Network Working Group J. Uberti
Internet-Draft Google
Intended status: Standards Track C. Jennings
Expires: March 22, 2014 Cisco
September 18, 2013
Javascript Session Establishment Protocol
draft-ietf-rtcweb-jsep-04
Abstract
This document describes the mechanisms for allowing a Javascript
application to control the signaling plane of a multimedia session
via the interface specified in the W3C RTCPeerConnection API, and
discusses how this relates to existing signaling protocols.
Status of This Memo
This Internet-Draft is submitted in full conformance with the
provisions of BCP 78 and BCP 79.
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This Internet-Draft will expire on March 22, 2014.
Copyright Notice
Copyright (c) 2013 IETF Trust and the persons identified as the
document authors. All rights reserved.
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Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 3
1.1. General Design of JSEP . . . . . . . . . . . . . . . . . 3
1.2. Other Approaches Considered . . . . . . . . . . . . . . . 5
2. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 6
3. Semantics and Syntax . . . . . . . . . . . . . . . . . . . . 6
3.1. Signaling Model . . . . . . . . . . . . . . . . . . . . . 6
3.2. Session Descriptions and State Machine . . . . . . . . . 7
3.3. Session Description Format . . . . . . . . . . . . . . . 9
3.4. ICE . . . . . . . . . . . . . . . . . . . . . . . . . . . 10
3.4.1. ICE Candidate Trickling . . . . . . . . . . . . . . . 10
3.4.1.1. ICE Candidate Format . . . . . . . . . . . . . . 10
3.5. Interactions With Forking . . . . . . . . . . . . . . . . 11
3.5.1. Sequential Forking . . . . . . . . . . . . . . . . . 11
3.5.2. Parallel Forking . . . . . . . . . . . . . . . . . . 12
3.6. Session Rehydration . . . . . . . . . . . . . . . . . . . 13
4. Interface . . . . . . . . . . . . . . . . . . . . . . . . . . 13
4.1. Methods . . . . . . . . . . . . . . . . . . . . . . . . . 14
4.1.1. createOffer . . . . . . . . . . . . . . . . . . . . . 14
4.1.2. createAnswer . . . . . . . . . . . . . . . . . . . . 15
4.1.3. SessionDescriptionType . . . . . . . . . . . . . . . 15
4.1.3.1. Use of Provisional Answers . . . . . . . . . . . 16
4.1.3.2. Rollback . . . . . . . . . . . . . . . . . . . . 17
4.1.4. setLocalDescription . . . . . . . . . . . . . . . . . 17
4.1.5. setRemoteDescription . . . . . . . . . . . . . . . . 18
4.1.6. localDescription . . . . . . . . . . . . . . . . . . 18
4.1.7. remoteDescription . . . . . . . . . . . . . . . . . . 18
4.1.8. updateIce . . . . . . . . . . . . . . . . . . . . . . 19
4.1.9. addIceCandidate . . . . . . . . . . . . . . . . . . . 19
5. SDP Interaction Procedures . . . . . . . . . . . . . . . . . 19
5.1. SDP Requirements Overview . . . . . . . . . . . . . . . . 19
5.2. Constructing an Offer . . . . . . . . . . . . . . . . . . 21
5.2.1. Initial Offers . . . . . . . . . . . . . . . . . . . 21
5.2.2. Subsequent Offers . . . . . . . . . . . . . . . . . . 25
5.2.3. Constraints Handling . . . . . . . . . . . . . . . . 26
5.2.3.1. OfferToReceiveAudio . . . . . . . . . . . . . . . 26
5.2.3.2. OfferToReceiveVideo . . . . . . . . . . . . . . . 27
5.2.3.3. VoiceActivityDetection . . . . . . . . . . . . . 27
5.2.3.4. IceRestart . . . . . . . . . . . . . . . . . . . 27
5.3. Generating an Answer . . . . . . . . . . . . . . . . . . 27
5.3.1. Initial Answers . . . . . . . . . . . . . . . . . . . 27
5.3.2. Subsequent Answers . . . . . . . . . . . . . . . . . 31
5.3.3. Constraints Handling . . . . . . . . . . . . . . . . 31
5.4. Parsing an Offer . . . . . . . . . . . . . . . . . . . . 31
5.5. Parsing an Answer . . . . . . . . . . . . . . . . . . . . 31
5.6. Applying a Local Description . . . . . . . . . . . . . . 31
5.7. Applying a Remote Description . . . . . . . . . . . . . . 31
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6. Configurable SDP Parameters . . . . . . . . . . . . . . . . . 31
7. Security Considerations . . . . . . . . . . . . . . . . . . . 33
8. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 33
9. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 33
10. References . . . . . . . . . . . . . . . . . . . . . . . . . 33
10.1. Normative References . . . . . . . . . . . . . . . . . . 33
10.2. Informative References . . . . . . . . . . . . . . . . . 35
Appendix A. JSEP Implementation Examples . . . . . . . . . . . . 36
A.1. Example API Flows . . . . . . . . . . . . . . . . . . . . 36
A.1.1. Call using ROAP . . . . . . . . . . . . . . . . . . . 36
A.1.2. Call using XMPP . . . . . . . . . . . . . . . . . . . 37
A.1.3. Adding video to a call, using XMPP . . . . . . . . . 38
A.1.4. Simultaneous add of video streams, using XMPP . . . . 39
A.1.5. Call using SIP . . . . . . . . . . . . . . . . . . . 40
A.1.6. Handling early media (e.g. 1-800-GO FEDEX), using SIP 40
A.2. Example Session Descriptions . . . . . . . . . . . . . . 41
A.2.1. createOffer . . . . . . . . . . . . . . . . . . . . . 41
A.2.2. createAnswer . . . . . . . . . . . . . . . . . . . . 43
Appendix B. Change log . . . . . . . . . . . . . . . . . . . . . 44
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 45
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
control the media plane, but to leave the signaling plane up to the
application as much as possible. The rationale is that different
applications may prefer to use different protocols, such as the
existing SIP or Jingle call signaling protocols, or something custom
to the particular application, perhaps for a novel use case. In this
approach, the key information that needs to be exchanged is the
multimedia session description, which specifies the necessary
transport and media configuration information necessary to establish
the media plane.
The browser environment also has its own challenges that pose
problems for an embedded signaling state machine. One of these is
that the user may reload the web page at any time. If the browser is
fully in charge of the signaling state, this will result in the loss
of the call when this state is wiped by the reload. However, if the
state can be stored at the server, and pushed back down to the new
page, the call can be resumed with minimal interruption.
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With these considerations in mind, this document describes the
Javascript Session Establishment Protocol (JSEP) that allows for full
control of the signaling state machine from Javascript. This
mechanism effectively removes the browser almost completely from the
core signaling flow; the only interface needed is a way for the
application to pass in the local and remote session descriptions
negotiated by whatever signaling mechanism is used, and a way to
interact with the ICE state machine.
In this document, the use of JSEP is described as if it always occurs
between two browsers. Note though in many cases it will actually be
between a browser and some kind of server, such as a gateway or MCU.
This distinction is invisible to the browser; it just follows the
instructions it is given via the API.
JSEP's handling of session descriptions is simple and
straightforward. Whenever an offer/answer exchange is needed, the
initiating side creates an offer by calling a createOffer() API. The
application optionally modifies that offer, and then uses it to set
up its local config via the setLocalDescription() API. The offer is
then sent off to the remote side over its preferred signaling
mechanism (e.g., WebSockets); upon receipt of that offer, the remote
party installs it using the setRemoteDescription() API.
When the call is accepted, the callee uses the createAnswer() API to
generate an appropriate answer, applies it using
setLocalDescription(), and sends the answer back to the initiator
over the signaling channel. When the offerer gets that answer, it
installs it using setRemoteDescription(), and initial setup is
complete. This process can be repeated for additional offer/answer
exchanges.
Regarding ICE [RFC5245], JSEP decouples the ICE state machine from
the overall signaling state machine, as the ICE state machine must
remain in the browser, because only the browser has the necessary
knowledge of candidates and other transport info. Performing this
separation also provides additional flexibility; in protocols that
decouple session descriptions from transport, such as Jingle, the
transport information can be sent separately; in protocols that
don't, such as SIP, the information can be used in the aggregated
form. Sending transport information separately can allow for faster
ICE and DTLS startup, since the necessary roundtrips can occur while
waiting for the remote side to accept the session.
Through its abstraction of signaling, the JSEP approach does require
the application to be aware of the signaling process. While the
application does not need to understand the contents of session
descriptions to set up a call, the application must call the right
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APIs at the right times, convert the session descriptions and ICE
information into the defined messages of its chosen signaling
protocol, and perform the reverse conversion on the messages it
receives from the other side.
One way to mitigate this is to provide a Javascript library that
hides this complexity from the developer; said library would
implement a given signaling protocol along with its state machine and
serialization code, presenting a higher level call-oriented interface
to the application developer. For example, this library could easily
adapt the JSEP API into the API that was proposed for the ROAP
signaling protocol [I-D.jennings-rtcweb-signaling], which would
perform a ROAP call setup under the covers, interacting with the
application only when it needs a signaling message to be sent. In
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.
1.2. Other Approaches Considered
One approach that was considered instead of JSEP was to include a
lightweight signaling protocol. Instead of providing session
descriptions to the API, the API would produce and consume messages
from this protocol. While providing a more high-level API, this put
more control of signaling within the browser, forcing the browser to
have to understand and handle concepts like signaling glare. In
addition, it prevented the application from driving the state machine
to a desired state, as is needed in the page reload case.
A second approach that was considered but not chosen was to decouple
the management of the media control objects from session
descriptions, instead offering APIs that would control each component
directly. This was rejected based on a feeling that requiring
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
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approach would instead expose a getCapabilities API which would
provide the application with the information it needed in order to
generate its own session descriptions. This increases the amount of
work that the application needs to do; it needs to know how to
generate session descriptions from capabilities, and especially how
to generate the correct answer from an arbitrary offer and the
supported capabilities. While this could certainly be addressed by
using a library like the one mentioned above, it basically forces the
use of said library even for a simple example. Providing createOffer
/createAnswer avoids this problem, but still allows applications to
generate their own offers/answers (to a large extent) if they choose,
using the description generated by createOffer as an indication of
the browser's capabilities.
2. Terminology
The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
"SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
document are to be interpreted as described in [RFC2119].
3. Semantics and Syntax
3.1. Signaling Model
JSEP does not specify a particular signaling model or state machine,
other than the generic need to exchange SDP media descriptions in the
fashion described by [RFC3264] (offer/answer) in order for both sides
of the session to know how to conduct the session. JSEP provides
mechanisms to create offers and answers, as well as to apply them to
a session. However, the browser is totally decoupled from the actual
mechanism by which these offers and answers are communicated to the
remote side, including addressing, retransmission, forking, and glare
handling. These issues are left entirely up to the application; the
application has complete control over which offers and answers get
handed to the browser, and when.
+-----------+ +-----------+
| Web App |<--- App-Specific Signaling -->| Web App |
+-----------+ +-----------+
^ ^
| SDP | SDP
V V
+-----------+ +-----------+
| Browser |<----------- Media ------------>| Browser |
+-----------+ +-----------+
Figure 1: JSEP Signaling Model
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3.2. Session Descriptions and State Machine
In order to establish the media plane, the user agent needs specific
parameters to indicate what to transmit to the remote side, as well
as how to handle the media that is received. These parameters are
determined by the exchange of session descriptions in offers and
answers, and there are certain details to this process that must be
handled in the JSEP APIs.
Whether a session description applies to the local side or the remote
side affects the meaning of that description. For example, the list
of codecs sent to a remote party indicates what the local side is
willing to receive, which, when intersected with the set of codecs
the remote side supports, specifies what the remote side should send.
However, not all parameters follow this rule; for example, the SRTP
parameters [RFC4568] sent to a remote party indicate what the local
side will use to encrypt, and thereby what the remote party should
expect to receive; the remote party will have to accept these
parameters, with no option to choose a different value.
In addition, various RFCs put different conditions on the format of
offers versus answers. For example, a offer may propose multiple
SRTP configurations, but an answer may only contain a single SRTP
configuration.
Lastly, while the exact media parameters are only known only after a
offer and an answer have been exchanged, it is possible for the
offerer to receive media after they have sent an offer and before
they have received an answer. To properly process incoming media in
this case, the offerer's media handler must be aware of the details
of the offer before the answer arrives.
Therefore, in order to handle session descriptions properly, the user
agent needs:
1. To know if a session description pertains to the local or remote
side.
2. To know if a session description is an offer or an answer.
3. To allow the offer to be specified independently of the answer.
JSEP addresses this by adding both a setLocalDescription and a
setRemoteDescription method and having session description objects
contain a type field indicating the type of session description being
supplied. This satisfies the requirements listed above for both the
offerer, who first calls setLocalDescription(sdp [offer]) and then
later setRemoteDescription(sdp [answer]), as well as for the
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answerer, who first calls setRemoteDescription(sdp [offer]) and then
later setLocalDescription(sdp [answer]).
JSEP also allows for an answer to be treated as provisional by the
application. Provisional answers provide a way for an answerer to
communicate initial session parameters back to the offerer, in order
to allow the session to begin, while allowing a final answer to be
specified later. This concept of a final answer is important to the
offer/answer model; when such an answer is received, any extra
resources allocated by the caller can be released, now that the exact
session configuration is known. These "resources" can include things
like extra ICE components, TURN candidates, or video decoders.
Provisional answers, on the other hand, do no such deallocation
results; as a result, multiple dissimilar provisional answers can be
received and applied during call setup.
In [RFC3264], the constraint at the signaling level is that only one
offer can be outstanding for a given session, but from the media
stack level, a new offer can be generated at any point. For example,
when using SIP for signaling, if one offer is sent, then cancelled
using a SIP CANCEL, another offer can be generated even though no
answer was received for the first offer. To support this, the JSEP
media layer can provide an offer whenever the Javascript application
needs one for the signaling. The answerer can send back zero or more
provisional answers, and finally end the offer-answer exchange by
sending a final answer. The state machine for this is as follows:
setRemote(OFFER) setLocal(PRANSWER)
/-----\ /-----\
| | | |
v | v |
+---------------+ | +---------------+ |
| |----/ | |----/
| | setLocal(PRANSWER) | |
| Remote-Offer |------------------- >| Local-Pranswer|
| | | |
| | | |
+---------------+ +---------------+
^ | |
| | setLocal(ANSWER) |
setRemote(OFFER) | |
| V setLocal(ANSWER) |
+---------------+ |
| | |
| | |
| Stable |<---------------------------+
| | |
| | |
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+---------------+ setRemote(ANSWER) |
^ | |
| | setLocal(OFFER) |
setRemote(ANSWER) | |
| V |
+---------------+ +---------------+
| | | |
| | setRemote(PRANSWER) | |
| Local-Offer |------------------- >|Remote-Pranswer|
| | | |
| |----\ | |----\
+---------------+ | +---------------+ |
^ | ^ |
| | | |
\-----/ \-----/
setLocal(OFFER) setRemote(PRANSWER)
Figure 2: JSEP State Machine
Aside from these state transitions, there is no other difference
between the handling of provisional ("pranswer") and final ("answer")
answers.
3.3. Session Description Format
In the WebRTC specification, session descriptions are formatted as
SDP messages. While this format is not optimal for manipulation from
Javascript, it is widely accepted, and frequently updated with new
features. Any alternate encoding of session descriptions would have
to keep pace with the changes to SDP, at least until the time that
this new encoding eclipsed SDP in popularity. As a result, JSEP
currently uses SDP as the internal representation for its session
descriptions.
However, to simplify Javascript processing, and provide for future
flexibility, the SDP syntax is encapsulated within a
SessionDescription object, which can be constructed from SDP, and be
serialized out to SDP. If future specifications agree on a JSON
format for session descriptions, we could easily enable this object
to generate and consume that JSON.
Other methods may be added to SessionDescription in the future to
simplify handling of SessionDescriptions from Javascript. In the
meantime, Javascript libraries can be used to perform these
manipulations.
Note that most applications should be able to treat the
SessionDescriptions produced and consumed by these various API calls
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as opaque blobs; that is, the application will not need to read or
change them. The W3C API will provide appropriate APIs to allow the
application to control various session parameters, which will provide
the necessary information to the browser about what sort of
SessionDescription to produce.
3.4. ICE
When a new ICE candidate is available, the ICE Agent will notify the
application via a callback; these candidates will automatically be
added to the local session description. When all candidates have
been gathered, the callback will also be invoked to signal that the
gathering process is complete.
3.4.1. ICE Candidate Trickling
Candidate trickling is a technique through which a caller may
incrementally provide candidates to the callee after the initial
offer has been dispatched; the semantics of "Trickle ICE" are defined
in [I-D.ivov-mmusic-trickle-ice]. This process allows the callee to
begin acting upon the call and setting up the ICE (and perhaps DTLS)
connections immediately, without having to wait for the caller to
gather all possible candidates. This results in faster call startup
in cases where gathering is not performed prior to initiating the
call.
JSEP supports optional candidate trickling by providing APIs that
provide control and feedback on the ICE candidate gathering process.
Applications that support candidate trickling can send the initial
offer immediately and send individual candidates when they get the
notified of a new candidate; applications that do not support this
feature can simply wait for the indication that gathering is
complete, and then create and send their offer, with all the
candidates, at this time.
Upon receipt of trickled candidates, the receiving application will
supply them to its ICE Agent. This triggers the ICE Agent to start
using the new remote candidates for connectivity checks.
3.4.1.1. ICE Candidate Format
As with session descriptions, the syntax of the IceCandidate object
provides some abstraction, but can be easily converted to and from
the SDP candidate lines.
The candidate lines are the only SDP information that is contained
within IceCandidate, as they represent the only information needed
that is not present in the initial offer (i.e. for trickle
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candidates). This information is carried with the same syntax as the
"candidate-attribute" field defined for ICE. For example:
candidate:1 1 UDP 1694498815 192.0.2.33 10000 typ host
The IceCandidate object also contains fields to indicate which m=
line it should be associated with. The m line can be identified in
one of two ways; either by a m-line index, or a MID. The m-line
index is a zero-based index, referring to the Nth m-line in the SDP.
The MID uses the "media stream identification", as defined in
[RFC5888] , to identify the m-line. WebRTC implementations creating
an ICE Candidate object MUST populate both of these fields.
Implementations receiving an ICE Candidate object SHOULD use the MID
if they implement that functionality, or the m-line index, if not.
3.5. Interactions With Forking
Some call signaling systems allow various types of forking where an
SDP Offer may be provided to more than one device. For example, SIP
[RFC3261] defines both a "Parallel Search" and "Sequential Search".
Although these are primarily signaling level issues that are outside
the scope of JSEP, they do have some impact on the configuration of
the media plane which is relevant. When forking happens at the
signaling layer, the Javascript application responsible for the
signaling needs to make the decisions about what media should be sent
or received at any point of time, as well as which remote endpoint it
should communicate with; JSEP is used to make sure the media engine
can make the RTP and media perform as required by the application.
The basic operations that the applications can have the media engine
do are:
Start exchanging media to a given remote peer, but keep all the
resources reserved in the offer.
Start exchanging media with a given remote peer, and free any
resources in the offer that are not being used.
3.5.1. Sequential Forking
Sequential forking involves a call being dispatched to multiple
remote callees, where each callee can accept the call, but only one
active session ever exists at a time; no mixing of received media is
performed.
JSEP handles sequential forking well, allowing the application to
easily control the policy for selecting the desired remote endpoint.
When an answer arrives from one of the callees, the application can
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choose to apply it either as a provisional answer, leaving open the
possibility of using a different answer in the future, or apply it as
a final answer, ending the setup flow.
In a "first-one-wins" situation, the first answer will be applied as
a final answer, and the application will reject any subsequent
answers. In SIP parlance, this would be ACK + BYE.
In a "last-one-wins" situation, all answers would be applied as
provisional answers, and any previous call leg will be terminated.
At some point, the application will end the setup process, perhaps
with a timer; at this point, the application could reapply the
existing remote description as a final answer.
3.5.2. Parallel Forking
Parallel forking involves a call being dispatched to multiple remote
callees, where each callee can accept the call, and multiple
simultaneous active signaling sessions can be established as a
result. If multiple callees send media at the same time, the
possibilities for handling this are described in Section 3.1 of
[RFC3960]. Most SIP devices today only support exchanging media with
a single device at a time, and do not try to mix multiple early media
audio sources, as that could result in a confusing situation. For
example, consider having a European ringback tone mixed together with
the North American ringback tone - the resulting sound would not be
like either tone, and would confuse the user. If the signaling
application wishes to only exchange media with one of the remote
endpoints at a time, then from a media engine point of view, this is
exactly like the sequential forking case.
In the parallel forking case where the Javascript application wishes
to simultaneously exchange media with multiple peers, the flow is
slightly more complex, but the Javascript application can follow the
strategy that [RFC3960] describes using UPDATE. (It is worth noting
that use cases where this is the desired behavior are very unusual.)
The UPDATE approach allows the signaling to set up a separate media
flow for each peer that it wishes to exchange media with. In JSEP,
this offer used in the UPDATE would be formed by simply creating a
new PeerConnection and making sure that the same local media streams
have been added into this new PeerConnection. Then the new
PeerConnection object would produce a SDP offer that could be used by
the signaling to perform the UPDATE strategy discussed in [RFC3960].
As a result of sharing the media streams, the application will end up
with N parallel PeerConnection sessions, each with a local and remote
description and their own local and remote addresses. The media flow
from these sessions can be managed by specifying SDP direction
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attributes in the descriptions, or the application can choose to play
out the media from all sessions mixed together. Of course, if the
application wants to only keep a single session, it can simply
terminate the sessions that it no longer needs.
3.6. Session Rehydration
In the event that the local application state is reinitialized,
either due to a user reload of the page, or a decision within the
application to reload itself (perhaps to update to a new version), it
is possible to keep an existing session alive, via a process called
"rehydration". 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
outside of the page, perhaps on the application server, or in browser
local storage. The page is then reloaded, the saved signaling state
is retrieved, and a new PeerConnection object is created for the
session. The previously obtained MediaStreams are re-acquired, and
are given the same IDs as the original session; this ensures the IDs
in use by the remote side continue to work. Next, a new offer is
generated by the new PeerConnection; this offer will have new ICE and
possibly new DTLS-SRTP certificate fingerprints (since the old ICE
and SRTP state has been lost). Finally, this offer is used to re-
initiate the session with the existing remote endpoint, who simply
sees the new offer as 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
the actual internal PeerConnection object in the browser was kept
alive for some time after the web page was killed and provided some
way for a new page to acquire the old PeerConnection object.]
4. Interface
This section details the basic operations that must be present to
implement JSEP functionality. The actual API exposed in the W3C API
may have somewhat different syntax, but should map easily to these
concepts.
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4.1. Methods
4.1.1. createOffer
The createOffer method generates a blob of SDP that contains a
[RFC3264] offer with the supported configurations for the session,
including descriptions of the local MediaStreams attached to this
PeerConnection, the codec/RTP/RTCP options supported by this
implementation, and any candidates that have been gathered by the ICE
Agent. A constraints parameters may be supplied to provide
additional control over the generated offer. This constraints
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
functionality for the session (certain parts that are supported but
not desired by default may be omitted); for each SDP line, the
generation of the SDP will follow the process defined for generating
an initial offer from the document that specifies the given SDP line.
The exact handling of initial offer generation is detailed in
Section 5.2.1. below.
In the event createOffer is called after the session is established,
createOffer will generate an offer to modify the current session
based on any changes that have been made to the session, e.g. adding
or removing MediaStreams, or requesting an ICE restart. For each
existing stream, the generation of each SDP line must follow the
process defined for generating an updated offer from the document
that specifies the given SDP line. For each new stream, the
generation of the SDP must follow the process of generating an
initial offer, as mentioned above. If no changes have been made, or
for SDP lines that are unaffected by the requested changes, the offer
will only contain the parameters negotiated by the last offer-answer
exchange. The exact handling of subsequent offer generation is
detailed in Section 5.2.2. below.
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Session descriptions generated by createOffer must be immediately
usable by setLocalDescription; if a system has limited resources
(e.g. a finite number of decoders), createOffer should return an
offer that reflects the current state of the system, so that
setLocalDescription will succeed when it attempts to acquire those
resources. Because this method may need to inspect the system state
to determine the currently available resources, it may be implemented
as an async operation.
Calling this method may do things such as generate new ICE
credentials, but does not result in candidate gathering, or cause
media to start or stop flowing.
4.1.2. createAnswer
The createAnswer method generates a blob of SDP that contains a
[RFC3264] SDP answer with the supported configuration for the session
that is compatible with the parameters supplied in the offer. Like
createOffer, the returned blob contains descriptions of the local
MediaStreams attached to this PeerConnection, the codec/RTP/RTCP
options negotiated for this session, and any candidates that have
been gathered by the ICE Agent. A constraints parameter may be
supplied to provide additional control over the generated answer.
As an answer, the generated SDP will contain a specific configuration
that specifies how the media plane should be established; for each
SDP line, the generation of the SDP must follow the process defined
for generating an answer from the document that specifies the given
SDP line. The exact handling of answer generation is detailed in
Section 5.3. below.
Session descriptions generated by createAnswer must be immediately
usable by setLocalDescription; like createOffer, the returned
description should reflect the current state of the system. Because
this method may need to inspect the system state to determine the
currently available resources, it may need to be implemented as an
async operation.
Calling this method may do things such as generate new ICE
credentials, but does not trigger candidate gathering or change media
state.
4.1.3. SessionDescriptionType
Session description objects (RTCSessionDescription) may be of type
"offer", "pranswer", and "answer". These types provide information
as to how the description parameter should be parsed, and how the
media state should be changed.
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"offer" indicates that a description should be parsed as an offer;
said description may include many possible media configurations. A
description used as an "offer" may be applied anytime the
PeerConnection is in a stable state, or as an update to a previously
supplied but unanswered "offer".
"pranswer" indicates that a description should be parsed as an
answer, but not a final answer, and so should not result in the
freeing of allocated resources. It may result in the start of media
transmission, if the answer does not specify an inactive media
direction. A description used as a "pranswer" may be applied as a
response to an "offer", or an update to a previously sent "answer".
"answer" indicates that a description should be parsed as an answer,
the offer-answer exchange should be considered complete, and any
resources (decoders, candidates) that are no longer needed can be
released. A description used as an "answer" may be applied as a
response to a "offer", or an update to a previously sent "pranswer".
The only difference between a provisional and final answer is that
the final answer results in the freeing of any unused resources that
were allocated as a result of the offer. As such, the application
can use some discretion on whether an answer should be applied as
provisional or final, and can change the type of the session
description as needed. For example, in a serial forking scenario, an
application may receive multiple "final" answers, one from each
remote endpoint. The application could choose to accept the initial
answers as provisional answers, and only apply an answer as final
when it receives one that meets its criteria (e.g. a live user
instead of voicemail).
4.1.3.1. Use of Provisional Answers
Most web applications will not need to create answers using the
"pranswer" type. The preferred handling for a web application would
be to create and send an "inactive" answer more or less immediately
after receiving the offer, instead of waiting for a human user to
physically answer the call. Later, when the human input is received,
the application can create a new "sendrecv" offer to update the
previous offer/answer pair and start the media flow. This approach
is preferred because it minimizes the amount of time that the offer-
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
physically 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.
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Consider a typical web application that will set up a data channel,
an audio channel, and a video channel. When an endpoint receives an
offer with these channels, it could send an answer accepting the data
channel for two-way data, and accepting the audio and video tracks as
inactive or receive-only. It could then ask the user to accept the
call, acquire the local media streams, and send a new offer to the
remote side moving the audio and video to be two-way media. By the
time the human has accepted the call and sent the new offer, it is
likely that the ICE and DTLS handshaking for all the channels will
already be set up.
4.1.3.2. 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.)
4.1.4. setLocalDescription
The setLocalDescription method instructs the PeerConnection to apply
the supplied SDP blob as its local configuration. The type field
indicates whether the blob should be processed as an offer,
provisional answer, or final answer; offers and answers are checked
differently, using the various rules that exist for each SDP line.
This API changes the local media state; among other things, it sets
up local resources for receiving and decoding media. In order to
successfully handle scenarios where the application wants to offer to
change from one media format to a different, incompatible format, the
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PeerConnection must be able to simultaneously support use of both the
old and new local descriptions (e.g. support codecs that exist in
both descriptions) until a final answer is received, at which point
the PeerConnection can fully adopt the new local description, or roll
back to the old description if the remote side denied the change.
This API indirectly controls the candidate gathering process. When a
local description is supplied, and the number of transports currently
in use does not match the number of transports needed by the local
description, the PeerConnection will create transports as needed and
begin gathering candidates for them.
If setRemoteDescription was previous called with an offer, and
setLocalDescription is called with an answer (provisional or final),
and the media directions are compatible, and media are available to
send, this will result in the starting of media transmission.
4.1.5. setRemoteDescription
The setRemoteDescription method instructs the PeerConnection to apply
the supplied SDP blob as the desired remote configuration. As in
setLocalDescription, the type field of the indicates how the blob
should be processed.
This API changes the local media state; among other things, it sets
up local resources for sending and encoding media.
If setRemoteDescription was previously called with an offer, and
setLocalDescription is called with an answer (provisional or final),
and the media directions are compatible, and media are available to
send, this will result in the starting of media transmission.
4.1.6. localDescription
The localDescription method returns a copy of the current local
configuration, i.e. what was most recently passed to
setLocalDescription, plus any local candidates that have been
generated by the ICE Agent.
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
been established, or if the PeerConnection has been closed.
4.1.7. remoteDescription
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The remoteDescription method returns a copy of the current remote
configuration, i.e. what was most recently passed to
setRemoteDescription, plus any remote candidates that have been
supplied via processIceMessage.
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
been established, or if the PeerConnection has been closed.
4.1.8. updateIce
The updateIce method allows the configuration of the ICE Agent to be
changed during the session, primarily for changing which types of
local candidates are provided to the application and used for
connectivity checks. A callee may initially configure the ICE Agent
to use only relay candidates, to avoid leaking location information,
but update this configuration to use all candidates once the call is
accepted.
Regardless of the configuration, the gathering process collects all
available candidates, but excluded candidates will not be surfaced in
onicecandidate callback or used for connectivity checks.
This call may result in a change to the state of the ICE Agent, and
may result in a change to media state if it results in connectivity
being established.
4.1.9. addIceCandidate
The addIceCandidate method provides a remote candidate to the ICE
Agent, which, if parsed successfully, will be added to the remote
description according to the rules defined for Trickle ICE.
Connectivity checks will be sent to the new candidate.
This call will result in a change to the state of the ICE Agent, and
may result in a change to media state if it results in connectivity
being established.
5. SDP Interaction Procedures
This section describes the specific procedures to be followed when
creating and parsing SDP objects.
5.1. SDP Requirements Overview
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The key specifications that govern creation and processing of offers
and answers are listed below. This list is derived from
[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.
R-9 [RFC5763] MUST be implemented to signal DTLS certificate
fingerprints.
R-10 [RFC5506] MAY be implemented to signal Reduced-Size RTCP
messages.
R-11 [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.
R-12 [RFC4568] MUST NOT be implemented to signal SDES SRTP keying
information.
R-13 A [I-D.ietf-mmusic-msid] MUST be supported, in order to signal
associations between RTP objects and W3C MediaStreams and
MediaStreamTracks in a standard way.
R-14 The bundle mechanism in
[I-D.ietf-mmusic-sdp-bundle-negotiation] MUST be supported to
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signal the use or multiplexing RTP somethings on a single UDP
port, in order to avoid excessive use of port number resources.
As required by [RFC4566] Section 5.13 JSEP implementations MUST
ignore unknown attributes (a=) lines.
Example SDP for RTCWeb call flows can be found in
[I-D.nandakumar-rtcweb-sdp]. [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. Constructing an Offer
When createOffer is called, a new SDP description must be created
that includes the functionality specified in
[I-D.ietf-rtcweb-rtp-usage]. The exact details of this process are
explained below.
5.2.1. Initial Offers
When createOffer is called for the first time, the result is known as
the initial offer.
The first step in generating an initial offer is to generate session-
level attributes, as specified in [RFC4566], Section 5.
Specifically:
o The first SDP line MUST be "v=0", as specified in [RFC4566],
Section 5.1
o The second SDP line MUST be an "o=" line, as specified in
[RFC4566], Section 5.2. The value of the <username> field SHOULD
be "-". The value of the <sess-id> field SHOULD be a
cryptographically random number. To ensure uniqueness, this
number SHOULD be at least 64 bits long. The value of the <sess-
version> field SHOULD be zero. The value of the <nettype>
<addrtype> <unicast-address> tuple SHOULD be set to a non-
meaningful address, such as IN IP4 0.0.0.0, to prevent leaking the
local address in this field. As mentioned in [RFC4566], the
entire o= line needs to be unique, but selecting a random number
for <sess-id> is sufficient to accomplish this.
o The third SDP line MUST be a "s=" line, as specified in [RFC4566],
Section 5.3; a single space SHOULD be used as the session name,
e.g. "s= "
o Session Information ("i="), URI ("u="), Email Address ("e="),
Phone Number ("p="), Bandwidth ("b="), Repeat Times ("r="), and
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Time Zones ("z=") lines are not useful in this context and SHOULD
NOT be included.
o Encryption Keys ("k=") lines do not provide sufficient security
and MUST NOT be included.
o A "t=" line MUST be added, as specified in [RFC4566], Section 5.9;
both <start-time> and <stop-time> SHOULD be set to zero, e.g. "t=0
0".
The next step is to generate m= sections for each MediaStreamTrack
that has been added to the PeerConnection via the addStream method.
Note that this method takes a MediaStream, which can contain multiple
MediaStreamTracks, and therefore multiple m= sections can be
generated even if addStream is only called once.
Each m= section should be generated as specified in [RFC4566],
Section 5.14. The <proto> field MUST be set to "RTP/SAVPF". If a m=
section is not being bundled into another m= section, it MUST
generate a unique set of ICE credentials and gather its own set of
candidates. Otherwise, it MUST use the same ICE credentials and
candidates that were used in the m= section that it is being bundled
into. For DTLS, all m= sections MUST use the same certificate [OPEN
ISSUE: how this is configured] and will therefore have the same
fingerprint values.
Each m= section MUST include the following:
o An "a=mid" line, as specified in [RFC5888], Section 4.
o An "a=msid" line, as specified in [I-D.ietf-mmusic-msid],
Section 2.
o [OPEN ISSUE: Use of App Token versus stream-correlator ]
o An "a=sendrecv" line, as specified in [RFC3264], Section 5.1.
o For each supported codec, "a=rtpmap" and "a=fmtp" lines, as
specified in [RFC4566], Section 6. For audio, the codecs
specified in [I-D.ietf-rtcweb-audio], Section 3, MUST be be
supported.
o For each primary codec where RTP retransmission should be used, a
corresponding "a=rtpmap" line indicating "rtx" with the clock rate
of the primary codec and an "a=fmtp" line that references the
payload type fo the primary codec, as specified in [RFC4588],
Section 8.1.
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o For each supported FEC mechanism, a corresponding "a=rtpmap" line
indicating the desired FEC codec.
o "a=ice-ufrag" and "a=ice-passwd" lines, as specified in [RFC5245],
Section 15.4.
o An "a=ice-options" line, with the "trickle" option, as specified
in [I-D.ivov-mmusic-trickle-ice], Section 4.
o For each candidate that has been gathered during the most recent
gathering phase, an "a=candidate" line, as specified in [RFC5245],
Section 4.3., paragraph 3.
o For the current default candidate, a "c=" line, as specific in
[RFC5245], Section 4.3., paragraph 6. [OPEN ISSUE, pending
resolution in mmusic: If no candidates have yet been gathered yet,
the default candidate should be set to the null value defined in
[I-D.ivov-mmusic-trickle-ice], Section 5.1.]
o An "a=fingerprint" line, as specified in [RFC4572], Section 5.
Use of the SHA-256 algorithm for the fingerprint is REQUIRED; if
the browser also supports stronger hashes, additional
"a=fingerprint" lines with these hashes MAY also be added.
o An "a=setup" line, as specified in [RFC4145], Section 4, and
clarified for use in DTLS-SRTP scenarios in [RFC5763], Section 5.
The role value in the offer MUST be "actpass".
o An "a=rtcp-mux" line, as specified in [RFC5761], Section 5.1.1.
o An "a=rtcp-rsize" line, as specified in [RFC5506], Section 5.
o For each supported RTP header extension, an "a=extmap" line, as
specified in [RFC5285], Section 5. The list of header extensions
that SHOULD/MUST be supported is specified in
[I-D.ietf-rtcweb-rtp-usage], Section 5.2. Any header extensions
that require encryption MUST be specified as indicated in
[RFC6904], Section 4.
o For each supported RTCP feedback mechanism, an "a=rtcp-fb"
mechanism, as specified in [RFC4585], Section 4.2. The list of
RTCP feedback mechanisms that SHOULD/MUST be supported is
specified in [I-D.ietf-rtcweb-rtp-usage], Section 5.1.
o An "a=ssrc" line, as specified in [RFC5576], Section 4.1,
indicating the SSRC to be used for sending media.
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o If RTX is supported for this media type, another "a=ssrc" line
with the RTX SSRC, and an "a=ssrc-group" line, as specified in
[RFC5576], section 4.2, with semantics set to "FID" and including
the primary and RTX SSRCs.
o If FEC is supported for this media type, another "a=ssrc" line
with the FEC SSRC, and an "a=ssrc-group" line, as specified in
[RFC5576], section 4.2, with semantics set to "FEC" and including
the primary and FEC SSRCs.
o [OPEN ISSUE: Handling of a=imageattr]
o [TODO: bundle-only]
Lastly, if a data channel has been created, a m= section MUST be
generated for data. The <media> field MUST be set to "application"
and the <proto> field MUST be set to "DTLS/SCTP", as specified in
[I-D.ietf-mmusic-sctp-sdp], Section 3. The "a=mid", "a=ice-ufrag",
"a=ice-passwd", "a=ice-options", "a=candidate", "a=fingerprint", and
"a=setup" lines MUST be included as mentioned above. [OPEN ISSUE:
additional SCTP-specific stuff to be included, as indicated in
[I-D.jesup-rtcweb-data-protocol] (currently none)]
Once all m= sections have been generated, a session-level "a=group"
attribute MUST be added as specified in [RFC5888]. This attribute
MUST have semantics "BUNDLE", and identify the m= sections to be
bundled. [OPEN ISSUE: Need to determine exactly how this decision is
made.]
Attributes that are common between all m= sections MAY be moved to
session-level, if desired.
Attributes other than the ones specified above MAY be included,
except for the following attributes which are specifically
incompatible with the requirements of [I-D.ietf-rtcweb-rtp-usage],
and MUST NOT be included:
o "a=crypto"
o "a=key-mgmt"
o "a=ice-lite"
Note that when BUNDLE is used, any additional attributes that are
added MUST follow the advice in
[I-D.nandakumar-mmusic-sdp-mux-attributes] on how those attributes
interact with BUNDLE.
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5.2.2. Subsequent Offers
When createOffer is called a second (or later) time, the processing
is different, depending on the current signaling state.
If the initial offer was not applied using setLocalDescription,
meaning the PeerConnection is still in the "stable" state, the steps
for generating an initial offer should be followed, with this
exception:
o The "o=" line MUST stay the same.
If the initial offer was applied using setLocalDescription, but an
answer from the remote side has not yet been applied, meaning the
PeerConnection is still in the "local-offer" state, the steps for
generating an initial offer should be followed, with these
exceptions:
o The "o=" line MUST stay the same, except for the <session-version>
field, which MUST increase by 1 from the previously applied local
description.
o The "s=" and "t=" lines MUST stay the same.
o Each "a=mid" line MUST stay the same.
o Each "a=ice-ufrag" and "a=ice-pwd" line MUST stay the same.
o For MediaStreamTracks that are still present, the "a=msid",
"a=ssrc", and "a=ssrc-group" lines MUST stay the same.
o If any MediaStreamTracks have been removed, either through the
removeStream method or by removing them from an added MediaStream,
their m= sections MUST be marked as recvonly by changing the value
of the [RFC3264] directional attribute to "a=recvonly". The
"a=msid", "a=ssrc", and "a=ssrc-group" lines MUST be removed from
the associated m= sections.
If the initial offer was applied using setLocalDescription, and an
answer from the remote side has been applied using
setRemoteDescription, meaning the PeerConnection is in the "remote-
pranswer" or "stable" states, an offer is generated based on the
negotiated session descriptions by following the steps mentioned for
the "local-offer" state above, along with these exceptions: [OPEN
ISSUE: should this be permitted in the remote-pranswer state?]
o If a m= section was rejected, i.e. has had its port set to zero in
either the local or remote description, it MUST remain rejected
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and have a zero port in the new offer, as indicated in RFC3264,
Section 5.1.
o If a m= section exists in the current local description, but has
its state set to inactive or recvonly, and a new MediaStreamTrack
is added, the previously existing m= section MUST be recycled
instead of creating a new m= section. [OPEN ISSUE: Nail down
exactly what this means. Should the codecs remain the same?
(No.) Should ICE restart? (No.) Can the "a=mid" attribute be
changed? (Yes?)]
o If a m= section exists in the current local description, but does
not have an associated MediaStreamTrack (i.e. it is inactive or
recvonly), a corresponding m= section MUST be generated in the new
offer, but without "a=msid", "a=ssrc", or "a=ssrc-group"
attributes, and the appropriate directional attribute must be
specified.
In addition, for each previously existing, non-rejected m= section in
the new offer, the following adjustments are made based on the
contents of the corresponding m= section in the current remote
description:
o The m= line and corresponding "a=rtpmap" and "a=fmtp" lines MUST
only include codecs present in the remote description.
o The RTP header extensions MUST only include those that are present
in the remote description.
o The RTCP feedback extensions MUST only include those that are
present in the remote description.
o The "a=rtcp-mux" line MUST only be added if present in the remote
description.
o The "a=rtcp-rsize" line MUST only be added if present in the
remote description.
5.2.3. Constraints Handling
The createOffer method takes as a parameter a MediaConstraints
object. Special processing is performed when generating a SDP
description if the following constraints are present.
5.2.3.1. OfferToReceiveAudio
If the "OfferToReceiveAudio" constraint is specified, with a value of
"true", the offer MUST include a non-rejected m= section with media
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type "audio", even if no audio MediaStreamTrack has been added to the
PeerConnection. This allows the offerer to receive audio even when
not sending it; accordingly, the directional attribute on the audio
m= section MUST be set to recvonly. If this constraint is specified
when an audio MediaStreamTrack has already been added to the
PeerConnection, or a non-rejected m= section with media type "audio"
previously existed, it has no effect.
5.2.3.2. OfferToReceiveVideo
If the "OfferToReceiveAudio" constraint is specified, with a value of
"true", the offer MUST include a m= section with media type "video",
even if no video MediaStreamTrack has been added to the
PeerConnection. This allows the offerer to receive video even when
not sending it; accordingly, the directional attribute on the video
m= section MUST be set to recvonly. If this constraint is specified
when an video MediaStreamTrack has already been added to the
PeerConnection, or a non-rejected m= section with media type "video"
previously existed, it has no effect.
5.2.3.3. VoiceActivityDetection
If the "VoiceActivityDetection" constraint is specified, with a value
of "true", the offer MUST indicate support for silence suppression by
including comfort noise ("CN") codecs for each supported clock rate,
as specified in [RFC3389], Section 5.1. [OPEN issue: should this do
anything in signaling, or should it just control built-in DTX modes
in audio codecs? Opus has built-in DTX, but G.711 does not.]
5.2.3.4. IceRestart
If the "IceRestart" constraint is specified, with a value of "true",
the offer MUST indicate an ICE restart by generating new ICE ufrag
and pwd attributes, as specified in RFC5245, Section 9.1.1.1. If
this constraint is specified on an initial offer, it has no effect
(since a new ICE ufrag and pwd are already generated).
5.3. Generating an Answer
When createAnswer is called, a new SDP description must be created
that is compatible with the supplied remote description as well as
the requirements specified in [I-D.ietf-rtcweb-rtp-usage]. The exact
details of this process are explained below.
5.3.1. Initial Answers
When createAnswer is called for the first time after a remote
description has been provided, the result is known as the initial
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answer. If no remote description has been installed, an answer
cannot be generated, and an error MUST be returned.
Note that the remote description SDP may not have been created by a
WebRTC endpoint and may not conform to all the requirements listed in
Section 5.2. For many cases, this is not a problem. However, if any
mandatory SDP attributes are missing, or functionality listed as
mandatory-to-use is not present (e.g. ICE, DTLS) [TODO: find
reference for this], this MUST be treated as an error. [OPEN ISSUE:
Should this cause setRemoteDescription to fail, or should this cause
createAnswer to reject those particular m= sections?]
The first step in generating an initial answer is to generate
session-level attributes. The process here is identical to that
indicated in the Initial Offers section above, with the addition that
The next step is to generate m= sections for each m= section that is
present in the remote offer, as specified in [RFC3264], Section 6.
For the purposes of this discussion, any session-level attributes in
the offer that are also valid as media-level attributes SHALL be
considered to be present in each m= section.
If any of the offered m= sections have been rejected, by stopping the
associated remote MediaStreamTrack, the corresponding m= section in
the answer MUST be marked as rejected by setting the port in the m=
line to zero, as indicated in [RFC3264], Section 6., and processing
continues with the next m= section.
For each non-rejected m= section of a given media type, if there is a
local MediaStreamTrack of the specified type which has been added to
the PeerConnection via addStream and not yet associated with a m=
section, the MediaStreamTrack is associated with the m= section at
this time. If there are more m= sections of a certain type than
MediaStreamTracks, some m= sections will not have an associated
MediaStreamTrack. If there are more MediaStreamTracks of a certain
type than m= sections, only the first N MediaStreamTracks will be
able to be associated in the constructed answer. The remainder will
need to be associated in a subsequent offer.
Each m= section should then generated as specified in [RFC3264],
Section 6.1. The <proto> field MUST be set to "RTP/SAVPF". If the
offer supports BUNDLE, all m= sections to be BUNDLEd must use the
same ICE credentials and candidates; all m= sections not being
BUNDLEd must use unique ICE credentials and candidates. Each m=
section MUST include the following:
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o If present in the offer, an "a=mid" line, as specified in
[RFC5888], Section 9.1. The "mid" value MUST match that specified
in the offer.
o If a local MediaStreamTrack has been associated, an "a=msid" line,
as specified in [I-D.ietf-mmusic-msid], Section 2.
o [OPEN ISSUE: Use of App Token versus stream-correlator ]
o If a local MediaStreamTrack has been associated, an "a=sendrecv"
line, as specified in [RFC3264], Section 6.1. If no local
MediaStreamTrack has been associated, an "a=recvonly" line.
[TODO: handle non-sendrecv offered m= sections]
o For each supported codec that is present in the offer, "a=rtpmap"
and "a=fmtp" lines, as specified in [RFC4566], Section 6, and
[RFC3264], Section 6.1. For audio, the codecs specified in
[I-D.ietf-rtcweb-audio], Section 3, MUST be be supported. Note
that for simplicity, the answerer MAY use different payload types
for codecs than the offerer, as it is not prohibited by
Section 6.1.
o If "rtx" is present in the offer, for each primary codec where RTP
retransmission should be used, a corresponding "a=rtpmap" line
indicating "rtx" with the clock rate of the primary codec and an
"a=fmtp" line that references the payload type fo the primary
codec, as specified in [RFC4588], Section 8.1.
o For each supported FEC mechanism that is present in the offer, a
corresponding "a=rtpmap" line indicating the desired FEC codec.
o "a=ice-ufrag" and "a=ice-passwd" lines, as specified in [RFC5245],
Section 15.4.
o If the "trickle" ICE option is present in the offer, an "a=ice-
options" line, with the "trickle" option, as specified in
[I-D.ivov-mmusic-trickle-ice], Section 4.
o For each candidate that has been gathered during the most recent
gathering phase, an "a=candidate" line, as specified in [RFC5245],
Section 4.3., paragraph 3.
o For the current default candidate, a "c=" line, as specific in
[RFC5245], Section 4.3., paragraph 6. [OPEN ISSUE, pending
resolution in mmusic: If no candidates have yet been gathered yet,
the default candidate should be set to the null value defined in
[I-D.ivov-mmusic-trickle-ice], Section 5.1.]
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o An "a=fingerprint" line, as specified in [RFC4572], Section 5.
Use of the SHA-256 algorithm for the fingerprint is REQUIRED; if
the browser also supports stronger hashes, additional
"a=fingerprint" lines with these hashes MAY also be added.
o An "a=setup" line, as specified in [RFC4145], Section 4, and
clarified for use in DTLS-SRTP scenarios in [RFC5763], Section 5.
The role value in the answer MUST be "active" or "passive"; the
"active" role is RECOMMENDED.
o If present in the offer, an "a=rtcp-mux" line, as specified in
[RFC5761], Section 5.1.1.
o If present in the offer, an "a=rtcp-rsize" line, as specified in
[RFC5506], Section 5.
o For each supported RTP header extension that is present in the
offer, an "a=extmap" line, as specified in [RFC5285], Section 5.
The list of header extensions that SHOULD/MUST be supported is
specified in [I-D.ietf-rtcweb-rtp-usage], Section 5.2. Any header
extensions that require encryption MUST be specified as indicated
in [RFC6904], Section 4.
o For each supported RTCP feedback mechanism that is present in the
offer, an "a=rtcp-fb" mechanism, as specified in [RFC4585],
Section 4.2. The list of RTCP feedback mechanisms that SHOULD/
MUST be supported is specified in [I-D.ietf-rtcweb-rtp-usage],
Section 5.1.
o If a local MediaStreamTrack has been associated, an "a=ssrc" line,
as specified in [RFC5576], Section 4.1, indicating the SSRC to be
used for sending media.
o If a local MediaStreamTrack has been associated, and RTX has been
negotiated for this m= section, another "a=ssrc" line with the RTX
SSRC, and an "a=ssrc-group" line, as specified in [RFC5576],
section 4.2, with semantics set to "FID" and including the primary
and RTX SSRCs.
o If a local MediaStreamTrack has been associated, and FEC has been
negotiated for this m= section, another "a=ssrc" line with the FEC
SSRC, and an "a=ssrc-group" line, as specified in [RFC5576],
section 4.2, with semantics set to "FEC" and including the primary
and FEC SSRCs.
o [OPEN ISSUE: Handling of a=imageattr]
o [TODO: bundle-only]
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If a data channel m= section has been offered, a m= section MUST also
be generated for data. The <media> field MUST be set to
"application" and the <proto> field MUST be set to "DTLS/SCTP", as
specified in [I-D.ietf-mmusic-sctp-sdp], Section 3. The "a=mid", "a
=ice-ufrag", "a=ice-passwd", "a=ice-options", "a=candidate",
"a=fingerprint", and "a=setup" lines MUST be included as mentioned
above. [OPEN ISSUE: additional SCTP-specific stuff to be included,
as indicated in [I-D.jesup-rtcweb-data-protocol] (currently none)]
[TODO: processing of BUNDLE group]
Attributes that are common between all m= sections MAY be moved to
session-level, if desired.
The attributes prohibited in creation of offers are also prohibited
in the creation of answers.
5.3.2. Subsequent Answers
5.3.3. Constraints Handling
5.4. Parsing an Offer
5.5. Parsing an Answer
5.6. Applying a Local Description
5.7. Applying a Remote Description
6. Configurable SDP Parameters
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.
The following elements of the SDP media description MUST NOT be
changed between the createOffer and the setLocalDescription, since
they reflect transport attributes that are solely under browser
control, and the browser MUST NOT honor an attempt to change them:
o The number, type and port number of m-lines.
o The generated ICE credentials (a=ice-ufrag and a=ice-pwd).
o The set of ICE candidates and their parameters (a=candidate).
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The following modifications, if done by the browser to a description
between createOffer/createAnswer and the setLocalDescription, MUST be
honored by the browser:
o Remove or reorder codecs (m=)
The following parameters may be controlled by constraints passed into
createOffer/createAnswer. As an open issue, these changes may also
be be performed by manipulating the SDP returned from createOffer/
createAnswer, as indicated above, as long as the capabilities of the
endpoint are not exceeded (e.g. asking for a resolution greater than
what the endpoint can encode):
o disable BUNDLE (a=group)
o disable RTCP mux (a=rtcp-mux)
o change send resolution or frame rate
o change desired recv resolution or frame rate
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)
For example, an application could implement call hold by adding an
a=inactive attribute to its local description, and then applying and
signaling that description.
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.
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7. Security Considerations
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
This document requires no actions from IANA.
9. Acknowledgements
Significant text incorporated in the draft as well and review was
provided by Harald Alvestrand and Suhas Nandakumar. Dan Burnett,
Neil Stratford, Eric Rescorla, Anant Narayanan, Andrew Hutton,
Richard Ejzak, and Adam Bergkvist all provided valuable feedback on
this proposal. Matthew Kaufman provided the observation that keeping
state out of the browser allows a call to continue even if the page
is reloaded.
10. References
10.1. Normative References
[I-D.ietf-mmusic-msid]
Alvestrand, H., "Cross Session Stream Identification in
the Session Description Protocol", draft-ietf-mmusic-
msid-01 (work in progress), August 2013.
[I-D.ietf-mmusic-sctp-sdp]
Loreto, S. and G. Camarillo, "Stream Control Transmission
Protocol (SCTP)-Based Media Transport in the Session
Description Protocol (SDP)", draft-ietf-mmusic-sctp-sdp-04
(work in progress), June 2013.
[I-D.ietf-mmusic-sdp-bundle-negotiation]
Holmberg, C., Alvestrand, H., and C. Jennings,
"Multiplexing Negotiation Using Session Description
Protocol (SDP) Port Numbers", draft-ietf-mmusic-sdp-
bundle-negotiation-04 (work in progress), June 2013.
[I-D.ietf-rtcweb-audio]
Valin, J. and C. Bran, "WebRTC Audio Codec and Processing
Requirements", draft-ietf-rtcweb-audio-02 (work in
progress), August 2013.
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[I-D.ietf-rtcweb-rtp-usage]
Perkins, C., Westerlund, M., and J. Ott, "Web Real-Time
Communication (WebRTC): Media Transport and Use of RTP",
draft-ietf-rtcweb-rtp-usage-09 (work in progress),
September 2013.
[I-D.nandakumar-mmusic-sdp-mux-attributes]
Nandakumar, S., "A Framework for SDP Attributes when
Multiplexing", draft-nandakumar-mmusic-sdp-mux-
attributes-03 (work in progress), July 2013.
[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
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
with Session Description Protocol (SDP)", RFC 3264, June
2002.
[RFC4145] Yon, D. and G. Camarillo, "TCP-Based Media Transport in
the Session Description Protocol (SDP)", RFC 4145,
September 2005.
[RFC4566] Handley, M., Jacobson, V., and C. Perkins, "SDP: Session
Description Protocol", RFC 4566, July 2006.
[RFC4572] Lennox, J., "Connection-Oriented Media Transport over the
Transport Layer Security (TLS) Protocol in the Session
Description Protocol (SDP)", RFC 4572, July 2006.
[RFC4585] Ott, J., Wenger, S., Sato, N., Burmeister, C., and J. Rey,
"Extended RTP Profile for Real-time Transport Control
Protocol (RTCP)-Based Feedback (RTP/AVPF)", RFC 4585, July
2006.
[RFC5124] Ott, J. and E. Carrara, "Extended Secure RTP Profile for
Real-time Transport Control Protocol (RTCP)-Based Feedback
(RTP/SAVPF)", RFC 5124, February 2008.
[RFC5245] Rosenberg, J., "Interactive Connectivity Establishment
(ICE): A Protocol for Network Address Translator (NAT)
Traversal for Offer/Answer Protocols", RFC 5245, April
2010.
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[RFC5285] Singer, D. and H. Desineni, "A General Mechanism for RTP
Header Extensions", RFC 5285, July 2008.
[RFC5761] Perkins, C. and M. Westerlund, "Multiplexing RTP Data and
Control Packets on a Single Port", RFC 5761, April 2010.
[RFC5888] Camarillo, G. and H. Schulzrinne, "The Session Description
Protocol (SDP) Grouping Framework", RFC 5888, June 2010.
[RFC6904] Lennox, J., "Encryption of Header Extensions in the Secure
Real-time Transport Protocol (SRTP)", RFC 6904, April
2013.
10.2. Informative References
[I-D.ivov-mmusic-trickle-ice]
Ivov, E., Rescorla, E., and J. Uberti, "Trickle ICE:
Incremental Provisioning of Candidates for the Interactive
Connectivity Establishment (ICE) Protocol", draft-ivov-
mmusic-trickle-ice-01 (work in progress), March 2013.
[I-D.jennings-rtcweb-signaling]
Jennings, C., Rosenberg, J., and R. Jesup, "RTCWeb Offer/
Answer Protocol (ROAP)", draft-jennings-rtcweb-
signaling-01 (work in progress), October 2011.
[I-D.jesup-rtcweb-data-protocol]
Jesup, R., Loreto, S., and M. Tuexen, "WebRTC Data Channel
Protocol", draft-jesup-rtcweb-data-protocol-04 (work in
progress), February 2013.
[I-D.nandakumar-rtcweb-sdp]
Nandakumar, S. and C. Jennings, "SDP for the WebRTC",
draft-nandakumar-rtcweb-sdp-02 (work in progress), July
2013.
[RFC3389] Zopf, R., "Real-time Transport Protocol (RTP) Payload for
Comfort Noise (CN)", RFC 3389, September 2002.
[RFC3556] Casner, S., "Session Description Protocol (SDP) Bandwidth
Modifiers for RTP Control Protocol (RTCP) Bandwidth", RFC
3556, July 2003.
[RFC3960] Camarillo, G. and H. Schulzrinne, "Early Media and Ringing
Tone Generation in the Session Initiation Protocol (SIP)",
RFC 3960, December 2004.
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[RFC4568] Andreasen, F., Baugher, M., and D. Wing, "Session
Description Protocol (SDP) Security Descriptions for Media
Streams", RFC 4568, July 2006.
[RFC4588] Rey, J., Leon, D., Miyazaki, A., Varsa, V., and R.
Hakenberg, "RTP Retransmission Payload Format", RFC 4588,
July 2006.
[RFC5506] Johansson, I. and M. Westerlund, "Support for Reduced-Size
Real-Time Transport Control Protocol (RTCP): Opportunities
and Consequences", RFC 5506, April 2009.
[RFC5576] Lennox, J., Ott, J., and T. Schierl, "Source-Specific
Media Attributes in the Session Description Protocol
(SDP)", RFC 5576, June 2009.
[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]
Bergkvist, A., Burnett, D., Narayanan, A., and C.
Jennings, "WebRTC 1.0: Real-time Communication Between
Browsers", World Wide Web Consortium WD WD-
webrtc-20111027, October 2011,
<http://www.w3.org/TR/2011/WD-webrtc-20111027>.
Appendix A. JSEP Implementation Examples
A.1. Example API Flows
Below are several sample flows for the new PeerConnection and library
APIs, demonstrating when the various APIs are called in different
situations and with various transport protocols. For clarity and
simplicity, the createOffer/createAnswer calls are assumed to be
synchronous in these examples, whereas the actual APIs are async.
A.1.1. Call using ROAP
This example demonstrates a ROAP call, without the use of trickle
candidates.
// Call is initiated toward Answerer
OffererJS->OffererUA: pc = new PeerConnection();
OffererJS->OffererUA: pc.addStream(localStream, null);
OffererUA->OffererJS: iceCallback(candidate);
OffererJS->OffererUA: offer = pc.createOffer(null);
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OffererJS->OffererUA: pc.setLocalDescription("offer", offer);
OffererJS->AnswererJS: {"type":"OFFER", "sdp":offer }
// OFFER arrives at Answerer
AnswererJS->AnswererUA: pc = new PeerConnection();
AnswererJS->AnswererUA: pc.setRemoteDescription("offer", msg.sdp);
AnswererUA->AnswererJS: onaddstream(remoteStream);
AnswererUA->OffererUA: iceCallback(candidate);
// Answerer accepts call
AnswererJS->AnswererUA: pc.addStream(localStream, null);
AnswererJS->AnswererUA: answer = pc.createAnswer(msg.sdp, null);
AnswererJS->AnswererUA: pc.setLocalDescription("answer", answer);
AnswererJS->OffererJS: {"type":"ANSWER","sdp":answer }
// ANSWER arrives at Offerer
OffererJS->OffererUA: pc.setRemoteDescription("answer", answer);
OffererUA->OffererJS: onaddstream(remoteStream);
// ICE Completes (at Answerer)
AnswererUA->OffererUA: Media
// ICE Completes (at Offerer)
OffererJS->AnswererJS: {"type":"OK" }
OffererUA->AnswererUA: Media
A.1.2. Call using XMPP
This example demonstrates an XMPP call, making use of trickle
candidates.
// Call is initiated toward Answerer
OffererJS->OffererUA: pc = new PeerConnection();
OffererJS->OffererUA: pc.addStream(localStream, null);
OffererJS->OffererUA: offer = pc.createOffer(null);
OffererJS->OffererUA: pc.setLocalDescription("offer", offer);
OffererJS: xmpp = createSessionInitiate(offer);
OffererJS->AnswererJS: <jingle action="session-initiate"/>
OffererJS->OffererUA: pc.startIce();
OffererUA->OffererJS: onicecandidate(cand);
OffererJS: createTransportInfo(cand);
OffererJS->AnswererJS: <jingle action="transport-info"/>
// session-initiate arrives at Answerer
AnswererJS->AnswererUA: pc = new PeerConnection();
AnswererJS: offer = parseSessionInitiate(xmpp);
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AnswererJS->AnswererUA: pc.setRemoteDescription("offer", offer);
AnswererUA->AnswererJS: onaddstream(remoteStream);
// transport-infos arrive at Answerer
AnswererJS->AnswererUA: candidate = parseTransportInfo(xmpp);
AnswererJS->AnswererUA: pc.addIceCandidate(candidate);
AnswererUA->AnswererJS: onicecandidate(cand)
AnswererJS: createTransportInfo(cand);
AnswererJS->OffererJS: <jingle action="transport-info"/>
// transport-infos arrive at Offerer
OffererJS->OffererUA: candidates = parseTransportInfo(xmpp);
OffererJS->OffererUA: pc.addIceCandidate(candidates);
// Answerer accepts call
AnswererJS->AnswererUA: pc.addStream(localStream, null);
AnswererJS->AnswererUA: answer = pc.createAnswer(offer, null);
AnswererJS: xmpp = createSessionAccept(answer);
AnswererJS->AnswererUA: pc.setLocalDescription("answer", answer);
AnswererJS->OffererJS: <jingle action="session-accept"/>
// session-accept arrives at Offerer
OffererJS: answer = parseSessionAccept(xmpp);
OffererJS->OffererUA: pc.setRemoteDescription("answer", answer);
OffererUA->OffererJS: onaddstream(remoteStream);
// ICE Completes (at Answerer)
AnswererUA->OffererUA: Media
// ICE Completes (at Offerer)
OffererUA->AnswererUA: Media
A.1.3. Adding video to a call, using XMPP
This example demonstrates an XMPP call, where the XMPP content-add
mechanism is used to add video media to an existing session. For
simplicity, candidate exchange is not shown.
Note that the offerer for the change to the session may be different
than the original call offerer.
// Offerer adds video stream
OffererJS->OffererUA: pc.addStream(videoStream)
OffererJS->OffererUA: offer = pc.createOffer(null);
OffererJS: xmpp = createContentAdd(offer);
OffererJS->OffererUA: pc.setLocalDescription("offer", offer);
OffererJS->AnswererJS: <jingle action="content-add"/>
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// content-add arrives at Answerer
AnswererJS: offer = parseContentAdd(xmpp);
AnswererJS->AnswererUA: pc.setRemoteDescription("offer", offer);
AnswererJS->AnswererUA: answer = pc.createAnswer(offer, null);
AnswererJS->AnswererUA: pc.setLocalDescription("answer", answer);
AnswererJS: xmpp = createContentAccept(answer);
AnswererJS->OffererJS: <jingle action="content-accept"/>
// content-accept arrives at Offerer
OffererJS: answer = parseContentAccept(xmpp);
OffererJS->OffererUA: pc.setRemoteDescription("answer", answer);
A.1.4. Simultaneous add of video streams, using XMPP
This example demonstrates an XMPP call, where new video sources are
added at the same time to a call that already has video; since adding
these sources only affects one side of the call, there is no
conflict. The XMPP description-info mechanism is used to indicate
the new sources to the remote side.
// Offerer and "Answerer" add video streams at the same time
OffererJS->OffererUA: pc.addStream(offererVideoStream2)
OffererJS->OffererUA: offer = pc.createOffer(null);
OffererJS: xmpp = createDescriptionInfo(offer);
OffererJS->OffererUA: pc.setLocalDescription("offer", offer);
OffererJS->AnswererJS: <jingle action="description-info"/>
AnswererJS->AnswererUA: pc.addStream(answererVideoStream2)
AnswererJS->AnswererUA: offer = pc.createOffer(null);
AnswererJS: xmpp = createDescriptionInfo(offer);
AnswererJS->AnswererUA: pc.setLocalDescription("offer", offer);
AnswererJS->OffererJS: <jingle action="description-info"/>
// description-info arrives at "Answerer", and is acked
AnswererJS: offer = parseDescriptionInfo(xmpp);
AnswererJS->OffererJS: <iq type="result"/> // ack
// description-info arrives at Offerer, and is acked
OffererJS: offer = parseDescriptionInfo(xmpp);
OffererJS->AnswererJS: <iq type="result"/> // ack
// ack arrives at Offerer; remote offer is used as an answer
OffererJS->OffererUA: pc.setRemoteDescription("answer", offer);
// ack arrives at "Answerer"; remote offer is used as an answer
AnswererJS->AnswererUA: pc.setRemoteDescription("answer", offer);
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A.1.5. Call using SIP
This example demonstrates a simple SIP call (e.g. where the client
talks to a SIP proxy over WebSockets).
// Call is initiated toward Answerer
OffererJS->OffererUA: pc = new PeerConnection();
OffererJS->OffererUA: pc.addStream(localStream, null);
OffererUA->OffererJS: onicecandidate(candidate);
OffererJS->OffererUA: offer = pc.createOffer(null);
OffererJS->OffererUA: pc.setLocalDescription("offer", offer);
OffererJS: sip = createInvite(offer);
OffererJS->AnswererJS: SIP INVITE w/ SDP
// INVITE arrives at Answerer
AnswererJS->AnswererUA: pc = new PeerConnection();
AnswererJS: offer = parseInvite(sip);
AnswererJS->AnswererUA: pc.setRemoteDescription("offer", offer);
AnswererUA->AnswererJS: onaddstream(remoteStream);
AnswererUA->OffererUA: onicecandidate(candidate);
// Answerer accepts call
AnswererJS->AnswererUA: pc.addStream(localStream, null);
AnswererJS->AnswererUA: answer = pc.createAnswer(offer, null);
AnswererJS: sip = createResponse(200, answer);
AnswererJS->AnswererUA: pc.setLocalDescription("answer", answer);
AnswererJS->OffererJS: 200 OK w/ SDP
// 200 OK arrives at Offerer
OffererJS: answer = parseResponse(sip);
OffererJS->OffererUA: pc.setRemoteDescription("answer", answer);
OffererUA->OffererJS: onaddstream(remoteStream);
OffererJS->AnswererJS: ACK
// ICE Completes (at Answerer)
AnswererUA->OffererUA: Media
// ICE Completes (at Offerer)
OffererUA->AnswererUA: Media
A.1.6. Handling early media (e.g. 1-800-GO FEDEX), using SIP
This example demonstrates how early media could be handled; for
simplicity, only the offerer side of the call is shown.
// Call is initiated toward Answerer
OffererJS->OffererUA: pc = new PeerConnection();
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OffererJS->OffererUA: pc.addStream(localStream, null);
OffererUA->OffererJS: onicecandidate(candidate);
OffererJS->OffererUA: offer = pc.createOffer(null);
OffererJS->OffererUA: pc.setLocalDescription("offer", offer);
OffererJS: sip = createInvite(offer);
OffererJS->AnswererJS: SIP INVITE w/ SDP
// 180 Ringing is received by offerer, w/ SDP
OffererJS: answer = parseResponse(sip);
OffererJS->OffererUA: pc.setRemoteDescription("pranswer", answer);
OffererUA->OffererJS: onaddstream(remoteStream);
// ICE Completes (at Offerer)
OffererUA->AnswererUA: Media
// 200 OK arrives at Offerer
OffererJS: answer = parseResponse(sip);
OffererJS->OffererUA: pc.setRemoteDescription("answer", answer);
OffererJS->AnswererJS: ACK
A.2. Example Session Descriptions
A.2.1. createOffer
This SDP shows a typical initial offer, created by createOffer for a
PeerConnection with a single audio MediaStreamTrack, a single video
MediaStreamTrack, and a single data channel. Host candidates have
also already been gathered. Note some lines have been broken into
two lines for formatting reasons.
v=0
o=- 4962303333179871722 1 IN IP4 0.0.0.0
s=-
t=0 0
a=group:BUNDLE audio video data
m=audio 56500 RTP/SAVPF 111 0 8 126
c=IN IP4 192.0.2.1
a=rtcp:56501 IN IP4 192.0.2.1
a=candidate:3348148302 1 udp 2113937151 192.0.2.1 56500
typ host generation 0
a=candidate:3348148302 2 udp 2113937151 192.0.2.1 56501
typ host generation 0
a=ice-ufrag:ETEn1v9DoTMB9J4r
a=ice-pwd:OtSK0WpNtpUjkY4+86js7ZQl
a=ice-options:trickle
a=mid:audio
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
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a=sendrecv
a=rtcp-mux
a=rtcp-rsize
a=fingerprint:sha-256
19:E2:1C:3B:4B:9F:81:E6:B8:5C:F4:A5:A8:D8:73:04
:BB:05:2F:70:9F:04:A9:0E:05:E9:26:33:E8:70:88:A2
a=setup:actpass
a=rtpmap:111 opus/48000/2
a=fmtp:111 minptime=10
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:126 telephone-event/8000
a=maxptime:60
a=ssrc:1732846380 cname:EocUG1f0fcg/yvY7
a=msid:47017fee-b6c1-4162-929c-a25110252400
f83006c5-a0ff-4e0a-9ed9-d3e6747be7d9
m=video 56502 RTP/SAVPF 100 115 116 117
c=IN IP4 192.0.2.1
a=rtcp:56503 IN IP4 192.0.2.1
a=candidate:3348148302 1 udp 2113937151 192.0.2.1 56502
typ host generation 0
a=candidate:3348148302 2 udp 2113937151 192.0.2.1 56503
typ host generation 0
a=ice-ufrag:BGKkWnG5GmiUpdIV
a=ice-pwd:mqyWsAjvtKwTGnvhPztQ9mIf
a=ice-options:trickle
a=mid:video
a=extmap:2 urn:ietf:params:rtp-hdrext:toffset
a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
a=sendrecv
a=rtcp-mux
a=rtcp-rsize
a=fingerprint:sha-256
19:E2:1C:3B:4B:9F:81:E6:B8:5C:F4:A5:A8:D8:73:04
:BB:05:2F:70:9F:04:A9:0E:05:E9:26:33:E8:70:88:A2
a=setup:actpass
a=rtpmap:100 VP8/90000
a=rtcp-fb:100 ccm fir
a=rtcp-fb:100 nack
a=rtcp-fb:100 goog-remb
a=rtpmap:115 rtx/90000
a=fmtp:115 apt=100
a=rtpmap:116 red/90000
a=rtpmap:117 ulpfec/90000
a=ssrc:1366781083 cname:EocUG1f0fcg/yvY7
a=ssrc:1366781084 cname:EocUG1f0fcg/yvY7
a=ssrc:1366781085 cname:EocUG1f0fcg/yvY7
a=ssrc-group:FID 1366781083 1366781084
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a=ssrc-group:FEC 1366781083 1366781085
a=msid:61317484-2ed4-49d7-9eb7-1414322a7aae
f30bdb4a-5db8-49b5-bcdc-e0c9a23172e0
m=application 56504 DTLS/SCTP 5000
c=IN IP4 192.0.2.1
a=candidate:3348148302 1 udp 2113937151 192.0.2.1 56504
typ host generation 0
a=ice-ufrag:VD5v2BnbZm3mgP3d
a=ice-pwd:+Jlkuox+VVIUDqxcfIDuTZMH
a=ice-options:trickle
a=mid:data
a=fingerprint:sha-256 19:E2:1C:3B:4B:9F:81:E6:B8:5C:F4:A5:A8:D8:73:04
:BB:05:2F:70:9F:04:A9:0E:05:E9:26:33:E8:70:88:A2
a=setup:actpass
a=fmtp:5000 protocol=webrtc-datachannel; streams=10
A.2.2. createAnswer
This SDP shows a typical initial answer to the above offer, created
by createAnswer for a PeerConnection with a single audio
MediaStreamTrack, a single video MediaStreamTrack, and a single data
channel. Host candidates have also already been gathered. Note some
lines have been broken into two lines for formatting reasons.
v=0
o=- 6729291447651054566 1 IN IP4 0.0.0.0
s=-
t=0 0
a=group:BUNDLE audio video data
m=audio 20000 RTP/SAVPF 111 0 8 126
c=IN IP4 192.0.2.2
a=candidate:2299743422 1 udp 2113937151 192.0.2.2 20000
typ host generation 0
a=ice-ufrag:6sFvz2gdLkEwjZEr
a=ice-pwd:cOTZKZNVlO9RSGsEGM63JXT2
a=fingerprint:sha-256 6B:8B:F0:65:5F:78:E2:51:3B:AC:6F:F3:3F:46:1B:35
:DC:B8:5F:64:1A:24:C2:43:F0:A1:58:D0:A1:2C:19:08
a=setup:active
a=mid:audio
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=sendrecv
a=rtcp-mux
a=rtpmap:111 opus/48000/2
a=fmtp:111 minptime=10
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:126 telephone-event/8000
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a=maxptime:60
a=ssrc:3429951804 cname:Q/NWs1ao1HmN4Xa5
a=msid:PI39StLS8W7ZbQl1sJsWUXkr3Zf12fJUvzQ1
PI39StLS8W7ZbQl1sJsWUXkr3Zf12fJUvzQ1a0
m=video 20000 RTP/SAVPF 100 115 116 117
c=IN IP4 192.0.2.2
a=candidate:2299743422 1 udp 2113937151 192.0.2.2 20000
typ host generation 0
a=ice-ufrag:6sFvz2gdLkEwjZEr
a=ice-pwd:cOTZKZNVlO9RSGsEGM63JXT2
a=fingerprint:sha-256 6B:8B:F0:65:5F:78:E2:51:3B:AC:6F:F3:3F:46:1B:35
:DC:B8:5F:64:1A:24:C2:43:F0:A1:58:D0:A1:2C:19:08
a=setup:active
a=mid:video
a=extmap:2 urn:ietf:params:rtp-hdrext:toffset
a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
a=sendrecv
a=rtcp-mux
a=rtpmap:100 VP8/90000
a=rtcp-fb:100 ccm fir
a=rtcp-fb:100 nack
a=rtcp-fb:100 goog-remb
a=rtpmap:115 rtx/90000
a=fmtp:115 apt=100
a=rtpmap:116 red/90000
a=rtpmap:117 ulpfec/90000
a=ssrc:3229706345 cname:Q/NWs1ao1HmN4Xa5
a=ssrc:3229706346 cname:Q/NWs1ao1HmN4Xa5
a=ssrc:3229706347 cname:Q/NWs1ao1HmN4Xa5
a=ssrc-group:FID 3229706345 3229706346
a=ssrc-group:FEC 3229706345 3229706347
a=msid:PI39StLS8W7ZbQl1sJsWUXkr3Zf12fJUvzQ1
PI39StLS8W7ZbQl1sJsWUXkr3Zf12fJUvzQ1v0
m=application 20000 DTLS/SCTP 5000
c=IN IP4 192.0.2.2
a=candidate:2299743422 1 udp 2113937151 192.0.2.2 20000
typ host generation 0
a=ice-ufrag:6sFvz2gdLkEwjZEr
a=ice-pwd:cOTZKZNVlO9RSGsEGM63JXT2
a=fingerprint:sha-256 6B:8B:F0:65:5F:78:E2:51:3B:AC:6F:F3:3F:46:1B:35
:DC:B8:5F:64:1A:24:C2:43:F0:A1:58:D0:A1:2C:19:08
a=setup:active
a=mid:data
a=fmtp:5000 protocol=webrtc-datachannel; streams=10
Appendix B. Change log
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Changes in draft-04:
o Filled in sections on createOffer and createAnswer.
o Added SDP examples.
o Fixed references.
Changes in draft-03:
o Added text describing relationship to W3C specification
Changes in draft-02:
o Converted from nroff
o Removed comparisons to old approaches abandoned by the working
group
o Removed stuff that has moved to W3C specification
o Align SDP handling with W3C draft
o Clarified section on forking.
Changes in draft-01:
o Added diagrams for architecture and state machine.
o Added sections on forking and rehydration.
o Clarified meaning of "pranswer" and "answer".
o Reworked how ICE restarts and media directions are controlled.
o Added list of parameters that can be changed in a description.
o Updated suggested API and examples to match latest thinking.
o Suggested API and examples have been moved to an appendix.
Changes in draft -00:
o Migrated from draft-uberti-rtcweb-jsep-02.
Authors' Addresses
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Justin Uberti
Google
747 6th Ave S
Kirkland, WA 98033
USA
Email: justin@uberti.name
Cullen Jennings
Cisco
170 West Tasman Drive
San Jose, CA 95134
USA
Email: fluffy@iii.ca
Uberti & Jennings Expires March 22, 2014 [Page 46]
