . . . . . . . . 19
6.2.2. Handling ICE Messages . . . . . . . . . . . . . . . . . 19
6.2.3. Generating the Answer . . . . . . . . . . . . . . . . . 20
6.2.4. Applying the Answer . . . . . . . . . . . . . . . . . . 20
6.2.5. Serializing the Answer . . . . . . . . . . . . . . . . 20
6.3. Completing the Session . . . . . . . . . . . . . . . . . . 20
6.3.1. Receiving the Answer . . . . . . . . . . . . . . . . . 20
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6.4. Updates to the Session . . . . . . . . . . . . . . . . . . 20
7. Security Considerations . . . . . . . . . . . . . . . . . . . . 21
8. IANA Considerations . . . . . . . . . . . . . . . . . . . . . . 21
9. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 21
10. References . . . . . . . . . . . . . . . . . . . . . . . . . . 21
10.1. Normative References . . . . . . . . . . . . . . . . . . . 21
10.2. Informative References . . . . . . . . . . . . . . . . . . 21
Appendix A. JSEP Implementation Examples . . . . . . . . . . . . . 22
A.1. Example API . . . . . . . . . . . . . . . . . . . . . . . . 22
A.2. Example API Flows . . . . . . . . . . . . . . . . . . . . . 23
A.2.1. Call using ROAP . . . . . . . . . . . . . . . . . . . . 23
A.2.2 Call using XMPP . . . . . . . . . . . . . . . . . . . . 24
A.2.3. Adding video to a call, using XMPP . . . . . . . . . . 25
A.2.4. Simultaneous add of video streams, using XMPP . . . . . 26
A.2.5. Call using SIP . . . . . . . . . . . . . . . . . . . . 27
A.2.6. Handling early media (e.g. 1-800-FEDEX), using SIP . . 28
A.3. Full Example Application . . . . . . . . . . . . . . . . . 28
Appendix B. Change log . . . . . . . . . . . . . . . . . . . . . . 30
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 30
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1. Introduction
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.
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 [I-D.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.
1.1. 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 the issues mentioned above, 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. 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, JSEP decouples the ICE state machine from the overall
signaling state machine, as the ICE state machine must remain in the
browser, since only the browser has the necessary knowledge of
candidates and other transport info. 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 descriptions 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. 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
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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. Exposing
createOffer/createAnswer avoids that problem, but still allows
applications to generate their own offers/answers if they choose,
using the description generated by createOffer as an indication of
the browser's capabilities.
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 session. 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.
+-----------+ +-----------+
| Web App |<--- App-Specific Signaling --->| Web App |
+-----------+ +-----------+
| |
| SDP | SDP
V V
+-----------+ +-----------+
| Browser |<----------- Media ------------>| Browser |
+-----------+ +-----------+
Figure 1: JSEP Signaling Model
4.2. 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
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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,
and what the remote party should send. 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
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, 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 a setLocalDescription and a
setRemoteDescription method, and both these methods take a parameter
to indicate the type of session description being supplied. This
satisfies the requirements listed above for both the offerer, who
first calls setLocalDescription("offer", sdp) and then later
setRemoteDescription("answer", sdp), as well as for the answerer, who
first calls setRemoteDescription("offer", sdp) and then later
setLocalDescription("answer", sdp). While it could be possible to
implicitly determine the value of the offer/answer argument,
requiring it to be specified explicitly is more robust, allowing
invalid combinations (i.e. an answer before an offer) to generate an
appropriate error.
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It also allows for an answer to be treated as provisional by the
application. Provisional answers provide a way for an answerer to
communicate session parameters back to the offerer, in order for the
session to begin, while allowing a final answer to be specified
later. This concept of a final answer is important to the
offer/answer model; when such an answer is received, any extra
resources allocated by the caller can be released, now that the exact
session configuration is known. These "resources" can include things
like extra ICE components, TURN candidates, or video decoders.
Provisional answers, on the other hand, do no such deallocation; as a
result, multiple dissimilar provisional answers can be received and
applied during call setup.
As in [RFC3264], an offerer can send an offer, and update it as long
as it has not been answered. 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:
+-----------+
| |
| |
| Stable |<---------------\
| | |
| | |
+-----------+ |
^ | |
| | OFFER |
ANSWER | | | ANSWER
| V |
+-----------+ +-----------+
| | | |
| | PRANSWER | |
| Offer |--------->| Pranswer |
| | | |
| |----\ | |----\
+-----------+ | +-----------+ |
^ | ^ |
| | | |
\-----/ \-----/
OFFER 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.
4.3. Session Description Format
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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
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
JSEP does away with the SDP Agent within the browser, and this
functionality is now controlled directly by the application, which
uses the setLocalDescription and setRemoteDescription APIs to tell
the browser what SDP has been negotiated. The ICE Agent remains in
the browser, as it still needs to drive the process of gathering
candidates, connectivity checks, and related ICE functionality.
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.
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.
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 the
onicecandidate callback with a new candidate; applications that do
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not support this feature can simply wait for the final onicecandidate
callback that indicates gathering is complete, and create and send
their offer, with all the candidates, at this time.
Upon receipt of trickled candidates, the receiving application can
supply them to its ICE Agent by calling an addIceCandidate method.
This triggers the ICE Agent to start using this remote candidate for
connectivity checks. Applications that do not make use of candidate
tricking can ignore addIceCandidate entirely, and use the
onicecandidate callback 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 fields 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 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).
4.7. Interactions With Forking
4.7.1. Serial Forking
Serial 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 serial 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 send a terminate message to
any subsequent answers. In SIP parlance, this would be ACK + BYE.
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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 existing
remote description as a final answer.
4.7.2. Parallel Forking
Parallel forking involves a call being dispatched to multiple remote
callees, where each callee can accept the call, and multiple
simultaneous active sessions can be established as a result. If
multiple callees send media, this media is mixed and played out at
the caller side.
JSEP can handle parallel forking by "cloning" the session when needed
to create multiple parallel sessions. When the first answer is
received, the caller can clone the existing session, and then apply
the answer as a final answer to the original session. Upon receiving
the next answer, the cloned session is cloned again, and the received
answer is applied as a final answer to the first clone. This process
repeats until the caller decides to end the setup flow, and closes
the final cloned session.
Cloned sessions inherit the local session description and candidates
from their parent, and an empty remote description; only sessions
that have not yet applied an answer can be cloned. Each cloned
session may discover new peer-reflexive candidates; these candidates
will be supplied via the onicecandidate callback to that specific
session. Since the clone uses the same local description as its
parent, creating a clone will fail if it is not possible to reserve
the same resources for the clone as have already been reserved by the
parent.
As a result of this cloning, the application will end up with N
parallel 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.
4.8. Session Rehydration
In the event that the local application state is reinitialized,
either due to a user reload of the page, or a decision within the
application to reload itself (perhaps to update to a new version), it
is possible to keep an existing session alive via a process called
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"rehydration".
With rehydration, the current local session description is persisted
somewhere outside of the page, perhaps on the application server, or
in browser local storage. The page is then reloaded, and a new
session object is created in Javascript. The saved local session is
now retrieved, but the previous ICE candidates will no longer be
valid in this case, so we will need to perform an ICE restart; to do
so, we simply generate a new ICE ufrag/pwd combo for the local
description.
The modified local description is then installed via
setLocalDescription, and sent off as an offer to the remote side, who
will reply with an answer that can be supplied to
setRemoteDescription. ICE processing proceeds as usual, and as soon
as connectivity is established, the session will be back up and
running again.
5. 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.
5.1. Methods
5.1.1. 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. A constraints parameters may be supplied to provide additional
control over the generated offer, e.g. to get a full set of session
capabilities, or to request a new set of ICE credentials.
In the initial offer, the generated SDP will contain all desired
functionality for the session (certain parts that are supported but
not desired by default may be omitted); 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
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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. 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 does not change state; its use is not required.
5.1.2. 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. A constraints 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 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. 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 does not change state; its use is not required.
5.1.3. SessionDescriptionType
The strings "offer", "pranswer", and "answer" serve as type 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.
"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
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PeerConnection is in a stable state, or as an update to a previously
sent 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 "answer".
"answer" indicates that a description should be parsed as an answer,
the offer-answer exchange should be considered complete, and any
resources (decoders, candidates) that are no longer needed can be
released. A description used as an "answer" may be applied as a
response to a "offer", or an update to a previously sent "pranswer".
The application can use some discretion on whether an answer should
be applied as provisional or final. For example, in a serial forking
scenario, an application may receive multiple "final" answers, one
from each remote endpoint. The application could 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).
5.1.4. 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,
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
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.
If setRemoteDescription was previous called with an offer, and
setLocalDescription is called with an answer (provisional or final),
and the media directions are compatible, this will result in the
starting of media transmission.
5.1.5. setRemoteDescription
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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.
If setRemoteDescription was previous called with an offer, and
setLocalDescription is called with an answer (provisional or final),
and the media directions are compatible, this will result in the
starting of media transmission.
5.1.6. 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.
5.1.7. 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.
A null object will be returned if the remote description has not yet
been established.
5.1.8. 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
onicecallback or used for connectivity checks.
This call may result in a change to the state of the ICE Agent, and
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may result in a change to media state if it results in connectivity
being established.
5.1.9. addIceCandidate
The addIceCandidate method provides a remote candidate to the ICE
Agent, which will be added to the remote description. 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.
5.2. Configurable SDP Parameters
The following is a partial list of SDP parameters that an application
may want to control, in either local or remote descriptions, using
this API.
- remove or reorder codecs (m=)
- change codec attributes (a=fmtp; ptime)
- enable/disable BUNDLE (a=group)
- enable/disable RTCP mux (a=rtcp-mux)
- remove or reorder SRTP crypto-suites (a=crypto)
- change SRTP parameters or keys (a=crypto)
- change send resolution or framerate (TBD)
- change desired recv resolution or framerate (TBD)
- change total bandwidth (b=)
- remove desired AVPF mechanisms (a=rtcp-fb)
- remove RTP header extensions (a=rtphdr-ext)
- add/change SSRC grouping (e.g. FID, RTX, etc) (a=ssrc-group)
- add SSRC attributes (a=ssrc)
- change ICE ufrag/password (a=ice-ufrag/pwd)
- change media send/recv state (a=sendonly/recvonly/inactive)
For example, an application could implement call hold by adding an
a=inactive attribute to its local description, and then applying and
signaling that description.
6. Media Setup Overview
The example here shows a typical call setup using the JSEP model,
indicating the functions that are called and the state changes that
occur. 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
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6.1. Initiating the Session
The initiator creates a PeerConnection, hooks up to its ICE callback,
and adds the desired MediaStreams (presumably obtained via
getUserMedia). The ICE gathering process begins to gather candidates
for a default number of streams, as the exact number will not be
known until the local description is applied. The PeerConnection is
in the NEW state.
OffererJS->OffererUA: var pc = new PeerConnection(config, null);
OffererJS->OffererUA: pc.onicecandidate = onIceCandidate;
OffererJS->OffererUA: pc.addStream(stream);
6.1.1. 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
MediaConstraints 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);
6.1.2. 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("offer", offer);
6.1.3. Handling ICE Callbacks
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The initiator starts to receive callbacks on its onicecandidate
handler. Candidates are provided to the IceCallback as they are
allocated; when the last allocation completes or times out, this
callback will be invoked with a null argument.
OffererUA->OffererJS: onIceCandidate(candidate);
6.1.4. Serializing the Offer and Candidates
At this point, the offerer is ready to send its offer to the callee
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.
6.2. Receiving the Session
Through the chosen signaling protocol, the recipient is notified of
an incoming session request. It creates a PeerConnection, and sets up
its own ICE callback. The ICE gathering process begins to gather
candidates for a default number of streams.
AnswererJS->AnswererUA: var pc = new PeerConnection(config, null);
AnswererJS->AnswererUA: pc.onicecandidate = onIceCandidate;
6.2.1. 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("offer", offer);
AnswererUA->AnswererJS: onAddStream(stream);
6.2.2. 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.
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AnswererJS->AnswererUA: pc.addIceCandidate(candidate);
6.2.3. Generating the Answer
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, MediaConstraints 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);
6.2.4. 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 (assuming both sides have indicated this in
their descriptions).
AnswererJS->AnswererUA: pc.setLocalDescription("answer", answer);
AnswererUA->OffererUA: <media>
6.2.5. 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.
6.3. Completing the Session
6.3.1. 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 (assuming both sides
have indicated this in their descriptions).
OffererJS->OffererUA: pc.setRemoteDescription("answer", answer);
OffererUA->AnswererUA: <media>
6.4. 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
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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.
7. Security Considerations
TODO
8. IANA Considerations
This document requires no actions from IANA.
9. Acknowledgements
Harald Alvestrand, Dan Burnett, Neil Stratford, Eric Rescorla, Anant
Narayanan, and Adam Bergkvist 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. Richard Ejzak provided the specifics on session cloning.
10. References
10.1. 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.
[RFC4566] Handley, M., Jacobson, V., and C. Perkins, "SDP: Session
Description Protocol", RFC 4566, July 2006.
10.2. 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:
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Real-time Communication Between Browsers", May 2011.
Available at http://dev.w3.org/2012/webrtc/editor/webrtc.html
Appendix A. JSEP Implementation Examples
A.1. Example API
The interface below shows a basic Javascript API that could be used
to expose the functionality discussed in this document. This API is
used for the examples in the following parts of this Appendix.
// actions, for setLocalDescription/setRemoteDescription
enum SessionDescriptionType { "offer", "pranswer", "answer" }
// constraints that can be supplied to the ctor or createXXXX
enum MediaConstraints {
"offerConfig", // controls the kind of offer created;
// "default" (normal offer)
// "caps" (all capabilities)
// "new" (brand new description)
// "iceRestart" (new ICE creds)
"iceTransports", // controls ICE candidates; can be
// "none" (no candidates)
// "relay" (only relay candidates)
// "all" (all available candidates)
}
[Constructor (int index, DOMString id, in DOMString candidateLine)]
interface IceCandidate {
// the m= line index for this candidate
readonly attribute int mLineIndex
// the mid for the m= line for this candidate
readonly attribute DOMString mLineId;
// creates a SDP-ized form of this candidate
stringifier DOMString ();
};
[Constructor (DOMString sdp)]
interface SessionDescription {
// adds the specified candidate to the description
void addCandidate(IceCandidate candidate);
// serializes the description to SDP
stringifier DOMString ();
};
[Constructor (DOMString configuration,
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optional MediaConstraints constraints)]
interface PeerConnection {
// creates a blob of SDP to be provided as an offer.
SessionDescription createOffer (
SessionDescriptionCallback successCb,
optional ErrorCallback errorCb,
optional MediaContraints constraints);
// creates a blob of SDP to be provided as an answer.
SessionDescription createAnswer (
SessionDescription offer,
SessionDescriptionCallback successCb,
optional ErrorCallback errorCb,
optional MediaContraints constraints);
// sets the local session description
void setLocalDescription (
SessionDescriptionType action,
SessionDescription desc);
// sets the remote session description
void setRemoteDescription (
SessionDescriptionType action,
SessionDescription desc)
// returns the current local session description
readonly attribute SessionDescription localDescription;
// returns the current remote session description
readonly attribute SessionDescription remoteDescription;
// updates the constraints for ICE processing
void updateIce (
optional DOMString configuration,
optional MediaConstraints constraints);
// starts using a received remote ICE candidate
void addIceCandidate (
IceCandidate candidate);
// notifies the application of a new local ICE candidate
attribute Function? onicecandidate;
};
A.2. 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.2.1. Call using ROAP
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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);
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: peer.addStream(localStream, null);
AnswererJS->AnswererUA: answer = peer.createAnswer(msg.sdp, null);
AnswererJS->AnswererUA: peer.setLocalDescription("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
A.2.2 Call using XMPP
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("offer", offer);
OffererJS: xmpp = createSessionInitiate(offer);
OffererJS->AnswererJS: <jingle action="session-initiate"/>
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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: peer.addStream(localStream, null);
AnswererJS->AnswererUA: answer = peer.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: 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();
OffererUA->AnswererUA: Media
A.2.3. 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.
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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.2.4. 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("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
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// 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.2.5. 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);
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: peer.addStream(localStream, null);
AnswererJS->AnswererUA: answer = peer.createAnswer(offer, null);
AnswererJS: sip = createResponse(200, answer);
AnswererJS->AnswererUA: peer.setLocalDescription("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
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// ICE Completes (at Offerer)
OffererUA->OffererJS: onopen();
OffererUA->AnswererUA: Media
A.2.6. 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);
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->OffererJS: onopen();
OffererUA->AnswererUA: Media
// 200 OK arrives at Offerer
OffererJS: answer = parseResponse(sip);
OffererJS->OffererUA: pc.setRemoteDescription("answer", answer);
OffererJS->AnswererJS: ACK
A.3. Full Example Application
The following example demonstrates a simple video calling
application, using both trickle candidates and provisional answers to
speed up call setup.
// Usage:
// Caller calls start(true)
// Callee calls start(false) to prepare the call/start connecting,
// and then accept() to start transmitting.
var signalingChannel = createSignalingChannel();
var pc = null;
var localStream = null;
signalingChannel.onmessage = handleMessage;
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// Set up the call, get access to local media,
// and establish connectivity.
function start(isCaller) {
// Create a PeerConnection and hook up the IceCallback.
pc = new webkitPeerConnection(null, null);
pc.onicecandidate = function(evt) {
sendMessage("candidate", evt.candidate);
};
// Get the local stream and show it in the local video element;
// if we're the caller, ship off an offer once we get the stream.
navigator.webkitGetUserMedia(
{"audio": true, "video": true}, function (stream) {
selfView.src = webkitURL.createObjectURL(stream);
localStream = stream;
if (isCaller) {
pc.addStream(stream);
pc.createOffer(function(sdp) {
setLocalAndSendMessage("offer", sdp);
});
});
// When the remote stream arrives, show it in the remote
// video element.
pc.onaddstream = function(evt) {
remoteView.src = webkitURL.createObjectURL(evt.stream);
};
}
// The callee has accepted the call, attach their media
// and send a final answer.
function accept() {
// The addStream could also be done for the pranswer,
// although that would delay the pranswer
// (due to the need for user consent)
pc.addStream(localStream); // assumes we have the stream already
pc.createAnswer(msg.sdp, function(sdp) {
setLocalAndSendMessage("answer", sdp);
});
}
// -- internal methods --
// Apply SDP locally and send it to the remote side.
function setLocalAndSendMessage(type, sdp) {
pc.setLocalDescription(type, sdp);
sendMessage(type, sdp);
}
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// Send a signaling message to the remote side.
function sendMessage(type, obj) {
signalingChannel.send(
JSON.stringify({ "type": type, "sdp": obj }));
}
// Handle incoming signaling messages.
function handleMessage(str) {
var msg = JSON.parse(str);
switch (msg.type) {
case "offer":
// create the PeerConnection
start(false);
// feed the received offer into the PeerConnection
pc.setRemoteDescription(msg.type, msg.sdp);
// create provisional answer to allow ICE/DTLS to start
pc.createAnswer(msg.sdp, function(sdp) {
setDirection(sdp, "recvonly");
setLocalAndSendMessage("pranswer", sdp);
});
break;
case "pranswer":
case "answer":
pc.setRemoteDescription(msg.type, msg.sdp);
break;
case "candidate":
pc.addIceCandidate(msg.sdp);
break;
}
}
Appendix B. Change log
01: Added diagrams for architecture and state machine.
Added sections on forking and rehydration.
Clarified meaning of "pranswer" and "answer".
Reworked how ICE restarts and media directions are controlled.
Added list of parameters that can be changed in a description.
Updated suggested API and examples to match latest thinking.
Suggested API and examples have been moved to an appendix.
00: Migrated from draft-uberti-rtcweb-jsep-02.
Authors' Addresses
Justin Uberti
Google
5 Cambridge Center
Cambridge, MA 02142
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Email: justin@uberti.name
Cullen Jennings
Cisco
170 West Tasman Drive
San Jose, CA 95134
USA
Email: fluffy@cisco.com
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