wdiff draft-ietf-rtcweb-jsep-01.txt draft-ietf-rtcweb-jsep-02.txt

Network Working Group J. Uberti
Internet-Draft Google
Intended status: Standards Track C. Jennings
Expires: December 6, 2012 April 25, 2013 Cisco Systems, Inc.
June 4,
October 22, 2012

Javascript Session Establishment Protocol
draft-ietf-rtcweb-jsep-01
draft-ietf-rtcweb-jsep-02

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.

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This Internet-Draft will expire on July 26, 2012. April 25, 2013.

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Table of Contents

1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . . 4
1.1. Terminology . . . . . . . . . . . . . . . . . . . . . . . . 5 3
2. JSEP Approach . Other Approaches Considered . . . . . . . . . . . . . . . . . 5
3. Terminology . . . . . . . 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.
4.4.1. ICE Candidate Format Trickling . . . . . . . . . . . . . . . 10
4.4.1.1. ICE Candidate Format . . . . . 11
4.7. . . . . . . . . . . 10
4.5. Interactions With Forking . . . . . . . . . . . . . . . . . 11
4.7.1. Serial
4.5.1. Sequential Forking . . . . . . . . . . . . . . . . . . . . 11
4.7.2.
4.5.2. Parallel Forking . . . . . . . . . . . . . . . . . . . 12
4.8.
4.6. Session Rehydration . . . . . . . . . . . . . . . . . . . . 12 13
5. Interface . . . . . . . . . . . . . . . . . . . . . . . . . . . 13 14
5.1. Methods SDP Requirements . . . . . . . . . . . . . . . . . . . . . 14
5.2. Methods . . . . . 13
5.1.1. createOffer . . . . . . . . . . . . . . . . . . . . 15
5.2.1. createOffer . . 13
5.1.2. createAnswer . . . . . . . . . . . . . . . . . . . 15
5.2.2. createAnswer . . 14
5.1.3. SessionDescriptionType . . . . . . . . . . . . . . . . 14
5.1.4. setLocalDescription . . . 15
5.2.3. SessionDescriptionType . . . . . . . . . . . . . . . 15
5.1.5. setRemoteDescription . 16
5.2.3.1. Creating Answers . . . . . . . . . . . . . . . . 15
5.1.6. localDescription . 17
5.2.4. setLocalDescription . . . . . . . . . . . . . . . . . 17
5.2.5. setRemoteDescription . 16
5.1.7. remoteDescription . . . . . . . . . . . . . . . . 18
5.2.6. localDescription . . . 16
5.1.8. updateIce . . . . . . . . . . . . . . . . 18
5.2.7. remoteDescription . . . . . . . 16
5.1.9. addIceCandidate . . . . . . . . . . . 18
5.2.8. updateIce . . . . . . . . . 17
5.2. Configurable SDP Parameters . . . . . . . . . . . . . 18
5.2.9. addIceCandidate . . . 17
6. Media Setup Overview . . . . . . . . . . . . . . . . 19
6. Configurable SDP Parameters . . . . . 17
6.1. Initiating the Session . . . . . . . . . . . . 20
7. Security Considerations . . . . . . 18
6.1.1. Generating An Offer . . . . . . . . . . . . . 21
8. IANA Considerations . . . . . 18
6.1.2. Applying the Offer . . . . . . . . . . . . . . . . 22
9. Acknowledgements . . 18
6.1.3. Handling ICE Callbacks . . . . . . . . . . . . . . . . 18
6.1.4. Serializing the Offer and Candidates . . . . . 23
10. References . . . . 19
6.2. Receiving the Session . . . . . . . . . . . . . . . . . . . 19
6.2.1. Receiving the Offer . . . 24
10.1. Normative References . . . . . . . . . . . . . . . 19
6.2.2. Handling ICE Messages . . . . 24
10.2. Informative References . . . . . . . . . . . . . 19
6.2.3. Generating the Answer . . . . . 24
Appendix A. JSEP Implementation Examples . . . . . . . . . . . . 20
6.2.4. Applying the Answer 26
A.1. Example API Flows . . . . . . . . . . . . . . . . . . 20
6.2.5. Serializing the Answer . . 26
A.1.1. Call using ROAP . . . . . . . . . . . . . . 20
6.3. Completing the Session . . . . . 26
A.1.2. Call using XMPP . . . . . . . . . . . . . 20
6.3.1. Receiving the Answer . . . . . . 27
A.1.3. Adding video to a call, using XMPP . . . . . . . . . . . 20

6.4. Updates to the Session . . . . . . . . . . . . . . . . . . 20
7. Security Considerations . . . . . . . . . . . . . . . . . . . . 21
8. IANA Considerations . . . . . . . . . . . . . . . . . . . . . . 21
9. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 21
10. References . . . . . . . . . . . . . . 28
A.1.4. Simultaneous add of video streams, using XMPP . . . . 28
A.1.5. Call using SIP . . . . . . . . 21
10.1. Normative References . . . . . . . . . . . . 29
A.1.6. Handling early media (e.g. 1-800-GO FEDEX), using
SIP . . . . . . . 21
10.2. Informative References . . . . . . . . . . . . . . . . . . 21 30
Appendix A. JSEP Implementation Examples . . . . . . . . . . . . . 22
A.1. Example API . . . . . . . . . . . . . . . . . . . . . . . . 22
A.2. Example API Flows . . . . . . . . . . . . . . . . B. Change log . . . . . 23
A.2.1. Call using ROAP . . . . . . . . . . . . . . . . 32
Authors' Addresses . . . . 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

1. Introduction

The thinking behind WebRTC call setup has been 33

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 browser environment also has its own challenges that cause
problems for WebRTC attempted to implement this protocol-
agnostic an embedded signaling by providing a mechanism to exchange session
descriptions in the form state machine. One of SDP blobs. Upon starting a session, these is
that the
browser would generate user may reload the web page at any time. If this happens,
and the state machine is being run at a SDP blob, which would be passed server, the server can simply
push the current state back down to the
application for transport over its preferred page and resume the call
where it left off.

This document describes the Javascript Session Establishment Protocol
(JSEP) that pulls the signaling protocol. On state machine out of the remote side, this blob would be passed browser and
into Javascript. This mechanism effectively removes the browser
almost completely from the
application, and core signaling flow; the browser would then generate only interface
needed is a blob of its own in
response. Upon transmission back way for the application to pass in the initiator, this blob would be
plugged into their browser, local and the handshake would be complete.

Experimentation with this remote
session descriptions negotiated by whatever signaling mechanism turned up several shortcomings,
which generally stemmed from there being insufficient context at the
browser is
used, and a way to fully determine interact with the meaning ICE state machine.

JSEP's handling of a SDP blob. For example,
determining whether a blob session descriptions is simple and
straightforward. Whenever an offer or offer/answer exchange is needed, the
initiating side creates an answer, or
differentiating a new offer from by calling a retransmit. createOffer() API. The ROAP proposal, specified in [I-D.draft-jennings-rtcweb-signaling-
01], attempted
application optionally modifies that offer, and then uses it to resolve these issues by providing additional
structure in set
up its local config via the messaging - in essence, setLocalDescription() API. The offer is
then sent off to create a generic the remote side over its preferred signaling protocol
mechanism (e.g., WebSockets); upon receipt of that specifies how offer, the browser signaling state
machine should operate. However, even though remote
party installs it using the protocol setRemoteDescription() API.

When the call is
abstracted, accepted, the state machine forces a least-common-denominator
approach on callee uses the signaling interactions. For example, in Jingle, createAnswer() API to
generate an appropriate answer, applies it using
setLocalDescription(), and sends the answer back to the
call initiator can provide additional ICE candidates even after
over the
initial offer has been sent, which allows signaling channel. When the offer to offerer gets that answer, it
installs it using setRemoteDescription(), and initial setup is
complete. This process can be sent
immediately repeated for quicker call startup. However, in additional offer/answer
exchanges.

Regarding ICE, JSEP decouples the browser ICE state machine from the overall
signaling state machine, there is no notion of sending an updated offer before as 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 ICE state machine within must remain in the browser. Since
browser, because only the browser generates has the necessary knowledge of
candidates and other transport info. Performing this separation also
provides additional flexibility; in protocols that decouple session
descriptions on its own, and fully controls from transport, such as Jingle, the possible
states transport
information can be sent separately; in protocols that don't, such as
SIP, the information can be used in the aggregated form. Sending
transport information separately can allow for faster ICE and advancement of 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, modification
of 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 or use and ICE information into the
defined messages of alternate state machines
becomes difficult or impossible.

The browser environment also has its own challenges that cause
problems for an embedded chosen signaling state machine. One protocol, instead of these is
that simply
forwarding the user may reload messages emitted from the web page at any time. If browser.

One way to mitigate this happens,
and the state machine is being run at to provide a server, Javascript library that
hides this complexity from the server can simply
push developer, which would implement the current
state back down to the page machine and resume the call
where it left off.

If instead serialization of the state machine is run at the browser end, and is
instantiated within, for desired signaling protocol.
For example, this library could convert easily adapt the PeerConnection object, that
state machine will be reinitialized when the page is reloaded and JSEP API
into the
JavaScript re-executed. This actually complicates exact ROAP API [I-D.jennings-rtcweb-signaling], thereby
implementing the design ROAP signaling protocol. Such a library could of any
interoperability service, as all cases where an offer
course also implement other popular signaling protocols, including
SIP or answer has
already been generated but is now "forgotten" must now be handled by
trying Jingle. In this fashion we can enable greater control for the
experienced developer without forcing any additional complexity on
the novice developer.

2. Other Approaches Considered

Another approach that was considered for JSEP was to move the client state machine forward to
mechanism for generating offers and answers out of the same state 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
had been in previously needed in
order to match what has already been
delivered to and/or answered by generate its own session descriptions. This increases the far side, or handled by ensuring
amount of work that aborts are cleanly handled the application needs to do; it needs to know how
to generate session descriptions from every state capabilities, and especially
how to generate the negotiation
rapidly restarted.

1.1. Terminology

The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
"SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", correct answer from an arbitrary offer and "OPTIONAL" in the
supported capabilities. While this
document are to could certainly be interpreted as described in RFC 2119 [RFC2119].

2. JSEP Approach

To resolve addressed by
using a library like the issues one mentioned above, this document proposes the
Javascript Session Establishment Protocol (JSEP) that pulls it basically forces the
signaling state machine out
use 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 said library even 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, example. Exposing createOffer/
createAnswer avoids that problem, but still allows applications to
generate their own offers/answers if they choose, using the
initiating side creates an offer
description generated by calling a createOffer() API. The
application can do massaging createOffer as an indication of the
browser's capabilities.

Note also that offer, if it wants to, and then
uses while JSEP transfers more control to Javascript, it is
not intended to set up its local config via be an example of a setLocalDescription() "low-level" API. The offer general
argument against a low-level API is then sent off that there are too many necessary
API points, and they can be called in any order, leading to the remote side over its preferred
signaling mechanism (e.g. WebSockets); upon receipt of something
that offer,
the remote party installs it using a setRemoteDescription() API.

When the call is accepted, hard to specify and test. In the callee uses approach proposed here,
control is performed via session descriptions; this requires only a createAnswer() API
few APIs to
generate an appropriate answer, applies it using
setLocalDescription(), handle these descriptions, and sends they are evaluated in a
specific fashion, which reduces the answer back to the initiator
over the signaling channel. When the offerer gets that answer, it
installs it using setRemoteDescription(), number of possible states and initial setup is
complete. This process can
interactions.

3. Terminology

The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
"SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
document are to be repeated for additional offer/answer
exchanges.

Regarding ICE, interpreted as described in RFC 2119 [RFC2119].

4. Semantics and Syntax

4.1. Signaling Model

JSEP decouples the ICE state machine from the overall does not specify a particular signaling model or 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, than the information can be easily aggregated and recombined. Sending
transport information separately can allow for faster ICE generic need to exchange RFC 3264 offers and DTLS
startup, since the necessary roundtrips can occur while waiting answers
in order for both sides of the remote side session to accept know how to conduct the
session.

The JSEP approach does come with provides mechanisms to create offers and answers, as
well as to apply them to a minor downside. As session. However, the application
now is responsible for driving actual mechanism by
which these offers and answers are communicated to the signaling state machine, slightly
more application code remote side,
including addressing, retransmission, forking, and glare handling, is necessary
left entirely up to perform call setup; the
application must call application.

Figure 1: JSEP Signaling Model

4.2. Session Descriptions and State Machine

In order to establish the right APIs at media plane, the right times, and convert user agent needs specific
parameters to indicate what to transmit to the remote side, as well
as how to handle the media that is received. These parameters are
determined by the exchange of session descriptions in offers and ICE information into the defined
messages of its chosen signaling protocol, instead of simply
forwarding the messages emitted from the browser.

One way
answers, and there are certain details to mitigate this is to provide a Javascript library process that
hides this complexity from must be
handled in the developer, which would implement JSEP APIs.

Whether a session description was sent or received affects the
state machine and serialization
meaning of the desired signaling protocol. that description. For example, this library could convert easily adapt the JSEP API
into list of codecs sent to
a remote party indicates what the exact ROAP API, thereby implementing local side is willing to decode,
and what the ROAP signaling
protocol. Such a library could of course also implement other popular
signaling protocols, including SIP or Jingle. In remote party should send. Not all parameters follow
this fashion we can
enable greater control rule; for example, the experienced developer without forcing
any additional complexity on SRTP parameters [RFC4568] sent to a
remote party indicate what the novice developer.

3. Other Approaches Considered

Another approach that was considered for JSEP was local side will use to move encrypt, and
thereby how the
mechanism for generating 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 answers out of an answer have been exchanged, it is possible for the browser as
well. Instead
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 providing createOffer/createAnswer methods within the browser, this approach would instead expose a getCapabilities API
which would provide details
of the application with offerer before the information it needed answer arrives.

Therefore, in order to generate its own handle session descriptions. This increases the
amount of work that descriptions properly, the application needs to do; it needs to user
agent needs:

1. To know how
to generate if a session descriptions from capabilities, and especially
how description pertains to generate the correct answer from local or remote
side.

2. To know if a session description is an arbitrary offer and or an answer.

3. To allow the
supported capabilities. While this could certainly offer to be addressed specified independently of the answer.

JSEP addresses this by
using adding both a library like the one mentioned above, it basically forces setLocalDescription and a
setRemoteDescription method and having session description objects
contain a type field indicating the
use type of said library even session description being
supplied. This satisfies the requirements listed above for a simple example. Exposing
createOffer/createAnswer avoids that problem, but still allows
applications to generate their own offers/answers if they choose,
using both the description generated by createOffer
offerer, who first calls setLocalDescription(sdp [offer]) and then
later setRemoteDescription(sdp [answer]), as an indication of well as for the browser's capabilities.

Note also that while JSEP transfers more control
answerer, who first calls setRemoteDescription(sdp [offer]) and then
later setLocalDescription(sdp [answer]). While it could be possible
to Javascript, implicitly determine the value of the offer/answer argument,
requiring it is
not intended to be an example of a "low-level" API. The general
argument against a low-level API specified explicitly is that there are too many necessary
API points, and they can be called in any order, leading more robust, allowing
invalid combinations (i.e. an answer before an offer) to something
that is hard generate an
appropriate error.

JSEP also allows for an answer to specify and test. In be treated as provisional by the approach proposed here,
control is performed via session descriptions; this requires only
application. Provisional answers provide a
few APIs way for an answerer 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
communicate initial session parameters back to exchange RFC 3264 offers and answers the offerer, in order for both sides of
to allow the session to know how to conduct the
session. JSEP provides mechanisms to create offers and answers, as
well as to apply them begin, while allowing a final answer to be
specified later. This concept of a session. However, final answer is important to the actual mechanism
offer/answer model; when such an answer is received, any extra
resources allocated by
which these offers and answers are communicated to the remote side,
including addressing, retransmission, forking, caller can be released, now that the exact
session configuration is known. These "resources" can include things
like extra ICE components, TURN candidates, or video decoders.
Provisional answers, on the other hand, do no such deallocation
results; as a result, multiple dissimilar provisional answers can be
received and glare handling, applied during call setup.

In [RFC3264], the constraints at the signaling level is
left entirely up to that only one
offer can be outstanding for a given session but from the application.

+-----------+ media stack
level, a new offer can be generated at any point. For example, when
using SIP for signaling, if one offer is sent, then cancelled using a
SIP CANCEL, another offer can be generated even though no answer was
received for the first offer. To support this, the JSEP media layer
can provide an offer whenever the Javascript application needs one
for the signaling. The answerer can send back zero or more
provisional answers, and finally end the offer-answer exchange by
sending a final answer. The state machine for this is as follows:

+-----------+
| Web App |<--- App-Specific Signaling --->| Web App |
+-----------+
| |
| Stable |<---------------\
| | |
| | |
+-----------+ |
^ | | SDP
| SDP
V | OFFER |
ANSWER | | | ANSWER
| V |
+-----------+ +-----------+
| Browser |<----------- Media ------------>| Browser |
+-----------+ +-----------+

Figure 1: JSEP Signaling Model

4.2. Session Descriptions and | |
| | PRANSWER | |
| Offer |-------- >| Pranswer |
| | | |
| |----\ | |----\
+-----------+ | +-----------+ |
^ | ^ |
| | | |
\-----/ \-----/
OFFER PRANSWER

Figure 2: JSEP State Machine

In order to establish the media plane,

Aside from these state transitions, there is no other difference
between the user agent needs specific
parameters to indicate what to transmit to handling of provisional ("pranswer") and final ("answer")
answers.

4.3. Session Description Format

In the remote side, as well WebRTC specification, session descriptions are formatted as how to handle the media that
SDP messages. While this format is received. These parameters are
determined by the exchange not optimal for manipulation from
Javascript, it is widely accepted, and frequently updated with new
features. Any alternate encoding of session descriptions in offers and
answers, and there are certain details would have
to this process that must be
handled in keep pace with the JSEP APIs.

Whether a session description was sent or received affects changes to SDP, at least until the
meaning of time that description. For example, the list of codecs sent to
this new encoding eclipsed SDP in popularity. As a remote party indicates what result, JSEP
continues to use SDP as the local side is willing internal representation for its session
descriptions.

However, to decode, simplify Javascript processing, and what the remote party should send. Not all parameters follow this
rule; provide for example, future
flexibility, the SRTP parameters [RFC4568] sent SDP syntax is encapsulated within a
SessionDescription object, which can be constructed from SDP, and be
serialized out to SDP. If future specifications agree on a remote
party indicate what the local side will use JSON
format for session descriptions, we could easily enable this object
to encrypt, generate and thereby
how the remote party should expect consume that JSON.

Other methods may be added to receive.

In addition, various RFCs put different conditions on SessionDescription in the format future to
simplify handling 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, SessionDescriptions from Javascript. Though it
is possible for the
offerer unclear exactly what manipulations developer will commonly want 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
do to SDP, it would be aware of the details
of the offerer before the answer arrives.

Therefore, in order simple to handle session descriptions properly, the user
agent needs:

1. To know if write a session description pertains Javascript library to the local or
remote side.

2. To know if
perform these manipulations.

4.4. ICE

When a session description new ICE candidate is an offer or an answer.

3. To allow available, the offer to be specified independently of ICE Agent will notify the answer.

JSEP addresses this by adding both a setLocalDescription and
application via a
setRemoteDescription method, and both callback; these methods take a parameter candidates will automatically be
added to indicate the type of local session description being supplied. This
satisfies the requirements listed above for both description. When all candidates have
been gathered, 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 callback will also be possible invoked to
implicitly determine the value of signal that the offer/answer argument,
requiring it to be specified explicitly
gathering process is more robust, allowing
invalid combinations (i.e. an answer before an offer) complete.

4.4.1. ICE Candidate Trickling

Candidate trickling is a technique through which a caller may
incrementally provide candidates to generate an
appropriate error.

It also the callee after the initial
offer has been dispatched; the semantics of "Trickle ICE" are defined
in [I-D.rescorla-mmusic-ice-trickle]. This process 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 callee
to begin acting upon the offerer, in order for call and setting up the
session ICE (and perhaps
DTLS) connections immediately, without having to begin, while allowing a final answer wait for the caller
to be specified
later. gather all possible candidates. This concept of a final answer results in faster call
startup in cases where gathering is important not performed prior to initating
the
offer/answer model; when such an answer is received, any extra
resources allocated call.

JSEP supports optional candidate trickling by the caller can be released, now providing APIs that the exact
session configuration is known. These "resources" can include things
like extra ICE components, TURN candidates, or video decoders.
Provisional answers,
provide control and feedback on the other hand, do no such deallocation; as a
result, multiple dissimilar provisional answers ICE candidate gathering process.
Applications that support candidate trickling can be received send the initial
offer immediately 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 individual candidates when they get the handling
notified of provisional ("pranswer") and final ("answer")
answers.

4.3. Session Description Format
In the current WebRTC specification, session descriptions are
formatted as SDP messages. While this format is a new candidate; applications that do not optimal support this
feature can simply wait for
manipulation from Javascript, it the indication that gathering is widely accepted,
complete, and frequently
updated then create and send their offer, with new features. Any alternate encoding all the
candidates, at this time.

Upon receipt of session
descriptions would have to keep pace with trickled candidates, the changes receiving application will
supply them to SDP, at
least until its ICE Agent. This triggers the time that this new encoding eclipsed SDP in
popularity. As a result, JSEP continues ICE Agent to use SDP as start
using the internal
representation new remote candidates for its connectivity checks.

4.4.1.1. ICE Candidate Format

As with session descriptions.

However, to simplify Javascript processing, and provide for future
flexibility, descriptions, the SDP syntax is encapsulated within a
SessionDescription object, which of the IceCandidate object
provides some abstraction, but 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 converted to
simplify handling of SessionDescriptions from Javascript.

4.4. Separation of Signaling and ICE State Machines

JSEP does away with from
the SDP Agent a=candidate lines.

The a=candidate lines are the only SDP information that is contained
within IceCandidate, as they represent the browser, and this
functionality only information needed
that is now controlled directly by not present in the application, which
uses initial offer (i.e. for trickle
candidates). This information is carried with the setLocalDescription and setRemoteDescription APIs to tell same syntax as the browser what SDP has been negotiated. The ICE Agent remains
"a=candidate" line in
the browser, as it still needs SDP. For example:

a=candidate:1 1 UDP 1694498815 192.0.2.33 10000 typ host

The IceCandidate object also contains fields to drive the process indicate which m=
line it should be associated with. The m line can be identified in
one of gathering
candidates, connectivity checks, and related ICE functionality.

When two ways; either by a new ICE candidate m-line index, or a MID. The m-line
index is available, the ICE Agent will notify the
application via a callback; these candidates will automatically be
added zero-based index, referring to the local session description. When all candidates have been
gathered, Nth m-line in the callback will also be invoked SDP.
The MID uses the "media stream identification", as defined in [RFC
3388], to signal that identify the
gathering process is complete.

4.5. m-line. WebRTC implementations creating an
ICE Candidate Trickling object MUST populate both of these fields.
Implementations receiving an ICE Candidate trickling is a technique through which a caller object SHOULD use the MID
if they implement that functionality, or the m-line index, if not.

4.5. Interactions With Forking

Some call signaling systems allow various types of forking where an
SDP Offer may
incrementally provide candidates be provided to more than one device. For example, SIP
RFC 3261 defines both a "Parallel Search" and "Sequential Search".
Although these are primarily signaling level issues that are outside
the callee after scope of JSEP, they do have some impact on the initial
offer has been dispatched. This allows configuration of
the callee to begin acting
upon media plane, which is relevant. When forking is happening at the call and setting up
signaling layer, the ICE (and perhaps DTLS) connections
immediately, without having to wait Javascript application responsible for the caller
signaling needs to allocate all
possible candidates, resulting in faster call startup in many cases. make the decisions about what media should be sent
or received at any point of time and which remote endpoint it should
communicate with. JSEP supports optional candidate trickling by providing APIs that
provide control is used to make sure the media engine can
make the RTP and feedback on media perform as required by the ICE candidate gathering process.
Applications application. The
basic operations that support candidate trickling the applications can send have the initial
offer immediately and send individual candidates when they get media engine do
are:

Start exchanging media to a given remote peer but keep all the
onicecandidate callback
resources reserved in the offer.

Start exchanging media with a new candidate; applications given remote peer and free any
resources in the offer that do are not support this feature being used.

4.5.1. Sequential Forking

Sequential forking involves a call being dispatched to multiple
remote callees, where each callee 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 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 reject any subsequent
answers. In SIP parlance, this would be ACK + BYE.

In a "last-one-wins" situation, all answers would be applied as
provisional answers, and any previous call leg will be terminated.
At some point, the application will end the setup process, perhaps
with a timer; At at this point, the application could reapply the
existing remote description as a final answer.

4.7.2.

4.5.2. Parallel Forking

Parallel forking involves a call being dispatched to multiple remote
callees, where each callee can accept the call, and multiple
simultaneous active signaling sessions can be established as a
result. If multiple callees send media, this media is mixed and played out at the caller side.

JSEP can handle parallel forking by "cloning" same time, the session when needed
possibilities for handling this are described in Section 3.1 of RFC
3960. Most SIP devices today only support exchanging media with a
single device at a time, and do not try to create mix multiple parallel sessions. When the first answer is
received, the caller can clone the existing session, and then apply
the answer early media
audio sources, as that could result in a final answer to confusing situation. For
example. consider having a European ringback tone mixed together with
the original session. Upon receiving
the next answer, North American ringback tone - the cloned session is cloned again, resulting sound would not be
like either tone, and would confuse the received
answer is applied as a final answer user. If the signaling
application wishes to only exchange media with one of the first clone. This process
repeats until remote
endpoints at a time, then from a media engine point of view, this is
exactly like the caller decides to end sequential forking case.

In the setup flow, and closes parallel forking case where the final cloned session.

Cloned sessions inherit Javascript application wishes
to simultaneously exchange media with multiple peers, the local session description and candidates
from their parent, and an empty remote description; only sessions
that have not yet applied an answer flow is
slightly more complex, but the Javascript application can be cloned. Each cloned
session may discover new peer-reflexive candidates; these candidates
will be supplied via follow the onicecandidate callback to
strategy that specific
session. Since RFC 3960 describes using UPDATE. (It is worth noting
that use cases where this is the clone uses desired behavior are very unusual.)
The UPDATE approach allows the same local description as its
parent, creating signaling to set up a clone will fail if separate media
flow for each peer that it is not possible wishes to reserve exchange media with. In JSEP,
this offer used in the same resources for UPDATE would be formed by simply creating a
new PeerConnection and making sure that the clone as same local media streams
have already been reserved added into this new PeerConnection. Then the new
PeerConnection object would produce a SDP offer that could be used by
the
parent. signaling to perform the UPDATE strategy discussed in RFC 3690.

As a result of this cloning, sharing the media streams, the application will end up
with N parallel PeerConnection sessions, each with a local and remote
description and their own local and remote addresses. The media flow
from these sessions can be managed by specifying SDP direction
attributes in the descriptions, or the application can choose to play
out the media from all sessions mixed together. Of course, if the
application wants to only keep a single session, it can simply
terminate the sessions that it no longer needs.

4.8.

4.6. 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
"rehydration".

With rehydration, the current local session description signaling state 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 signaling state 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 and a new ICE ufrag/pwd combo PeerConnection object is created for the local
description.

The modified local description is then installed via
setLocalDescription, and sent off as an
session. At this point a new offer can be generated by the new
PeerConnection, with new ICE and SDES credentials. This can then be
used to re-initiate the session with the existing remote side, endpoint,
who simply sees the new offer as an in-call renegotiation, and 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.

Open Issue: EKR proposed an alternative rehydration approach where
the actual internal PeerConnection object in the browser was kept
alive for some time after the web page was killed and provided some
way for a new page to acquire the old PeerConnection object.

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. SDP Requirements

Note: The text in this section may not represent working group
consensus and is put here so that the working group can discuss it
and find out how to change it such that it does have consensus.

When generating SDP blobs, either for offers or answers, the
generated SDP needs to conform to the following specifications.
Similarly, in order to properly process received SDP blobs,
implementations need to implement the functionality described in the
following specifications. This list is derived from
[I-D.ietf-rtcweb-rtp-usage].

RFC4566 is the base SDP specification and MUST be implemented.

RFC5124 MUST be supported for signaling RTP/SAVPF RTP profile.

RFC5104 MUST be implemented to signal RTCP based feedback.

RFC5761 MUST be implemented to signal multiplexing of RTP and
RTCP.

RFC5245 MUST be implemented for signaling the ICE candidate lines
corresponding to each media stream.

RFC3264 MUST be implemented to signal information about media
direction.

The RFC5888 grouping framework MUST be implemented for signaling
the grouping information.

RFC5506 MAY be implemented to signal Reduced-Size RTCP messages.

RFC5576 MAY be implemented to signal RTP SSRC values.

RFC3556 with bandwidth modifiers MAY be supported for specifying
RTCP bandwidth as a fraction of the media bandwidth, RTCP fraction
allocated to the senders and setting maximum media bit-rate
boundaries.

As required by RFC 4566 Section 5.13 JSEP implementations MUST ignore
unknown attributes (a=) lines.

Example SDP for RTCWeb call flows can be found in
[I-D.nandakumar-rtcweb-sdp].

5.2. Methods

5.1.1.

5.2.1. 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
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 may do things such as generate new ICE
credentials, but does not change state; its use is not required.

5.1.2. media state.

5.2.2. 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|. the 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 may do things such as generate new ICE
credentials, but does not change state; its use is not required.

5.1.3. media state.

5.2.3. SessionDescriptionType

The strings

Session description objects (RTCSessionDescription) may be of type
"offer", "pranswer", and "answer" serve as type arguments
to setLocalDescription and setRemoteDescription. They "answer". These types provide information
as to how the description parameter should be parsed, and how the
media state should be changed.

"offer" indicates that a description should be parsed as an offer;
said description may include many possible media configurations. A
description used as an "offer" may be applied anytime the
PeerConnection is in a stable state, or as an update to a previously
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.

5.2.3.1. Creating Answers

Most web applications will not need to create answers using the
"pranswer" type. The general recommendation for a web application
would be to create an answer more or less immediately after receiving
the offer, instead of waiting for a human user to provide input.
Later when the human input is received, the applications can create a
new offer to update the previous offer/answer pair. Some
applications may not be able to do this, particularly ones that Some
application may not be able to do this, particular ones that are
attempting to gateway to other signaling protocols.

Consider a typical web application that will set up a data channel,
an audio channel, and a video channel. When an endpoint receives an
offer with these channels, it could send an answer accepting the data
channel for two-way data, and accepting the audio and video tracks as
receive-only. It could then ask the user if they wanted to transmit
audio and video to the far end, acquire the local media streams, and
send a new offer to the remote side moving the audio and video to be
two-way media. By the time the human has authorized sending media,
it is likely that the ICE and DTLS handshaking with the remote side
will already be set up.

5.2.4. setLocalDescription

The setLocalDescription method instructs the PeerConnection to apply
the supplied SDP blob as its local configuration. The type parameter field
indicates whether the blob should be processed as an offer,
provisional answer, or final answer; offers and answers are checked
differently, using the various rules that exist for each SDP line.

This API changes the local media state; among other things, it sets
up local resources for receiving and decoding media. In order to
successfully handle scenarios where the application wants to offer to
change from one media format to a different, incompatible format, the
PeerConnection must be able to simultaneously support use of both the
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.

5.2.5. setRemoteDescription

The setRemoteDescription method instructs the PeerConnection to apply
the supplied SDP blob as the desired remote configuration. As in
setLocalDescription, the |type| parameter type field of the indicates how the blob
should be processed.

This API changes the local media state; among other things, it sets
up local resources for sending and encoding media.

5.1.6.

5.2.6. 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.

5.2.7. 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.

5.2.8. 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
may result in a change to media state if it results in connectivity
being established.

5.1.9.

5.2.9. 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.

6. Configurable SDP Parameters

The following

Note: This section is still very early and is likely to
significantly change as we get a partial list better understanding of SDP parameters that an application
may want to control, the a) the
use cases for this b) the implications at the protocol level c)
feedback from implementors on what they can do.

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.

o remove or reorder codecs (m=)
-

o change codec attributes (a=fmtp; ptime)
-

o enable/disable BUNDLE (a=group)
-

o enable/disable RTCP mux (a=rtcp-mux)
- remove or reorder SRTP crypto-suites (a=crypto)
- change SRTP parameters or keys (a=crypto)
-

o change send resolution or framerate (TBD)
-

o change desired recv resolution or framerate (TBD)
-

o change total bandwidth (b=)
-

o remove desired AVPF mechanisms (a=rtcp-fb)
-

o remove RTP header extensions (a=rtphdr-ext)
-

o add/change SSRC grouping (e.g. FID, RTX, etc) (a=ssrc-group)
-

o add SSRC attributes (a=ssrc)
- change ICE ufrag/password (a=ice-ufrag/pwd)
-

o 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

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 the state changes that
occur. We assume the following architecture in Adam Bergkvist all provided valuable feedback on this example, where UA
is synonymous with "browser", and JS is synonymous with "web
application":

OffererUA <-> OffererJS <-> WebServer <-> AnswererJS <-> AnswererUA

6.1. Initiating the Session

The initiator creates a PeerConnection, hooks up to its ICE callback,
proposal. Suhas Nandakumar provided text and adds the desired MediaStreams (presumably obtained via
getUserMedia). The ICE gathering process begins to gather candidates input for a default number of streams, as the exact number will not be
known until SDP
requirements. Matthew Kaufman provided the local description is applied. The PeerConnection is
in observation that keeping
state out of 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 browser allows a session description to offer call to continue even if the
callee. This description includes the codecs page
is reloaded.

10. References

10.1. Normative References

[I-D.rescorla-mmusic-ice-trickle]
Rescorla, E., Uberti, J., and other necessary
session parameters, as well as information about each E. Ivov, "Trickle ICE:
Incremental Provisioning of Candidates for 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 Interactive
Connectivity Establishment (ICE) Protocol",
draft-rescorla-mmusic-ice-trickle-00 (work in progress),
October 2012.

[RFC2119] Bradner, S., "Key words 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
The initiator starts to receive callbacks on its onicecandidate
handler. Candidates are provided RFCs to the IceCallback as they are
allocated; when the last allocation completes or times out, this
callback will be invoked Indicate
Requirement Levels", BCP 14, RFC 2119, March 1997.

[RFC3264] Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model
with a null argument.

OffererUA->OffererJS: onIceCandidate(candidate);

6.1.4. Serializing the Offer Session Description Protocol (SDP)", RFC 3264,
June 2002.

[RFC4566] Handley, M., Jacobson, V., 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, C. Perkins, "SDP: Session
Description Protocol", RFC 4566, July 2006.

10.2. Informative References

[I-D.ietf-rtcweb-rtp-usage]
Perkins, C., Westerlund, M., 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 J. Ott, "Web Real-Time
Communication (WebRTC): Media Transport and list of candidates all at once.

6.2. Receiving the Session

Through the chosen signaling protocol, the recipient is notified Use of
an incoming session request. It creates a PeerConnection, RTP",
draft-ietf-rtcweb-rtp-usage-04 (work in progress),
July 2012.

[I-D.jennings-rtcweb-signaling]
Jennings, C., Rosenberg, J., 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, R. Jesup, "RTCWeb Offer/
Answer Protocol (ROAP)",
draft-jennings-rtcweb-signaling-01 (work in progress),
October 2011.

[I-D.nandakumar-rtcweb-sdp]
Nandakumar, S. 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 C. Jennings, "SDP 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 WebRTC",
draft-nandakumar-rtcweb-sdp-00 (work 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.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
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., progress),
October 2012.

[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 (SDP) Security Descriptions for
Offer/Answer Protocols", Media
Streams", RFC 5245, April 2010.

[webrtc-api] Bergkvist, Burnett, Jennings, Narayanan, "WebRTC 1.0:

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,
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. 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.

[W3C.WD-webrtc-20111027]
Bergkvist, A., Burnett, D., Narayanan, A., and C.
Jennings, "WebRTC 1.0: Real-time Communication Between
Browsers", World Wide Web Consortium WD WD-webrtc-
20111027, October 2011,
<http://www.w3.org/TR/2011/WD-webrtc-20111027>.

Appendix A. JSEP Implementation Examples

A.1. Example API Flows

Below are several sample flows for the new PeerConnection and library
APIs, demonstrating when the various APIs are called in different
situations and with various transport protocols. For clarity and
simplicity, the createOffer/createAnswer calls are assumed to be
synchronous in these examples, whereas the actual APIs are async.

A.2.1.

A.1.1. 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);
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

A.1.2. 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"/>

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.

A.1.3. Adding video to a call, using XMPP

This example demonstrates an XMPP call, where the XMPP content-add
mechanism is used to add video media to an existing session. For
simplicity, candidate exchange is not shown.

Note that the offerer for the change to the session may be different
than the original call offerer.

// Offerer adds video stream
OffererJS->OffererUA: pc.addStream(videoStream)
OffererJS->OffererUA: offer = pc.createOffer(null);
OffererJS: xmpp = createContentAdd(offer);
OffererJS->OffererUA: pc.setLocalDescription("offer", offer);
OffererJS->AnswererJS: <jingle action="content-add"/>

// content-add arrives at Answerer
AnswererJS: offer = parseContentAdd(xmpp);
AnswererJS->AnswererUA: pc.setRemoteDescription("offer", offer);
AnswererJS->AnswererUA: answer = pc.createAnswer(offer, null);
AnswererJS->AnswererUA: pc.setLocalDescription("answer", answer);
AnswererJS: xmpp = createContentAccept(answer);
AnswererJS->OffererJS: <jingle action="content-accept"/>

// content-accept arrives at Offerer
OffererJS: answer = parseContentAccept(xmpp);
OffererJS->OffererUA: pc.setRemoteDescription("answer", answer);

A.2.4.

A.1.4. Simultaneous add of video streams, using XMPP

This example demonstrates an XMPP call, where new video sources are
added at the same time to a call that already has video; since adding
these sources only affects one side of the call, there is no
conflict. The XMPP description-info mechanism is used to indicate
the new sources to the remote side.

// 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/> type="result"/> // ack

// description-info arrives at Offerer, and is acked
OffererJS: offer = parseDescriptionInfo(xmpp);
OffererJS->AnswererJS: <iq type="result/> 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.

A.1.5. Call using SIP

This example demonstrates a simple SIP call (e.g. where the client
talks to a SIP proxy over WebSockets).

// 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

// ICE Completes (at Offerer)
OffererUA->OffererJS: onopen();
OffererUA->AnswererUA: Media

A.2.6.

A.1.6. Handling early media (e.g. 1-800-FEDEX), 1-800-GO FEDEX), using SIP

This example demonstrates how early media could be handled; for
simplicity, only the offerer side of the call is shown.

// 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;
// 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

Appendix B. Change log

Changes in draft -02:

o Converted from nroff

o Removed comparisons to old approaches abandoned by 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 working
group

o Removed stuff that would delay the pranswer
// (due has moved 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 W3C specificaiton

o Align SDP locally and send it to the remote side.
function setLocalAndSendMessage(type, sdp) {
pc.setLocalDescription(type, sdp);
sendMessage(type, sdp);
}
// 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: handling with W3C draft

o Clarified section on forking.

Changes in draft -01:

o Added diagrams for architecture and state machine.

o Added sections on forking and rehydration.

o Clarified meaning of "pranswer" and "answer".

o Reworked how ICE restarts and media directions are controlled.

o Added list of parameters that can be changed in a description.

o Updated suggested API and examples to match latest thinking.

o Suggested API and examples have been moved to an appendix.
00:

Changes in draft -00:

o Migrated from draft-uberti-rtcweb-jsep-02.

Authors' Addresses

Justin Uberti
Google
5 Cambridge Center
Cambridge, MA 02142
747 6th Ave S
Kirkland, WA 98033
USA

Email: justin@uberti.name

Cullen Jennings
Cisco
170 West Tasman Drive
San Jose, CA 95134
USA

Email: fluffy@cisco.com fluffy@iii.ca