draft-ietf-rtcweb-jsep-02.txt   draft-ietf-rtcweb-jsep-03.txt 
Network Working Group J. Uberti Network Working Group J. Uberti
Internet-Draft Google Internet-Draft Google
Intended status: Standards Track C. Jennings Intended status: Standards Track C. Jennings
Expires: April 25, 2013 Cisco Expires: August 29, 2013 Cisco
October 22, 2012 February 25, 2013
Javascript Session Establishment Protocol Javascript Session Establishment Protocol
draft-ietf-rtcweb-jsep-02 draft-ietf-rtcweb-jsep-03
Abstract Abstract
This document proposes a mechanism for allowing a Javascript This document describes the mechanisms for allowing a Javascript
application to fully control the signaling plane of a multimedia application to fully control the signaling plane of a multimedia
session, and discusses how this would work with existing signaling session via the interface specified in the W3C RTCPeerConnection API,
protocols. and discusses how this relates to existing signaling protocols.
Status of this Memo Status of this Memo
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This Internet-Draft will expire on April 25, 2013. This Internet-Draft will expire on August 29, 2013.
Copyright Notice Copyright Notice
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Table of Contents Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . . 3 1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . . 4
2. Other Approaches Considered . . . . . . . . . . . . . . . . . 5 1.1. General Design of JSEP . . . . . . . . . . . . . . . . . . 4
3. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 6 1.2. Other Approaches Considered . . . . . . . . . . . . . . . 6
4. Semantics and Syntax . . . . . . . . . . . . . . . . . . . . . 7 2. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 6
4.1. Signaling Model . . . . . . . . . . . . . . . . . . . . . 7 3. Semantics and Syntax . . . . . . . . . . . . . . . . . . . . . 7
4.2. Session Descriptions and State Machine . . . . . . . . . . 7 3.1. Signaling Model . . . . . . . . . . . . . . . . . . . . . 7
4.3. Session Description Format . . . . . . . . . . . . . . . . 9 3.2. Session Descriptions and State Machine . . . . . . . . . . 7
4.4. ICE . . . . . . . . . . . . . . . . . . . . . . . . . . . 10 3.3. Session Description Format . . . . . . . . . . . . . . . . 10
4.4.1. ICE Candidate Trickling . . . . . . . . . . . . . . . 10 3.4. ICE . . . . . . . . . . . . . . . . . . . . . . . . . . . 10
4.4.1.1. ICE Candidate Format . . . . . . . . . . . . . . . 10 3.4.1. ICE Candidate Trickling . . . . . . . . . . . . . . . 10
4.5. Interactions With Forking . . . . . . . . . . . . . . . . 11 3.4.1.1. ICE Candidate Format . . . . . . . . . . . . . . . 11
4.5.1. Sequential Forking . . . . . . . . . . . . . . . . . . 11 3.5. Interactions With Forking . . . . . . . . . . . . . . . . 11
4.5.2. Parallel Forking . . . . . . . . . . . . . . . . . . . 12 3.5.1. Sequential Forking . . . . . . . . . . . . . . . . . . 12
4.6. Session Rehydration . . . . . . . . . . . . . . . . . . . 13 3.5.2. Parallel Forking . . . . . . . . . . . . . . . . . . . 12
5. Interface . . . . . . . . . . . . . . . . . . . . . . . . . . 14 3.6. Session Rehydration . . . . . . . . . . . . . . . . . . . 13
5.1. SDP Requirements . . . . . . . . . . . . . . . . . . . . . 14 4. Interface . . . . . . . . . . . . . . . . . . . . . . . . . . 14
5.2. Methods . . . . . . . . . . . . . . . . . . . . . . . . . 15 4.1. SDP Requirements . . . . . . . . . . . . . . . . . . . . . 14
5.2.1. createOffer . . . . . . . . . . . . . . . . . . . . . 15 4.2. Methods . . . . . . . . . . . . . . . . . . . . . . . . . 15
5.2.2. createAnswer . . . . . . . . . . . . . . . . . . . . . 15 4.2.1. createOffer . . . . . . . . . . . . . . . . . . . . . 15
5.2.3. SessionDescriptionType . . . . . . . . . . . . . . . . 16 4.2.2. createAnswer . . . . . . . . . . . . . . . . . . . . . 16
5.2.3.1. Creating Answers . . . . . . . . . . . . . . . . . 17 4.2.3. SessionDescriptionType . . . . . . . . . . . . . . . . 17
5.2.4. setLocalDescription . . . . . . . . . . . . . . . . . 17 4.2.3.1. Use of Provisional Answers . . . . . . . . . . . . 18
5.2.5. setRemoteDescription . . . . . . . . . . . . . . . . . 18 4.2.3.2. Rollback . . . . . . . . . . . . . . . . . . . . . 18
5.2.6. localDescription . . . . . . . . . . . . . . . . . . . 18 4.2.4. setLocalDescription . . . . . . . . . . . . . . . . . 19
5.2.7. remoteDescription . . . . . . . . . . . . . . . . . . 18 4.2.5. setRemoteDescription . . . . . . . . . . . . . . . . . 19
5.2.8. updateIce . . . . . . . . . . . . . . . . . . . . . . 18 4.2.6. localDescription . . . . . . . . . . . . . . . . . . . 20
5.2.9. addIceCandidate . . . . . . . . . . . . . . . . . . . 19 4.2.7. remoteDescription . . . . . . . . . . . . . . . . . . 20
6. Configurable SDP Parameters . . . . . . . . . . . . . . . . . 20 4.2.8. updateIce . . . . . . . . . . . . . . . . . . . . . . 20
7. Security Considerations . . . . . . . . . . . . . . . . . . . 21 4.2.9. addIceCandidate . . . . . . . . . . . . . . . . . . . 21
8. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 22 5. SDP Interaction Procedures . . . . . . . . . . . . . . . . . . 21
5.1. Constructing an Offer . . . . . . . . . . . . . . . . . . 21
5.2. Generating an Answer . . . . . . . . . . . . . . . . . . . 21
5.3. Parsing an Offer . . . . . . . . . . . . . . . . . . . . . 21
5.4. Parsing an Answer . . . . . . . . . . . . . . . . . . . . 21
5.5. Applying a Local Description . . . . . . . . . . . . . . . 21
5.6. Applying a Remote Description . . . . . . . . . . . . . . 21
6. Configurable SDP Parameters . . . . . . . . . . . . . . . . . 21
7. Security Considerations . . . . . . . . . . . . . . . . . . . 22
8. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 23
9. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 23 9. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 23
10. References . . . . . . . . . . . . . . . . . . . . . . . . . . 24 10. References . . . . . . . . . . . . . . . . . . . . . . . . . . 23
10.1. Normative References . . . . . . . . . . . . . . . . . . . 24 10.1. Normative References . . . . . . . . . . . . . . . . . . . 23
10.2. Informative References . . . . . . . . . . . . . . . . . . 24 10.2. Informative References . . . . . . . . . . . . . . . . . . 24
Appendix A. JSEP Implementation Examples . . . . . . . . . . . . 26 Appendix A. JSEP Implementation Examples . . . . . . . . . . . . 25
A.1. Example API Flows . . . . . . . . . . . . . . . . . . . . 26 A.1. Example API Flows . . . . . . . . . . . . . . . . . . . . 25
A.1.1. Call using ROAP . . . . . . . . . . . . . . . . . . . 26 A.1.1. Call using ROAP . . . . . . . . . . . . . . . . . . . 26
A.1.2. Call using XMPP . . . . . . . . . . . . . . . . . . . 27 A.1.2. Call using XMPP . . . . . . . . . . . . . . . . . . . 26
A.1.3. Adding video to a call, using XMPP . . . . . . . . . . 28 A.1.3. Adding video to a call, using XMPP . . . . . . . . . . 28
A.1.4. Simultaneous add of video streams, using XMPP . . . . 28 A.1.4. Simultaneous add of video streams, using XMPP . . . . 28
A.1.5. Call using SIP . . . . . . . . . . . . . . . . . . . . 29 A.1.5. Call using SIP . . . . . . . . . . . . . . . . . . . . 29
A.1.6. Handling early media (e.g. 1-800-GO FEDEX), using A.1.6. Handling early media (e.g. 1-800-GO FEDEX), using
SIP . . . . . . . . . . . . . . . . . . . . . . . . . 30 SIP . . . . . . . . . . . . . . . . . . . . . . . . . 30
Appendix B. Change log . . . . . . . . . . . . . . . . . . . . . 32 Appendix B. Change log . . . . . . . . . . . . . . . . . . . . . 31
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 33 Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 32
1. Introduction 1. Introduction
This document describes how the W3C WEBRTC RTCPeerConnection
interface[W3C.WD-webrtc-20111027] is used to control the setup,
management and teardown of a multimedia session.
1.1. General Design of JSEP
The thinking behind WebRTC call setup has been to fully specify and 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 control the media plane, but to leave the signaling plane up to the
application as much as possible. The rationale is that different application as much as possible. The rationale is that different
applications may prefer to use different protocols, such as the applications may prefer to use different protocols, such as the
existing SIP or Jingle call signaling protocols, or something custom existing SIP or Jingle call signaling protocols, or something custom
to the particular application, perhaps for a novel use case. In this to the particular application, perhaps for a novel use case. In this
approach, the key information that needs to be exchanged is the approach, the key information that needs to be exchanged is the
multimedia session description, which specifies the necessary multimedia session description, which specifies the necessary
transport and media configuration information necessary to establish transport and media configuration information necessary to establish
the media plane. the media plane.
The browser environment also has its own challenges that cause The browser environment also has its own challenges that pose
problems for an embedded signaling state machine. One of these is 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, that the user may reload the web page at any time. If the browser is
and the state machine is being run at a server, the server can simply fully in charge of the signaling state, this will result in the loss
push the current state back down to the page and resume the call of the call when this state is wiped by the reload. However, if the
where it left off. state can be stored at the server, and pushed back down to the new
page, the call can be resumed with minimal interruption.
This document describes the Javascript Session Establishment Protocol With these considerations in mind, this document describes the
(JSEP) that pulls the signaling state machine out of the browser and Javascript Session Establishment Protocol (JSEP) that allows for full
into Javascript. This mechanism effectively removes the browser control of the signaling state machine from Javascript. This
almost completely from the core signaling flow; the only interface mechanism effectively removes the browser almost completely from the
needed is a way for the application to pass in the local and remote core signaling flow; the only interface needed is a way for the
session descriptions negotiated by whatever signaling mechanism is application to pass in the local and remote session descriptions
used, and a way to interact with the ICE state machine. negotiated by whatever signaling mechanism is used, and a way to
interact with the ICE state machine.
In this document, the use of JSEP is described as if it always occurs
between two browsers. Note though in many cases it will actually be
between a browser and some kind of server, such as a gateway or MCU.
This distinction is invisible to the browser; it just follows the
instructions it is given via the API.
JSEP's handling of session descriptions is simple and JSEP's handling of session descriptions is simple and
straightforward. Whenever an offer/answer exchange is needed, the straightforward. Whenever an offer/answer exchange is needed, the
initiating side creates an offer by calling a createOffer() API. The initiating side creates an offer by calling a createOffer() API. The
application optionally modifies that offer, and then uses it to set application optionally modifies that offer, and then uses it to set
up its local config via the setLocalDescription() API. The offer is up its local config via the setLocalDescription() API. The offer is
then sent off to the remote side over its preferred signaling then sent off to the remote side over its preferred signaling
mechanism (e.g., WebSockets); upon receipt of that offer, the remote mechanism (e.g., WebSockets); upon receipt of that offer, the remote
party installs it using the setRemoteDescription() API. party installs it using the setRemoteDescription() API.
When the call is accepted, the callee uses the createAnswer() API to When the call is accepted, the callee uses the createAnswer() API to
generate an appropriate answer, applies it using generate an appropriate answer, applies it using
setLocalDescription(), and sends the answer back to the initiator setLocalDescription(), and sends the answer back to the initiator
over the signaling channel. When the offerer gets that answer, it over the signaling channel. When the offerer gets that answer, it
installs it using setRemoteDescription(), and initial setup is installs it using setRemoteDescription(), and initial setup is
complete. This process can be repeated for additional offer/answer complete. This process can be repeated for additional offer/answer
exchanges. exchanges.
Regarding ICE, JSEP decouples the ICE state machine from the overall Regarding ICE [RFC5245], JSEP decouples the ICE state machine from
signaling state machine, as the ICE state machine must remain in the the overall signaling state machine, as the ICE state machine must
browser, because only the browser has the necessary knowledge of remain in the browser, because only the browser has the necessary
candidates and other transport info. Performing this separation also knowledge of candidates and other transport info. Performing this
provides additional flexibility; in protocols that decouple session separation also provides additional flexibility; in protocols that
descriptions from transport, such as Jingle, the transport decouple session descriptions from transport, such as Jingle, the
information can be sent separately; in protocols that don't, such as transport information can be sent separately; in protocols that
SIP, the information can be used in the aggregated form. Sending don't, such as SIP, the information can be used in the aggregated
transport information separately can allow for faster ICE and DTLS form. Sending transport information separately can allow for faster
startup, since the necessary roundtrips can occur while waiting for ICE and DTLS startup, since the necessary roundtrips can occur while
the remote side to accept the session. waiting for the remote side to accept the session.
The JSEP approach does come with a minor downside. As the Through its abstraction of signaling, the JSEP approach does require
application now is responsible for driving the signaling state the application to be aware of the signaling process. While the
machine, slightly more application code is necessary to perform call application does not need to understand the contents of session
setup; the application must call the right APIs at the right times, descriptions to set up a call, the application must call the right
and convert the session descriptions and ICE information into the APIs at the right times, convert the session descriptions and ICE
defined messages of its chosen signaling protocol, instead of simply information into the defined messages of its chosen signaling
forwarding the messages emitted from the browser. protocol, and perform the reverse conversion on the messages it
receives from the other side.
One way to mitigate this is to provide a Javascript library that One way to mitigate this is to provide a Javascript library that
hides this complexity from the developer, which would implement the hides this complexity from the developer; said library would
state machine and serialization of the desired signaling protocol. implement a given signaling protocol along with its state machine and
For example, this library could convert easily adapt the JSEP API serialization code, presenting a higher level call-oriented interface
into the exact ROAP API [I-D.jennings-rtcweb-signaling], thereby to the application developer. For example, this library could easily
implementing the ROAP signaling protocol. Such a library could of adapt the JSEP API into the API that was proposed for the ROAP
course also implement other popular signaling protocols, including signaling protocol [I-D.jennings-rtcweb-signaling], which would
SIP or Jingle. In this fashion we can enable greater control for the perform a ROAP call setup under the covers, interacting with the
experienced developer without forcing any additional complexity on application only when it needs a signaling message to be sent. In
the novice developer. the same fashion, one could also implement other popular signaling
protocols, including SIP or Jingle. This allow JSEP to provide
greater control for the experienced developer without forcing any
additional complexity on the novice developer.
2. Other Approaches Considered 1.2. Other Approaches Considered
Another approach that was considered for JSEP was to move the One approach that was considered instead of JSEP was to include a
mechanism for generating offers and answers out of the browser as lightweight signaling protocol. Instead of providing session
well. Instead of providing createOffer/createAnswer methods within descriptions to the API, the API would produce and consume messages
the browser, this approach would instead expose a getCapabilities API from this protocol. While providing a more high-level API, this put
which would provide the application with the information it needed in more control of signaling within the browser, forcing the browser to
order to generate its own session descriptions. This increases the have to understand and handle concepts like signaling glare. In
amount of work that the application needs to do; it needs to know how addition, it prevented the application from driving the state machine
to generate session descriptions from capabilities, and especially to a desired state, as is needed in the page reload case.
how to generate the correct answer from an arbitrary offer and the
A second approach that was considered but not chosen was to decouple
the management of the media control objects from session
descriptions, instead offering APIs that would control each component
directly. This was rejected based on a feeling that requiring
exposure of this level of complexity to the application programmer
would not be beneficial; it would result in an API where even a
simple example would require a significant amount of code to
orchestrate all the needed interactions, as well as creating a large
API surface that needed to be agreed upon and documented. In
addition, these API points could be called in any order, resulting in
a more complex set of interactions with the media subsystem than the
JSEP approach, which specifies how session descriptions are to be
evaluated and applied.
One variation on JSEP that was considered was to keep the basic
session description-oriented API, but to move the mechanism for
generating offers and answers out of the browser. Instead of
providing createOffer/createAnswer methods within the browser, this
approach would instead expose a getCapabilities API which would
provide the application with the information it needed in order to
generate its own session descriptions. This increases the amount of
work that the application needs to do; it needs to know how to
generate session descriptions from capabilities, and especially how
to generate the correct answer from an arbitrary offer and the
supported capabilities. While this could certainly be addressed by supported capabilities. While this could certainly be addressed by
using a library like the one mentioned above, it basically forces the using a library like the one mentioned above, it basically forces the
use of said library even for a simple example. Exposing createOffer/ use of said library even for a simple example. Providing
createAnswer avoids that problem, but still allows applications to createOffer/createAnswer avoids this problem, but still allows
generate their own offers/answers if they choose, using the applications to generate their own offers/answers (to a large extent)
description generated by createOffer as an indication of the if they choose, using the description generated by createOffer as an
browser's capabilities. 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.
3. Terminology 2. Terminology
The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT", The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
"SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
document are to be interpreted as described in RFC 2119 [RFC2119]. document are to be interpreted as described in [RFC2119].
4. Semantics and Syntax 3. Semantics and Syntax
4.1. Signaling Model 3.1. Signaling Model
JSEP does not specify a particular signaling model or state machine, JSEP does not specify a particular signaling model or state machine,
other than the generic need to exchange RFC 3264 offers and answers other than the generic need to exchange SDP media descriptions in the
in order for both sides of the session to know how to conduct the fashion described by [RFC3264] (offer/answer) in order for both sides
session. JSEP provides mechanisms to create offers and answers, as of the session to know how to conduct the session. JSEP provides
well as to apply them to a session. However, the actual mechanism by mechanisms to create offers and answers, as well as to apply them to
which these offers and answers are communicated to the remote side, a session. However, the browser is totally decoupled from the actual
including addressing, retransmission, forking, and glare handling, is mechanism by which these offers and answers are communicated to the
left entirely up to the application. remote side, including addressing, retransmission, forking, and glare
handling. These issues are left entirely up to the application; the
application has complete control over which offers and answers get
handed to the browser, and when.
+-----------+ +-----------+ +-----------+ +-----------+
| Web App |<--- App-Specific Signaling -->| Web App | | Web App |<--- App-Specific Signaling -->| Web App |
+-----------+ +-----------+ +-----------+ +-----------+
^ ^ ^ ^
| SDP | SDP | SDP | SDP
V V V V
+-----------+ +-----------+ +-----------+ +-----------+
| Browser |<----------- Media ------------>| Browser | | Browser |<----------- Media ------------>| Browser |
+-----------+ +-----------+ +-----------+ +-----------+
Figure 1: JSEP Signaling Model Figure 1: JSEP Signaling Model
4.2. Session Descriptions and State Machine 3.2. Session Descriptions and State Machine
In order to establish the media plane, the user agent needs specific In order to establish the media plane, the user agent needs specific
parameters to indicate what to transmit to the remote side, as well parameters to indicate what to transmit to the remote side, as well
as how to handle the media that is received. These parameters are as how to handle the media that is received. These parameters are
determined by the exchange of session descriptions in offers and determined by the exchange of session descriptions in offers and
answers, and there are certain details to this process that must be answers, and there are certain details to this process that must be
handled in the JSEP APIs. handled in the JSEP APIs.
Whether a session description was sent or received affects the Whether a session description applies to the local side or the remote
meaning of that description. For example, the list of codecs sent to side affects the meaning of that description. For example, the list
a remote party indicates what the local side is willing to decode, of codecs sent to a remote party indicates what the local side is
and what the remote party should send. Not all parameters follow willing to receive, which, when intersected with the set of codecs
this rule; for example, the SRTP parameters [RFC4568] sent to a the remote side supports, specifies what the remote side should send.
remote party indicate what the local side will use to encrypt, and However, not all parameters follow this rule; for example, the SRTP
thereby how the remote party should expect to receive. parameters [RFC4568] sent to a remote party indicate what the local
side will use to encrypt, and thereby what the remote party should
expect to receive; the remote party will have to accept these
parameters, with no option to choose a different value.
In addition, various RFCs put different conditions on the format of In addition, various RFCs put different conditions on the format of
offers versus answers. For example, a offer may propose multiple offers versus answers. For example, a offer may propose multiple
SRTP configurations, but an answer may only contain a single SRTP SRTP configurations, but an answer may only contain a single SRTP
configuration. configuration.
Lastly, while the exact media parameters are only known only after a Lastly, while the exact media parameters are only known only after a
offer and an answer have been exchanged, it is possible for the offer and an answer have been exchanged, it is possible for the
offerer to receive media after they have sent an offer and before offerer to receive media after they have sent an offer and before
they have received an answer. To properly process incoming media in they have received an answer. To properly process incoming media in
this case, the offerer's media handler must be aware of the details this case, the offerer's media handler must be aware of the details
of the offerer before the answer arrives. of the offer before the answer arrives.
Therefore, in order to handle session descriptions properly, the user Therefore, in order to handle session descriptions properly, the user
agent needs: agent needs:
1. To know if a session description pertains to the local or remote 1. To know if a session description pertains to the local or remote
side. side.
2. To know if a session description is an offer or an answer. 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. 3. To allow the offer to be specified independently of the answer.
JSEP addresses this by adding both a setLocalDescription and a JSEP addresses this by adding both a setLocalDescription and a
setRemoteDescription method and having session description objects setRemoteDescription method and having session description objects
contain a type field indicating the type of session description being contain a type field indicating the type of session description being
supplied. This satisfies the requirements listed above for both the supplied. This satisfies the requirements listed above for both the
offerer, who first calls setLocalDescription(sdp [offer]) and then offerer, who first calls setLocalDescription(sdp [offer]) and then
later setRemoteDescription(sdp [answer]), as well as for the later setRemoteDescription(sdp [answer]), as well as for the
answerer, who first calls setRemoteDescription(sdp [offer]) and then answerer, who first calls setRemoteDescription(sdp [offer]) and then
later setLocalDescription(sdp [answer]). While it could be possible later setLocalDescription(sdp [answer]).
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.
JSEP also allows for an answer to be treated as provisional by the JSEP also allows for an answer to be treated as provisional by the
application. Provisional answers provide a way for an answerer to application. Provisional answers provide a way for an answerer to
communicate initial session parameters back to the offerer, in order communicate initial session parameters back to the offerer, in order
to allow the session to begin, while allowing a final answer to be to allow the session to begin, while allowing a final answer to be
specified later. This concept of a final answer is important to the specified later. This concept of a final answer is important to the
offer/answer model; when such an answer is received, any extra offer/answer model; when such an answer is received, any extra
resources allocated by the caller can be released, now that the exact resources allocated by the caller can be released, now that the exact
session configuration is known. These "resources" can include things session configuration is known. These "resources" can include things
like extra ICE components, TURN candidates, or video decoders. like extra ICE components, TURN candidates, or video decoders.
Provisional answers, on the other hand, do no such deallocation Provisional answers, on the other hand, do no such deallocation
results; as a result, multiple dissimilar provisional answers can be results; as a result, multiple dissimilar provisional answers can be
received and applied during call setup. received and applied during call setup.
In [RFC3264], the constraints at the signaling level is that only one In [RFC3264], the constraint at the signaling level is that only one
offer can be outstanding for a given session but from the media stack offer can be outstanding for a given session, but from the media
level, a new offer can be generated at any point. For example, when stack level, a new offer can be generated at any point. For example,
using SIP for signaling, if one offer is sent, then cancelled using a when using SIP for signaling, if one offer is sent, then cancelled
SIP CANCEL, another offer can be generated even though no answer was using a SIP CANCEL, another offer can be generated even though no
received for the first offer. To support this, the JSEP media layer answer was received for the first offer. To support this, the JSEP
can provide an offer whenever the Javascript application needs one media layer can provide an offer whenever the Javascript application
for the signaling. The answerer can send back zero or more needs one for the signaling. The answerer can send back zero or more
provisional answers, and finally end the offer-answer exchange by provisional answers, and finally end the offer-answer exchange by
sending a final answer. The state machine for this is as follows: sending a final answer. The state machine for this is as follows:
+-----------+ setRemote(OFFER) setLocal(PRANSWER)
| | /-----\ /-----\
| | | | | |
| Stable |<---------------\ v | v |
| | | +---------------+ | +---------------+ |
| | | | |----/ | |----/
+-----------+ | | | setLocal(PRANSWER) | |
^ | | | Remote-Offer |------------------- >| Local-Pranswer|
| | OFFER | | | | |
ANSWER | | | ANSWER | | | |
| V | +---------------+ +---------------+
+-----------+ +-----------+ ^ | |
| | | | | | setLocal(ANSWER) |
| | PRANSWER | | setRemote(OFFER) | | |
| Offer |-------- >| Pranswer | | V setLocal(ANSWER) |
| | | | +---------------+ |
| |----\ | |----\ | | |
+-----------+ | +-----------+ | | | |
^ | ^ | | Stable |<---------------------------+
| | | | | | |
\-----/ \-----/ | | |
OFFER PRANSWER +---------------+ setRemote(ANSWER) |
^ | |
| | setLocal(OFFER) |
setRemote(ANSWER)| | |
| V |
+---------------+ +---------------+
| | | |
| | setRemote(PRANSWER) | |
| Local-Offer |------------------- >|Remote-Pranswer|
| | | |
| |----\ | |----\
+---------------+ | +---------------+ |
^ | ^ |
| | | |
\-----/ \-----/
setLocal(OFFER) setRemote(PRANSWER)
Figure 2: JSEP State Machine Figure 2: JSEP State Machine
Aside from these state transitions, there is no other difference Aside from these state transitions, there is no other difference
between the handling of provisional ("pranswer") and final ("answer") between the handling of provisional ("pranswer") and final ("answer")
answers. answers.
4.3. Session Description Format 3.3. Session Description Format
In the WebRTC specification, session descriptions are formatted as In the WebRTC specification, session descriptions are formatted as
SDP messages. While this format is not optimal for manipulation from SDP messages. While this format is not optimal for manipulation from
Javascript, it is widely accepted, and frequently updated with new Javascript, it is widely accepted, and frequently updated with new
features. Any alternate encoding of session descriptions would have features. Any alternate encoding of session descriptions would have
to keep pace with the changes to SDP, at least until the time that 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 this new encoding eclipsed SDP in popularity. As a result, JSEP
continues to use SDP as the internal representation for its session currently uses SDP as the internal representation for its session
descriptions. descriptions.
However, to simplify Javascript processing, and provide for future However, to simplify Javascript processing, and provide for future
flexibility, the SDP syntax is encapsulated within a flexibility, the SDP syntax is encapsulated within a
SessionDescription object, which can be constructed from SDP, and be SessionDescription object, which can be constructed from SDP, and be
serialized out to SDP. If future specifications agree on a JSON serialized out to SDP. If future specifications agree on a JSON
format for session descriptions, we could easily enable this object format for session descriptions, we could easily enable this object
to generate and consume that JSON. to generate and consume that JSON.
Other methods may be added to SessionDescription in the future to Other methods may be added to SessionDescription in the future to
simplify handling of SessionDescriptions from Javascript. Though it simplify handling of SessionDescriptions from Javascript. In the
is unclear exactly what manipulations developer will commonly want to meantime, it would be simple to write a Javascript library to perform
do to SDP, it would be simple to write a Javascript library to these manipulations.
perform these manipulations.
4.4. ICE 3.4. ICE
When a new ICE candidate is available, the ICE Agent will notify the When a new ICE candidate is available, the ICE Agent will notify the
application via a callback; these candidates will automatically be application via a callback; these candidates will automatically be
added to the local session description. When all candidates have added to the local session description. When all candidates have
been gathered, the callback will also be invoked to signal that the been gathered, the callback will also be invoked to signal that the
gathering process is complete. gathering process is complete.
4.4.1. ICE Candidate Trickling 3.4.1. ICE Candidate Trickling
Candidate trickling is a technique through which a caller may Candidate trickling is a technique through which a caller may
incrementally provide candidates to the callee after the initial incrementally provide candidates to the callee after the initial
offer has been dispatched; the semantics of "Trickle ICE" are defined offer has been dispatched; the semantics of "Trickle ICE" are defined
in [I-D.rescorla-mmusic-ice-trickle]. This process allows the callee in [I-D.rescorla-mmusic-ice-trickle]. This process allows the callee
to begin acting upon the call and setting up the ICE (and perhaps to begin acting upon the call and setting up the ICE (and perhaps
DTLS) connections immediately, without having to wait for the caller DTLS) connections immediately, without having to wait for the caller
to gather all possible candidates. This results in faster call to gather all possible candidates. This results in faster call
startup in cases where gathering is not performed prior to initating startup in cases where gathering is not performed prior to initating
the call. the call.
skipping to change at page 10, line 47 skipping to change at page 11, line 14
offer immediately and send individual candidates when they get the offer immediately and send individual candidates when they get the
notified of a new candidate; applications that do not support this notified of a new candidate; applications that do not support this
feature can simply wait for the indication that gathering is feature can simply wait for the indication that gathering is
complete, and then create and send their offer, with all the complete, and then create and send their offer, with all the
candidates, at this time. candidates, at this time.
Upon receipt of trickled candidates, the receiving application will Upon receipt of trickled candidates, the receiving application will
supply them to its ICE Agent. This triggers the ICE Agent to start supply them to its ICE Agent. This triggers the ICE Agent to start
using the new remote candidates for connectivity checks. using the new remote candidates for connectivity checks.
4.4.1.1. ICE Candidate Format 3.4.1.1. ICE Candidate Format
As with session descriptions, the syntax of the IceCandidate object As with session descriptions, the syntax of the IceCandidate object
provides some abstraction, but can be easily converted to and from provides some abstraction, but can be easily converted to and from
the SDP a=candidate lines. the SDP candidate lines.
The a=candidate lines are the only SDP information that is contained The candidate lines are the only SDP information that is contained
within IceCandidate, as they represent the only information needed within IceCandidate, as they represent the only information needed
that is not present in the initial offer (i.e. for trickle that is not present in the initial offer (i.e. for trickle
candidates). This information is carried with the same syntax as the candidates). This information is carried with the same syntax as the
"a=candidate" line in SDP. For example: "candidate-attribute" field defined for ICE. For example:
a=candidate:1 1 UDP 1694498815 192.0.2.33 10000 typ host candidate:1 1 UDP 1694498815 192.0.2.33 10000 typ host
The IceCandidate object also contains fields to indicate which m= The IceCandidate object also contains fields to indicate which m=
line it should be associated with. The m line can be identified in line it should be associated with. The m line can be identified in
one of two ways; either by a m-line index, or a MID. The m-line one of two ways; either by a m-line index, or a MID. The m-line
index is a zero-based index, referring to the Nth m-line in the SDP. index is a zero-based index, referring to the Nth m-line in the SDP.
The MID uses the "media stream identification", as defined in [RFC The MID uses the "media stream identification", as defined in
3388], to identify the m-line. WebRTC implementations creating an [RFC3388] , to identify the m-line. WebRTC implementations creating
ICE Candidate object MUST populate both of these fields. an ICE Candidate object MUST populate both of these fields.
Implementations receiving an ICE Candidate object SHOULD use the MID Implementations receiving an ICE Candidate object SHOULD use the MID
if they implement that functionality, or the m-line index, if not. if they implement that functionality, or the m-line index, if not.
4.5. Interactions With Forking 3.5. Interactions With Forking
Some call signaling systems allow various types of forking where an Some call signaling systems allow various types of forking where an
SDP Offer may be provided to more than one device. For example, SIP SDP Offer may be provided to more than one device. For example, SIP
RFC 3261 defines both a "Parallel Search" and "Sequential Search". [RFC3261] defines both a "Parallel Search" and "Sequential Search".
Although these are primarily signaling level issues that are outside Although these are primarily signaling level issues that are outside
the scope of JSEP, they do have some impact on the configuration of the scope of JSEP, they do have some impact on the configuration of
the media plane, which is relevant. When forking is happening at the the media plane which is relevant. When forking happens at the
signaling layer, the Javascript application responsible for the signaling layer, the Javascript application responsible for the
signaling needs to make the decisions about what media should be sent signaling needs to make the decisions about what media should be sent
or received at any point of time and which remote endpoint it should or received at any point of time, as well as which remote endpoint it
communicate with. JSEP is used to make sure the media engine can should communicate with; JSEP is used to make sure the media engine
make the RTP and media perform as required by the application. The can make the RTP and media perform as required by the application.
basic operations that the applications can have the media engine do The basic operations that the applications can have the media engine
are: do are:
Start exchanging media to a given remote peer, but keep all the
Start exchanging media to a given remote peer but keep all the
resources reserved in the offer. resources reserved in the offer.
Start exchanging media with a given remote peer, and free any
Start exchanging media with a given remote peer and free any
resources in the offer that are not being used. resources in the offer that are not being used.
4.5.1. Sequential Forking 3.5.1. Sequential Forking
Sequential forking involves a call being dispatched to multiple Sequential forking involves a call being dispatched to multiple
remote callees, where each callee can accept the call, but only one 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 active session ever exists at a time; no mixing of received media is
performed. performed.
JSEP handles serial forking well, allowing the application to easily JSEP handles sequential forking well, allowing the application to
control the policy for selecting the desired remote endpoint. When easily control the policy for selecting the desired remote endpoint.
an answer arrives from one of the callees, the application can choose When an answer arrives from one of the callees, the application can
to apply it either as a provisional answer, leaving open the 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 possibility of using a different answer in the future, or apply it as
a final answer, ending the setup flow. a final answer, ending the setup flow.
In a "first-one-wins" situation, the first answer will be applied as In a "first-one-wins" situation, the first answer will be applied as
a final answer, and the application will reject any subsequent a final answer, and the application will reject any subsequent
answers. In SIP parlance, this would be ACK + BYE. answers. In SIP parlance, this would be ACK + BYE.
In a "last-one-wins" situation, all answers would be applied as In a "last-one-wins" situation, all answers would be applied as
provisional answers, and any previous call leg will be terminated. provisional answers, and any previous call leg will be terminated.
At some point, the application will end the setup process, perhaps At some point, the application will end the setup process, perhaps
with a timer; at this point, the application could reapply the with a timer; at this point, the application could reapply the
existing remote description as a final answer. existing remote description as a final answer.
4.5.2. Parallel Forking 3.5.2. Parallel Forking
Parallel forking involves a call being dispatched to multiple remote Parallel forking involves a call being dispatched to multiple remote
callees, where each callee can accept the call, and multiple callees, where each callee can accept the call, and multiple
simultaneous active signaling sessions can be established as a simultaneous active signaling sessions can be established as a
result. If multiple callees send media at the same time, the result. If multiple callees send media at the same time, the
possibilities for handling this are described in Section 3.1 of RFC possibilities for handling this are described in Section 3.1 of
3960. Most SIP devices today only support exchanging media with a [RFC3960]. Most SIP devices today only support exchanging media with
single device at a time, and do not try to mix multiple early media a single device at a time, and do not try to mix multiple early media
audio sources, as that could result in a confusing situation. For audio sources, as that could result in a confusing situation. For
example. consider having a European ringback tone mixed together with example, consider having a European ringback tone mixed together with
the North American ringback tone - the resulting sound would not be the North American ringback tone - the resulting sound would not be
like either tone, and would confuse the user. If the signaling like either tone, and would confuse the user. If the signaling
application wishes to only exchange media with one of the remote application wishes to only exchange media with one of the remote
endpoints at a time, then from a media engine point of view, this is endpoints at a time, then from a media engine point of view, this is
exactly like the sequential forking case. exactly like the sequential forking case.
In the parallel forking case where the Javascript application wishes In the parallel forking case where the Javascript application wishes
to simultaneously exchange media with multiple peers, the flow is to simultaneously exchange media with multiple peers, the flow is
slightly more complex, but the Javascript application can follow the slightly more complex, but the Javascript application can follow the
strategy that RFC 3960 describes using UPDATE. (It is worth noting strategy that [RFC3960] describes using UPDATE. (It is worth noting
that use cases where this is the desired behavior are very unusual.) that use cases where this is the desired behavior are very unusual.)
The UPDATE approach allows the signaling to set up a separate media The UPDATE approach allows the signaling to set up a separate media
flow for each peer that it wishes to exchange media with. In JSEP, flow for each peer that it wishes to exchange media with. In JSEP,
this offer used in the UPDATE would be formed by simply creating a this offer used in the UPDATE would be formed by simply creating a
new PeerConnection and making sure that the same local media streams new PeerConnection and making sure that the same local media streams
have been added into this new PeerConnection. Then the new have been added into this new PeerConnection. Then the new
PeerConnection object would produce a SDP offer that could be used by PeerConnection object would produce a SDP offer that could be used by
the signaling to perform the UPDATE strategy discussed in RFC 3690. the signaling to perform the UPDATE strategy discussed in [RFC3960] .
As a result of sharing the media streams, the application will end up As a result of sharing the media streams, the application will end up
with N parallel PeerConnection sessions, each with a local and remote with N parallel PeerConnection sessions, each with a local and remote
description and their own local and remote addresses. The media flow description and their own local and remote addresses. The media flow
from these sessions can be managed by specifying SDP direction from these sessions can be managed by specifying SDP direction
attributes in the descriptions, or the application can choose to play attributes in the descriptions, or the application can choose to play
out the media from all sessions mixed together. Of course, if the out the media from all sessions mixed together. Of course, if the
application wants to only keep a single session, it can simply application wants to only keep a single session, it can simply
terminate the sessions that it no longer needs. terminate the sessions that it no longer needs.
4.6. Session Rehydration 3.6. Session Rehydration
In the event that the local application state is reinitialized, In the event that the local application state is reinitialized,
either due to a user reload of the page, or a decision within the 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 application to reload itself (perhaps to update to a new version), it
is possible to keep an existing session alive via a process called is possible to keep an existing session alive, via a process called
"rehydration". "rehydration". The explicit goal of rehydration is to carry out this
session resumption with no interaction with the remote side other
than normal call signaling messages.
With rehydration, the current signaling state is persisted somewhere With rehydration, the current signaling state is persisted somewhere
outside of the page, perhaps on the application server, or in browser outside of the page, perhaps on the application server, or in browser
local storage. The page is then reloaded, and a new session object local storage. The page is then reloaded, the saved signaling state
is created in Javascript. The saved signaling state is now is retrieved, and a new PeerConnection object is created for the
retrieved, and a new PeerConnection object is created for the session. The previously obtained MediaStreams are re-acquired, and
session. At this point a new offer can be generated by the new are given the same IDs as the original session; this ensures the IDs
PeerConnection, with new ICE and SDES credentials. This can then be in use by the remote side continue to work. Next, a new offer is
used to re-initiate the session with the existing remote endpoint, generated by the new PeerConnection; this offer will have new ICE and
who simply sees the new offer as an in-call renegotiation, and will possibly new SDES credentials (since the old ICE and SRTP state has
reply with an answer that can be supplied to setRemoteDescription. been lost). Finally, this offer is used to re-initiate the session
ICE processing proceeds as usual, and as soon as connectivity is with the existing remote endpoint, who simply sees the new offer as
established, the session will be back up and running again. an in-call renegotiation, and replies 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 [OPEN ISSUE: EKR proposed an alternative rehydration approach where
the actual internal PeerConnection object in the browser was kept the actual internal PeerConnection object in the browser was kept
alive for some time after the web page was killed and provided some alive for some time after the web page was killed and provided some
way for a new page to acquire the old PeerConnection object. way for a new page to acquire the old PeerConnection object.]
5. Interface 4. Interface
This section details the basic operations that must be present to This section details the basic operations that must be present to
implement JSEP functionality. The actual API exposed in the W3C API implement JSEP functionality. The actual API exposed in the W3C API
may have somewhat different syntax, but should map easily to these may have somewhat different syntax, but should map easily to these
concepts. concepts.
5.1. SDP Requirements 4.1. SDP Requirements
Note: The text in this section may not represent working group Note: The text in this section may not represent working group
consensus and is put here so that the working group can discuss it 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. and find out how to change it such that it does have consensus.
When generating SDP blobs, either for offers or answers, the When generating SDP blobs, either for offers or answers, the
generated SDP needs to conform to the following specifications. generated SDP needs to conform to the following specifications.
Similarly, in order to properly process received SDP blobs, Similarly, in order to properly process received SDP blobs,
implementations need to implement the functionality described in the implementations need to implement the functionality described in the
following specifications. This list is derived from following specifications. This list is derived from
[I-D.ietf-rtcweb-rtp-usage]. [I-D.ietf-rtcweb-rtp-usage].
R-1 [RFC4566] is the base SDP specification and MUST be
implemented.
R-2 The [RFC5888] grouping framework MUST be implemented for
signaling grouping information, and MUST be used to identify m=
lines via the a=mid attribute.
R-3 [RFC5124] MUST be supported for signaling RTP/SAVPF RTP
profile.
R-4 [RFC4585] MUST be implemented to signal RTCP based feedback.
R-5 [RFC5245] MUST be implemented for signaling the ICE candidate
lines corresponding to each media stream.
R-6 [RFC5761] MUST be implemented to signal multiplexing of RTP and
RTCP.
R-7 The SDP atributes of "sendonly", "recvonly", "inactive", and
"sendrecv" from [RFC4566] MUST be implemented to signal
information about media direction.
R-8 [RFC5576] MUST be implemented to signal RTP SSRC values. [OPEN
ISSUE; depends on BUNDLE and how we choose to represent
multiple media sources]
R-9 [RFC5763] MUST be implemented to signal DTLS certificate
fingerprints.
RFC4566 is the base SDP specification and MUST be implemented. R-10 [RFC5506] MAY be implemented to signal Reduced-Size RTCP
messages.
RFC5124 MUST be supported for signaling RTP/SAVPF RTP profile. R-11 [RFC3556] with bandwidth modifiers MAY be supported for
specifying RTCP bandwidth as a fraction of the media bandwidth,
RFC5104 MUST be implemented to signal RTCP based feedback. RTCP fraction allocated to the senders and setting maximum
media bit-rate boundaries.
RFC5761 MUST be implemented to signal multiplexing of RTP and R-12 [RFC4568] MAY be implemented to signal SDES SRTP keying
RTCP. information.
R-13 A TBD-draft MUST be supported, in order to signal associations
RFC5245 MUST be implemented for signaling the ICE candidate lines between RTP objects and W3C MediaStreams and MediaStreamTracks
corresponding to each media stream. in a standard way. Though there is not yet WG consensus in
this area, this TBD-draft is very likely to be
RFC3264 MUST be implemented to signal information about media [I-D.alvestrand-mmusic-msid].
direction. R-14 A TBD-draft MUST be supported to signal the use or multiplexing
RTP somethings on a single UDP port, in order to avoid
The RFC5888 grouping framework MUST be implemented for signaling excessive use of port number resources. Though there is not
the grouping information. yet WG consensus in this area, this TBD-draft is very likely to
be [I-D.holmberg-mmusic-sdp-bundle-negotiation].
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 As required by [RFC4566] Section 5.13 JSEP implementations MUST
unknown attributes (a=) lines. ignore unknown attributes (a=) lines.
Example SDP for RTCWeb call flows can be found in Example SDP for RTCWeb call flows can be found in
[I-D.nandakumar-rtcweb-sdp]. [I-D.nandakumar-rtcweb-sdp]. [TODO: since we are starting to
specify how to handle SDP in this document, should these call flows
be merged into this document, or this link moved to the examples
section?]
5.2. Methods 4.2. Methods
5.2.1. createOffer 4.2.1. createOffer
The createOffer method generates a blob of SDP that contains a RFC The createOffer method generates a blob of SDP that contains a
3264 offer with the supported configurations for the session, [RFC3264] offer with the supported configurations for the session,
including descriptions of the local MediaStreams attached to this including descriptions of the local MediaStreams attached to this
PeerConnection, the codec/RTP/RTCP options supported by this PeerConnection, the codec/RTP/RTCP options supported by this
implementation, and any candidates that have been gathered by the ICE implementation, and any candidates that have been gathered by the ICE
Agent. A constraints parameters may be supplied to provide Agent. A constraints parameters may be supplied to provide
additional control over the generated offer, e.g. to get a full set additional control over the generated offer. This constraints
of session capabilities, or to request a new set of ICE credentials. parameter should allow for the following manipulations to be
performed:
o To indicate support for a media type even if no MediaStreamTracks
of that type have been added to the session (e.g., an audio call
that wants to receive video.)
o To trigger an ICE restart, for the purpose of reestablishing
connectivity.
o For re-offer cases, to request an offer that contains the full set
of supported capabilities, as opposed to just the currently
negotiated parameters.
In the initial offer, the generated SDP will contain all desired In the initial offer, the generated SDP will contain all desired
functionality for the session (certain parts that are supported but functionality for the session (certain parts that are supported but
not desired by default may be omitted); for each SDP line, the not desired by default may be omitted); for each SDP line, the
generation of the SDP must follow the appropriate process for generation of the SDP must follow the process defined for generating
generating an offer. In the event createOffer is called after the an initial offer from the document (listed in Section 4.1) that
session is established, createOffer will generate an offer that is specifies the given SDP line.
compatible with the current session, incorporating any changes that
have been made to the session since the last complete offer-answer In the event createOffer is called after the session is established,
exchange, such as addition or removal of streams. If no changes have createOffer will generate an offer to modify the current session
been made, the offer will be identical to the current local based on any changes that have been made to the session, e.g. adding
description. or removing MediaStreams, or requesting an ICE restart. For each
existing stream, the generation of each SDP line must follow the
process defined for generating an updated offer from the document
that specfies the given SDP line. For each new stream, the
generation of the SDP must follow the process of generating an
initial offer, as mentioned above. If no changes have been made, or
for SDP lines that are unaffected by the requested changes, the offer
will only contain the parameters negotiated by the last offer-answer
exchange.
Session descriptions generated by createOffer must be immediately Session descriptions generated by createOffer must be immediately
usable by setLocalDescription; if a system has limited resources usable by setLocalDescription; if a system has limited resources
(e.g. a finite number of decoders), createOffer should return an (e.g. a finite number of decoders), createOffer should return an
offer that reflects the current state of the system, so that offer that reflects the current state of the system, so that
setLocalDescription will succeed when it attempts to acquire those setLocalDescription will succeed when it attempts to acquire those
resources. Because this method may need to inspect the system state resources. Because this method may need to inspect the system state
to determine the currently available resources, it may be implemented to determine the currently available resources, it may be implemented
as an async operation. as an async operation.
Calling this method may do things such as generate new ICE Calling this method may do things such as generate new ICE
credentials, but does not change media state. credentials, but does not result in candidate gathering, or cause
media to start or stop flowing.
5.2.2. createAnswer 4.2.2. createAnswer
The createAnswer method generates a blob of SDP that contains a RFC The createAnswer method generates a blob of SDP that contains a
3264 SDP answer with the supported configuration for the session that [RFC3264] SDP answer with the supported configuration for the session
is compatible with the parameters supplied in the offer. Like that is compatible with the parameters supplied in the offer. Like
createOffer, the returned blob contains descriptions of the local createOffer, the returned blob contains descriptions of the local
MediaStreams attached to this PeerConnection, the codec/RTP/RTCP MediaStreams attached to this PeerConnection, the codec/RTP/RTCP
options negotiated for this session, and any candidates that have options negotiated for this session, and any candidates that have
been gathered by the ICE Agent. A constraints parameter may be been gathered by the ICE Agent. A constraints parameter may be
supplied to provide additional control over the generated answer. supplied to provide additional control over the generated answer.
As an answer, the generated SDP will contain a specific configuration As an answer, the generated SDP will contain a specific configuration
that specifies how the media plane should be established. that specifies how the media plane should be established; for each
SDP line, the generation of the SDP must follow the process defined
for generating an answer from the document that specifies the given
SDP line.
Session descriptions generated by createAnswer must be immediately Session descriptions generated by createAnswer must be immediately
usable by setLocalDescription; like createOffer, the returned usable by setLocalDescription; like createOffer, the returned
description should reflect the current state of the system. Because description should reflect the current state of the system. Because
this method may need to inspect the system state to determine the this method may need to inspect the system state to determine the
currently available resources, it may need to be implemented as an currently available resources, it may need to be implemented as an
async operation. async operation.
Calling this method may do things such as generate new ICE Calling this method may do things such as generate new ICE
credentials, but does not change media state. credentials, but does not trigger candidate gathering or change media
state.
5.2.3. SessionDescriptionType 4.2.3. SessionDescriptionType
Session description objects (RTCSessionDescription) may be of type Session description objects (RTCSessionDescription) may be of type
"offer", "pranswer", and "answer". These types provide information "offer", "pranswer", and "answer". These types provide information
as to how the description parameter should be parsed, and how the as to how the description parameter should be parsed, and how the
media state should be changed. media state should be changed.
"offer" indicates that a description should be parsed as an offer; "offer" indicates that a description should be parsed as an offer;
said description may include many possible media configurations. A said description may include many possible media configurations. A
description used as an "offer" may be applied anytime the description used as an "offer" may be applied anytime the
PeerConnection is in a stable state, or as an update to a previously PeerConnection is in a stable state, or as an update to a previously
sent but unanswered "offer". supplied but unanswered "offer".
"pranswer" indicates that a description should be parsed as an "pranswer" indicates that a description should be parsed as an
answer, but not a final answer, and so should not result in the 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 freeing of allocated resources. It may result in the start of media
transmission, if the answer does not specify an inactive media transmission, if the answer does not specify an inactive media
direction. A description used as a "pranswer" may be applied as a direction. A description used as a "pranswer" may be applied as a
response to an "offer", or an update to a previously sent "answer". response to an "offer", or an update to a previously sent "answer".
"answer" indicates that a description should be parsed as an answer, "answer" indicates that a description should be parsed as an answer,
the offer-answer exchange should be considered complete, and any the offer-answer exchange should be considered complete, and any
resources (decoders, candidates) that are no longer needed can be resources (decoders, candidates) that are no longer needed can be
released. A description used as an "answer" may be applied as a released. A description used as an "answer" may be applied as a
response to a "offer", or an update to a previously sent "pranswer". response to a "offer", or an update to a previously sent "pranswer".
The application can use some discretion on whether an answer should The only difference between a provisional and final answer is that
be applied as provisional or final. For example, in a serial forking the final answer results in the freeing of any unused resources that
scenario, an application may receive multiple "final" answers, one were allocated as a result of the offer. As such, the application
from each remote endpoint. The application could accept the initial can use some discretion on whether an answer should be applied as
provisional or final, and can change the type of the session
description as needed. For example, in a serial forking scenario, an
application may receive multiple "final" answers, one from each
remote endpoint. The application could choose to accept the initial
answers as provisional answers, and only apply an answer as final answers as provisional answers, and only apply an answer as final
when it receives one that meets its criteria (e.g. a live user when it receives one that meets its criteria (e.g. a live user
instead of voicemail). instead of voicemail).
5.2.3.1. Creating Answers 4.2.3.1. Use of Provisional Answers
Most web applications will not need to create answers using the Most web applications will not need to create answers using the
"pranswer" type. The general recommendation for a web application "pranswer" type. The preferred handling for a web application would
would be to create an answer more or less immediately after receiving be to create and send an "inactive" answer more or less immediately
the offer, instead of waiting for a human user to provide input. after receiving the offer, instead of waiting for a human user to
Later when the human input is received, the applications can create a physically answer the call. Later, when the human input is received,
new offer to update the previous offer/answer pair. Some the application can create a new "sendrecv" offer to update the
applications may not be able to do this, particularly ones that Some previous offer/answer pair and start the media flow. This approach
application may not be able to do this, particular ones that are is preferred because it minimizes the amount of time that the offer-
attempting to gateway to other signaling protocols. answer exchange is left open, in addition to avoiding media clipping
by ensuring the transport is ready to go by the time the call is
phyiscally answered. However, some applications may not be able to
do this, particularly ones that are attempting to gateway to other
signaling protocols. In these cases, "pranswer" can still allow the
application to warm up the transport.
Consider a typical web application that will set up a data channel, Consider a typical web application that will set up a data channel,
an audio channel, and a video channel. When an endpoint receives an an audio channel, and a video channel. When an endpoint receives an
offer with these channels, it could send an answer accepting the data 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 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 inactive or receive-only. It could then ask the user to accept the
audio and video to the far end, acquire the local media streams, and call, acquire the local media streams, and send a new offer to the
send a new offer to the remote side moving the audio and video to be remote side moving the audio and video to be two-way media. By the
two-way media. By the time the human has authorized sending media, time the human has accepted the call and sent the new offer, it is
it is likely that the ICE and DTLS handshaking with the remote side likely that the ICE and DTLS handshaking for all the channels will
will already be set up. already be set up.
5.2.4. setLocalDescription 4.2.3.2. Rollback
In certain situations it may be desirable to "undo" a change made to
setLocalDescription or setRemoteDescription. Consider a case where a
call is ongoing, and one side wants to change some of the session
parameters; that side generates an updated offer and then calls
setLocalDescription. However, the remote side, either before or
after setRemoteDescription, decides it does not want to accept the
new parameters, and sends a reject message back to the offerer. Now,
the offerer, and possibly the answerer as well, need to return to a
stable state and the previous local/remote description. To support
this, we introduce the concept of "rollback".
A rollback returns the state machine to its previous state, and the
local or remote description to its previous value. Any resources or
candidates that were allocated by the new local description are
discarded; any media that is received will be processed according to
the previous session description.
A rollback is performed by supplying a session description of type
"rollback" to either setLocalDescription or setRemoteDescription,
depending on which needs to be rolled back (i.e. if the new offer was
supplied to setLocalDescription, the rollback should be done on
setLocalDescription as well.)
4.2.4. setLocalDescription
The setLocalDescription method instructs the PeerConnection to apply The setLocalDescription method instructs the PeerConnection to apply
the supplied SDP blob as its local configuration. The type field the supplied SDP blob as its local configuration. The type field
indicates whether the blob should be processed as an offer, indicates whether the blob should be processed as an offer,
provisional answer, or final answer; offers and answers are checked provisional answer, or final answer; offers and answers are checked
differently, using the various rules that exist for each SDP line. differently, using the various rules that exist for each SDP line.
This API changes the local media state; among other things, it sets This API changes the local media state; among other things, it sets
up local resources for receiving and decoding media. In order to up local resources for receiving and decoding media. In order to
successfully handle scenarios where the application wants to offer to successfully handle scenarios where the application wants to offer to
change from one media format to a different, incompatible format, the change from one media format to a different, incompatible format, the
PeerConnection must be able to simultaneously support use of both the PeerConnection must be able to simultaneously support use of both the
old and new local descriptions (e.g. support codecs that exist in old and new local descriptions (e.g. support codecs that exist in
both descriptions) until a final answer is received, at which point both descriptions) until a final answer is received, at which point
the PeerConnection can fully adopt the new local description, or roll the PeerConnection can fully adopt the new local description, or roll
back to the old description if the remote side denied the change. back to the old description if the remote side denied the change.
This API indirectly controls the candidate gathering process. When a
local description is supplied, and the number of transports currently
in use does not match the number of transports needed by the local
description, the PeerConnection will create transports as needed and
begin gathering candidates for them.
If setRemoteDescription was previous called with an offer, and If setRemoteDescription was previous called with an offer, and
setLocalDescription is called with an answer (provisional or final), setLocalDescription is called with an answer (provisional or final),
and the media directions are compatible, this will result in the and the media directions are compatible, and media are available to
starting of media transmission. send, this will result in the starting of media transmission.
5.2.5. setRemoteDescription 4.2.5. setRemoteDescription
The setRemoteDescription method instructs the PeerConnection to apply The setRemoteDescription method instructs the PeerConnection to apply
the supplied SDP blob as the desired remote configuration. As in the supplied SDP blob as the desired remote configuration. As in
setLocalDescription, the type field of the indicates how the blob setLocalDescription, the type field of the indicates how the blob
should be processed. should be processed.
This API changes the local media state; among other things, it sets This API changes the local media state; among other things, it sets
up local resources for sending and encoding media. up local resources for sending and encoding media.
If setRemoteDescription was previous called with an offer, and If setRemoteDescription was previously called with an offer, and
setLocalDescription is called with an answer (provisional or final), setLocalDescription is called with an answer (provisional or final),
and the media directions are compatible, this will result in the and the media directions are compatible, and media are available to
starting of media transmission. send, this will result in the starting of media transmission.
5.2.6. localDescription 4.2.6. localDescription
The localDescription method returns a copy of the current local The localDescription method returns a copy of the current local
configuration, i.e. what was most recently passed to configuration, i.e. what was most recently passed to
setLocalDescription, plus any local candidates that have been setLocalDescription, plus any local candidates that have been
generated by the ICE Agent. generated by the ICE Agent.
TODO: Do we need to expose accessors for both the current and
proposed local description?
A null object will be returned if the local description has not yet A null object will be returned if the local description has not yet
been established. been established, or if the PeerConnection has been closed.
5.2.7. remoteDescription 4.2.7. remoteDescription
The remoteDescription method returns a copy of the current remote The remoteDescription method returns a copy of the current remote
configuration, i.e. what was most recently passed to configuration, i.e. what was most recently passed to
setRemoteDescription, plus any remote candidates that have been setRemoteDescription, plus any remote candidates that have been
supplied via processIceMessage. supplied via processIceMessage.
TODO: Do we need to expose accessors for both the current and
proposed remote description?
A null object will be returned if the remote description has not yet A null object will be returned if the remote description has not yet
been established. been established, or if the PeerConnection has been closed.
5.2.8. updateIce 4.2.8. updateIce
The updateIce method allows the configuration of the ICE Agent to be The updateIce method allows the configuration of the ICE Agent to be
changed during the session, primarily for changing which types of changed during the session, primarily for changing which types of
local candidates are provided to the application and used for local candidates are provided to the application and used for
connectivity checks. A callee may initially configure the ICE Agent connectivity checks. A callee may initially configure the ICE Agent
to use only relay candidates, to avoid leaking location information, to use only relay candidates, to avoid leaking location information,
but update this configuration to use all candidates once the call is but update this configuration to use all candidates once the call is
accepted. accepted.
Regardless of the configuration, the gathering process collects all Regardless of the configuration, the gathering process collects all
available candidates, but excluded candidates will not be surfaced in available candidates, but excluded candidates will not be surfaced in
onicecallback or used for connectivity checks. onicecandidate callback or used for connectivity checks.
This call may result in a change to the state of the ICE Agent, and 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 may result in a change to media state if it results in connectivity
being established. being established.
5.2.9. addIceCandidate 4.2.9. addIceCandidate
The addIceCandidate method provides a remote candidate to the ICE The addIceCandidate method provides a remote candidate to the ICE
Agent, which will be added to the remote description. Connectivity Agent, which, if parsed successfully, will be added to the remote
checks will be sent to the new candidate. description according to the rules defined for Trickle ICE.
Connectivity checks will be sent to the new candidate.
This call will result in a change to the state of the ICE Agent, and 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 may result in a change to media state if it results in connectivity
being established. being established.
6. Configurable SDP Parameters 5. SDP Interaction Procedures
Note: This section is still very early and is likely to This section describes the specific procedures to be followed when
significantly change as we get a better understanding of the a) the creating and parsing SDP objects. [Work In Progress]
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 5.1. Constructing an Offer
may want to control, in either local or remote descriptions, using
this API.
o remove or reorder codecs (m=) 5.2. Generating an Answer
o change codec attributes (a=fmtp; ptime) 5.3. Parsing an Offer
o enable/disable BUNDLE (a=group) 5.4. Parsing an Answer
o enable/disable RTCP mux (a=rtcp-mux) 5.5. Applying a Local Description
o change send resolution or framerate (TBD) 5.6. Applying a Remote Description
o change desired recv resolution or framerate (TBD) 6. Configurable SDP Parameters
o change total bandwidth (b=) Note: This section is still very early and is likely to
significantly change as we get a better understanding of a) the use
cases for this b) the implications at the protocol level c) feedback
from implementors on what they can do.
o remove desired AVPF mechanisms (a=rtcp-fb) The following elements of the SDP media description MUST NOT be
changed between the createOffer and the setLocalDescription, since
they reflect transport attributes that are solely under browser
control, and the browser MUST NOT honor an attempt to change them:
o remove RTP header extensions (a=rtphdr-ext) o The number, type and port number of m-lines.
o The generated ICE credentials (a=ice-ufrag and a=ice-pwd).
o The set of ICE candidates and their parameters (a=candidate).
o add/change SSRC grouping (e.g. FID, RTX, etc) (a=ssrc-group) The following modifications, if done by the browser to a description
between createOffer/createAnswer and the setLocalDescription, MUST be
honored by the browser:
o add SSRC attributes (a=ssrc) o Remove or reorder codecs (m=)
The following parameters may be controlled by constraints passed into
createOffer/createAnswer. As an open issue, these changes may also
be be performed by manipulating the SDP returned from createOffer/
createAnswer, as indicated above, as long as the capabilities of the
endpoint are not exceeded (e.g. asking for a resolution greater than
what the endpoint can encode):
o disable BUNDLE (a=group)
o disable RTCP mux (a=rtcp-mux)
o change send resolution or framerate
o change desired recv resolution or framerate
o change maximum total bandwidth (b=) [OPEN ISSUE: need to clarify
if this is CT or AS - see section 5.8 of RFC4566]
o remove desired AVPF mechanisms (a=rtcp-fb)
o remove RTP header extensions (a=extmap)
o change media send/recv state (a=sendonly/recvonly/inactive) o change media send/recv state (a=sendonly/recvonly/inactive)
For example, an application could implement call hold by adding an For example, an application could implement call hold by adding an
a=inactive attribute to its local description, and then applying and a=inactive attribute to its local description, and then applying and
signaling that description. signaling that description.
The application can also modify the SDP to reduce the capabilities in
the offer it sends to the far side in any way the application sees
fit, as long as it is a valid SDP offer and specifies a subset of
what the browser is expecting to do.
As always, the application is solely responsible for what it sends to
the other party, and all incoming SDP will be processed by the
browser to the extent of its capabilities. It is an error to assume
that all SDP is well-formed; however, one should be able to assume
that any implementation of this specification will be able to
process, as a remote offer or answer, unmodified SDP coming from any
other implementation of this specification.
7. Security Considerations 7. Security Considerations
TODO The intent of the WebRTC protocol suite is to provide an environment
that is securable by default: all media is encrypted, keys are
exchanged in a secure fashion, and the Javascript API includes
functions that can be used to verify the identity of communication
partners.
8. IANA Considerations 8. IANA Considerations
This document requires no actions from IANA. This document requires no actions from IANA.
9. Acknowledgements 9. Acknowledgements
Harald Alvestrand, Dan Burnett, Neil Stratford, Eric Rescorla, Anant Significant text incorporated in the draft as well and review was
Narayanan, and Adam Bergkvist all provided valuable feedback on this provided by Harald Alvestrand and Suhas Nandakumar. Dan Burnett,
proposal. Suhas Nandakumar provided text and input for SDP Neil Stratford, Eric Rescorla, Anant Narayanan, Andrew Hutton,
requirements. Matthew Kaufman provided the observation that keeping Richard Ejzak, 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 state out of the browser allows a call to continue even if the page
is reloaded. is reloaded.
10. References 10. References
10.1. Normative References 10.1. Normative References
[I-D.rescorla-mmusic-ice-trickle] [I-D.rescorla-mmusic-ice-trickle]
Rescorla, E., Uberti, J., and E. Ivov, "Trickle ICE: Rescorla, E., Uberti, J., and E. Ivov, "Trickle ICE:
Incremental Provisioning of Candidates for the Interactive Incremental Provisioning of Candidates for the Interactive
Connectivity Establishment (ICE) Protocol", Connectivity Establishment (ICE) Protocol",
draft-rescorla-mmusic-ice-trickle-00 (work in progress), draft-rescorla-mmusic-ice-trickle-00 (work in progress),
October 2012. October 2012.
[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate [RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
Requirement Levels", BCP 14, RFC 2119, March 1997. Requirement Levels", BCP 14, RFC 2119, March 1997.
[RFC3261] Rosenberg, J., Schulzrinne, H., Camarillo, G., Johnston,
A., Peterson, J., Sparks, R., Handley, M., and E.
Schooler, "SIP: Session Initiation Protocol", RFC 3261,
June 2002.
[RFC3264] Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model [RFC3264] Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model
with Session Description Protocol (SDP)", RFC 3264, with Session Description Protocol (SDP)", RFC 3264,
June 2002. June 2002.
[RFC3388] Camarillo, G., Eriksson, G., Holler, J., and H.
Schulzrinne, "Grouping of Media Lines in the Session
Description Protocol (SDP)", RFC 3388, December 2002.
[RFC3960] Camarillo, G. and H. Schulzrinne, "Early Media and Ringing
Tone Generation in the Session Initiation Protocol (SIP)",
RFC 3960, December 2004.
[RFC4566] Handley, M., Jacobson, V., and C. Perkins, "SDP: Session [RFC4566] Handley, M., Jacobson, V., and C. Perkins, "SDP: Session
Description Protocol", RFC 4566, July 2006. Description Protocol", RFC 4566, July 2006.
[RFC4585] Ott, J., Wenger, S., Sato, N., Burmeister, C., and J. Rey,
"Extended RTP Profile for Real-time Transport Control
Protocol (RTCP)-Based Feedback (RTP/AVPF)", RFC 4585,
July 2006.
[RFC5124] Ott, J. and E. Carrara, "Extended Secure RTP Profile for
Real-time Transport Control Protocol (RTCP)-Based Feedback
(RTP/SAVPF)", RFC 5124, February 2008.
[RFC5245] Rosenberg, J., "Interactive Connectivity Establishment
(ICE): A Protocol for Network Address Translator (NAT)
Traversal for Offer/Answer Protocols", RFC 5245,
April 2010.
[RFC5761] Perkins, C. and M. Westerlund, "Multiplexing RTP Data and
Control Packets on a Single Port", RFC 5761, April 2010.
[RFC5888] Camarillo, G. and H. Schulzrinne, "The Session Description
Protocol (SDP) Grouping Framework", RFC 5888, June 2010.
10.2. Informative References 10.2. Informative References
[I-D.alvestrand-mmusic-msid]
Alvestrand, H., "Cross Session Stream Identification in
the Session Description Protocol",
draft-alvestrand-mmusic-msid-01 (work in progress),
October 2012.
[I-D.holmberg-mmusic-sdp-bundle-negotiation]
Holmberg, C. and H. Alvestrand, "Multiplexing Negotiation
Using Session Description Protocol (SDP) Port Numbers",
draft-holmberg-mmusic-sdp-bundle-negotiation-00 (work in
progress), October 2011.
[I-D.ietf-rtcweb-rtp-usage] [I-D.ietf-rtcweb-rtp-usage]
Perkins, C., Westerlund, M., and J. Ott, "Web Real-Time Perkins, C., Westerlund, M., and J. Ott, "Web Real-Time
Communication (WebRTC): Media Transport and Use of RTP", Communication (WebRTC): Media Transport and Use of RTP",
draft-ietf-rtcweb-rtp-usage-04 (work in progress), draft-ietf-rtcweb-rtp-usage-04 (work in progress),
July 2012. July 2012.
[I-D.jennings-rtcweb-signaling] [I-D.jennings-rtcweb-signaling]
Jennings, C., Rosenberg, J., and R. Jesup, "RTCWeb Offer/ Jennings, C., Rosenberg, J., and R. Jesup, "RTCWeb Offer/
Answer Protocol (ROAP)", Answer Protocol (ROAP)",
draft-jennings-rtcweb-signaling-01 (work in progress), draft-jennings-rtcweb-signaling-01 (work in progress),
October 2011. October 2011.
[I-D.nandakumar-rtcweb-sdp] [I-D.nandakumar-rtcweb-sdp]
Nandakumar, S. and C. Jennings, "SDP for the WebRTC", Nandakumar, S. and C. Jennings, "SDP for the WebRTC",
draft-nandakumar-rtcweb-sdp-00 (work in progress), draft-nandakumar-rtcweb-sdp-00 (work in progress),
October 2012. October 2012.
[RFC3556] Casner, S., "Session Description Protocol (SDP) Bandwidth
Modifiers for RTP Control Protocol (RTCP) Bandwidth",
RFC 3556, July 2003.
[RFC4568] Andreasen, F., Baugher, M., and D. Wing, "Session [RFC4568] Andreasen, F., Baugher, M., and D. Wing, "Session
Description Protocol (SDP) Security Descriptions for Media Description Protocol (SDP) Security Descriptions for Media
Streams", RFC 4568, July 2006. Streams", RFC 4568, July 2006.
[RFC5245] Rosenberg, J., "Interactive Connectivity Establishment [RFC5506] Johansson, I. and M. Westerlund, "Support for Reduced-Size
(ICE): A Protocol for Network Address Translator (NAT) Real-Time Transport Control Protocol (RTCP): Opportunities
Traversal for Offer/Answer Protocols", RFC 5245, and Consequences", RFC 5506, April 2009.
April 2010.
[RFC5576] Lennox, J., Ott, J., and T. Schierl, "Source-Specific
Media Attributes in the Session Description Protocol
(SDP)", RFC 5576, June 2009.
[RFC5763] Fischl, J., Tschofenig, H., and E. Rescorla, "Framework
for Establishing a Secure Real-time Transport Protocol
(SRTP) Security Context Using Datagram Transport Layer
Security (DTLS)", RFC 5763, May 2010.
[W3C.WD-webrtc-20111027] [W3C.WD-webrtc-20111027]
Bergkvist, A., Burnett, D., Narayanan, A., and C. Bergkvist, A., Burnett, D., Narayanan, A., and C.
Jennings, "WebRTC 1.0: Real-time Communication Between Jennings, "WebRTC 1.0: Real-time Communication Between
Browsers", World Wide Web Consortium WD WD-webrtc- Browsers", World Wide Web Consortium WD WD-webrtc-
20111027, October 2011, 20111027, October 2011,
<http://www.w3.org/TR/2011/WD-webrtc-20111027>. <http://www.w3.org/TR/2011/WD-webrtc-20111027>.
Appendix A. JSEP Implementation Examples Appendix A. JSEP Implementation Examples
skipping to change at page 26, line 35 skipping to change at page 26, line 25
OffererJS->OffererUA: pc.setLocalDescription("offer", offer); OffererJS->OffererUA: pc.setLocalDescription("offer", offer);
OffererJS->AnswererJS: {"type":"OFFER", "sdp":offer } OffererJS->AnswererJS: {"type":"OFFER", "sdp":offer }
// OFFER arrives at Answerer // OFFER arrives at Answerer
AnswererJS->AnswererUA: pc = new PeerConnection(); AnswererJS->AnswererUA: pc = new PeerConnection();
AnswererJS->AnswererUA: pc.setRemoteDescription("offer", msg.sdp); AnswererJS->AnswererUA: pc.setRemoteDescription("offer", msg.sdp);
AnswererUA->AnswererJS: onaddstream(remoteStream); AnswererUA->AnswererJS: onaddstream(remoteStream);
AnswererUA->OffererUA: iceCallback(candidate); AnswererUA->OffererUA: iceCallback(candidate);
// Answerer accepts call // Answerer accepts call
AnswererJS->AnswererUA: peer.addStream(localStream, null); AnswererJS->AnswererUA: pc.addStream(localStream, null);
AnswererJS->AnswererUA: answer = peer.createAnswer(msg.sdp, null); AnswererJS->AnswererUA: answer = pc.createAnswer(msg.sdp, null);
AnswererJS->AnswererUA: peer.setLocalDescription("answer", answer); AnswererJS->AnswererUA: pc.setLocalDescription("answer", answer);
AnswererJS->OffererJS: {"type":"ANSWER","sdp":answer } AnswererJS->OffererJS: {"type":"ANSWER","sdp":answer }
// ANSWER arrives at Offerer // ANSWER arrives at Offerer
OffererJS->OffererUA: peer.setRemoteDescription("answer", answer); OffererJS->OffererUA: pc.setRemoteDescription("answer", answer);
OffererUA->OffererJS: onaddstream(remoteStream); OffererUA->OffererJS: onaddstream(remoteStream);
// ICE Completes (at Answerer) // ICE Completes (at Answerer)
AnswererUA->AnswererJS: onopen();
AnswererUA->OffererUA: Media AnswererUA->OffererUA: Media
// ICE Completes (at Offerer) // ICE Completes (at Offerer)
OffererUA->OffererJS: onopen();
OffererJS->AnswererJS: {"type":"OK" } OffererJS->AnswererJS: {"type":"OK" }
OffererUA->AnswererUA: Media OffererUA->AnswererUA: Media
A.1.2. Call using XMPP A.1.2. Call using XMPP
This example demonstrates an XMPP call, making use of trickle This example demonstrates an XMPP call, making use of trickle
candidates. candidates.
// Call is initiated toward Answerer // Call is initiated toward Answerer
OffererJS->OffererUA: pc = new PeerConnection(); OffererJS->OffererUA: pc = new PeerConnection();
skipping to change at page 27, line 41 skipping to change at page 27, line 36
AnswererJS->AnswererUA: pc.addIceCandidate(candidate); AnswererJS->AnswererUA: pc.addIceCandidate(candidate);
AnswererUA->AnswererJS: onicecandidate(cand) AnswererUA->AnswererJS: onicecandidate(cand)
AnswererJS: createTransportInfo(cand); AnswererJS: createTransportInfo(cand);
AnswererJS->OffererJS: <jingle action="transport-info"/> AnswererJS->OffererJS: <jingle action="transport-info"/>
// transport-infos arrive at Offerer // transport-infos arrive at Offerer
OffererJS->OffererUA: candidates = parseTransportInfo(xmpp); OffererJS->OffererUA: candidates = parseTransportInfo(xmpp);
OffererJS->OffererUA: pc.addIceCandidate(candidates); OffererJS->OffererUA: pc.addIceCandidate(candidates);
// Answerer accepts call // Answerer accepts call
AnswererJS->AnswererUA: peer.addStream(localStream, null); AnswererJS->AnswererUA: pc.addStream(localStream, null);
AnswererJS->AnswererUA: answer = peer.createAnswer(offer, null); AnswererJS->AnswererUA: answer = pc.createAnswer(offer, null);
AnswererJS: xmpp = createSessionAccept(answer); AnswererJS: xmpp = createSessionAccept(answer);
AnswererJS->AnswererUA: pc.setLocalDescription("answer", answer); AnswererJS->AnswererUA: pc.setLocalDescription("answer", answer);
AnswererJS->OffererJS: <jingle action="session-accept"/> AnswererJS->OffererJS: <jingle action="session-accept"/>
// session-accept arrives at Offerer // session-accept arrives at Offerer
OffererJS: answer = parseSessionAccept(xmpp); OffererJS: answer = parseSessionAccept(xmpp);
OffererJS->OffererUA: peer.setRemoteDescription("answer", answer); OffererJS->OffererUA: pc.setRemoteDescription("answer", answer);
OffererUA->OffererJS: onaddstream(remoteStream); OffererUA->OffererJS: onaddstream(remoteStream);
// ICE Completes (at Answerer) // ICE Completes (at Answerer)
AnswererUA->AnswererJS: onopen();
AnswererUA->OffererUA: Media AnswererUA->OffererUA: Media
// ICE Completes (at Offerer) // ICE Completes (at Offerer)
OffererUA->OffererJS: onopen();
OffererUA->AnswererUA: Media OffererUA->AnswererUA: Media
A.1.3. Adding video to a call, using XMPP A.1.3. Adding video to a call, using XMPP
This example demonstrates an XMPP call, where the XMPP content-add This example demonstrates an XMPP call, where the XMPP content-add
mechanism is used to add video media to an existing session. For mechanism is used to add video media to an existing session. For
simplicity, candidate exchange is not shown. simplicity, candidate exchange is not shown.
Note that the offerer for the change to the session may be different Note that the offerer for the change to the session may be different
than the original call offerer. than the original call offerer.
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OffererJS->AnswererJS: SIP INVITE w/ SDP OffererJS->AnswererJS: SIP INVITE w/ SDP
// INVITE arrives at Answerer // INVITE arrives at Answerer
AnswererJS->AnswererUA: pc = new PeerConnection(); AnswererJS->AnswererUA: pc = new PeerConnection();
AnswererJS: offer = parseInvite(sip); AnswererJS: offer = parseInvite(sip);
AnswererJS->AnswererUA: pc.setRemoteDescription("offer", offer); AnswererJS->AnswererUA: pc.setRemoteDescription("offer", offer);
AnswererUA->AnswererJS: onaddstream(remoteStream); AnswererUA->AnswererJS: onaddstream(remoteStream);
AnswererUA->OffererUA: onicecandidate(candidate); AnswererUA->OffererUA: onicecandidate(candidate);
// Answerer accepts call // Answerer accepts call
AnswererJS->AnswererUA: peer.addStream(localStream, null); AnswererJS->AnswererUA: pc.addStream(localStream, null);
AnswererJS->AnswererUA: answer = peer.createAnswer(offer, null); AnswererJS->AnswererUA: answer = pc.createAnswer(offer, null);
AnswererJS: sip = createResponse(200, answer); AnswererJS: sip = createResponse(200, answer);
AnswererJS->AnswererUA: peer.setLocalDescription("answer", answer); AnswererJS->AnswererUA: pc.setLocalDescription("answer", answer);
AnswererJS->OffererJS: 200 OK w/ SDP AnswererJS->OffererJS: 200 OK w/ SDP
// 200 OK arrives at Offerer // 200 OK arrives at Offerer
OffererJS: answer = parseResponse(sip); OffererJS: answer = parseResponse(sip);
OffererJS->OffererUA: peer.setRemoteDescription("answer", answer); OffererJS->OffererUA: pc.setRemoteDescription("answer", answer);
OffererUA->OffererJS: onaddstream(remoteStream); OffererUA->OffererJS: onaddstream(remoteStream);
OffererJS->AnswererJS: ACK OffererJS->AnswererJS: ACK
// ICE Completes (at Answerer) // ICE Completes (at Answerer)
AnswererUA->AnswererJS: onopen();
AnswererUA->OffererUA: Media AnswererUA->OffererUA: Media
// ICE Completes (at Offerer) // ICE Completes (at Offerer)
OffererUA->OffererJS: onopen();
OffererUA->AnswererUA: Media OffererUA->AnswererUA: Media
A.1.6. Handling early media (e.g. 1-800-GO FEDEX), using SIP A.1.6. Handling early media (e.g. 1-800-GO FEDEX), using SIP
This example demonstrates how early media could be handled; for This example demonstrates how early media could be handled; for
simplicity, only the offerer side of the call is shown. simplicity, only the offerer side of the call is shown.
// Call is initiated toward Answerer // Call is initiated toward Answerer
OffererJS->OffererUA: pc = new PeerConnection(); OffererJS->OffererUA: pc = new PeerConnection();
OffererJS->OffererUA: pc.addStream(localStream, null); OffererJS->OffererUA: pc.addStream(localStream, null);
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OffererJS->OffererUA: pc.setLocalDescription("offer", offer); OffererJS->OffererUA: pc.setLocalDescription("offer", offer);
OffererJS: sip = createInvite(offer); OffererJS: sip = createInvite(offer);
OffererJS->AnswererJS: SIP INVITE w/ SDP OffererJS->AnswererJS: SIP INVITE w/ SDP
// 180 Ringing is received by offerer, w/ SDP // 180 Ringing is received by offerer, w/ SDP
OffererJS: answer = parseResponse(sip); OffererJS: answer = parseResponse(sip);
OffererJS->OffererUA: pc.setRemoteDescription("pranswer", answer); OffererJS->OffererUA: pc.setRemoteDescription("pranswer", answer);
OffererUA->OffererJS: onaddstream(remoteStream); OffererUA->OffererJS: onaddstream(remoteStream);
// ICE Completes (at Offerer) // ICE Completes (at Offerer)
OffererUA->OffererJS: onopen();
OffererUA->AnswererUA: Media OffererUA->AnswererUA: Media
// 200 OK arrives at Offerer // 200 OK arrives at Offerer
OffererJS: answer = parseResponse(sip); OffererJS: answer = parseResponse(sip);
OffererJS->OffererUA: pc.setRemoteDescription("answer", answer); OffererJS->OffererUA: pc.setRemoteDescription("answer", answer);
OffererJS->AnswererJS: ACK OffererJS->AnswererJS: ACK
Appendix B. Change log Appendix B. Change log
Changes in draft -02: Changes in draft-03:
o Converted from nroff o Added text describing relationship to W3C specification
Changes in draft -02:
o Converted from nroff
o Removed comparisons to old approaches abandoned by the working o Removed comparisons to old approaches abandoned by the working
group group
o Removed stuff that has moved to W3C specificaiton o Removed stuff that has moved to W3C specificaiton
o Align SDP handling with W3C draft o Align SDP handling with W3C draft
o Clarified section on forking. o Clarified section on forking.
Changes in draft -01: Changes in draft -01:
o Added diagrams for architecture and state machine. o Added diagrams for architecture and state machine.
o Added sections on forking and rehydration. o Added sections on forking and rehydration.
o Clarified meaning of "pranswer" and "answer". o Clarified meaning of "pranswer" and "answer".
o Reworked how ICE restarts and media directions are controlled. o Reworked how ICE restarts and media directions are controlled.
o Added list of parameters that can be changed in a description. o Added list of parameters that can be changed in a description.
o Updated suggested API and examples to match latest thinking. o Updated suggested API and examples to match latest thinking.
o Suggested API and examples have been moved to an appendix. o Suggested API and examples have been moved to an appendix.
Changes in draft -00: Changes in draft -00:
o Migrated from draft-uberti-rtcweb-jsep-02. o Migrated from draft-uberti-rtcweb-jsep-02.
Authors' Addresses Authors' Addresses
Justin Uberti Justin Uberti
Google Google
 End of changes. 153 change blocks. 
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