Internet DRAFT - draft-ivov-rtcweb-noplan
draft-ivov-rtcweb-noplan
Network Working Group E. Ivov
Internet-Draft Jitsi
Intended status: Standards Track E. Marocco
Expires: December 19, 2013 Telecom Italia
P. Thatcher
Google
June 17, 2013
No Plan: Economical Use of the Offer/Answer Model in WebRTC Sessions
with Multiple Media Sources
draft-ivov-rtcweb-noplan-01
Abstract
This document describes a model for the lightweight use of SDP Offer/
Answer in WebRTC. The goal is to minimize reliance on Offer/Answer
exchanges in a WebRTC session and provide applications with the tools
necessary to implement the signalling that they may need in a way
that best fits their custom requirements and topologies. This
simplifies signalling of multiple media sources or providing RTP
Synchronisation source (SSRC) identification in multi-party sessions.
Another important goal of this model is to remove from clients
topological constraints such as the requirement to know in advance
all SSRC identifiers that they could potentially introduce in a
particular session.
The model described here is similar to the one employed by the data
channel JavaScript APIs in WebRTC, where methods are supported on
PeerConnection without being reflected in SDP.
This document does not question the use of SDP and the Offer/Answer
model or the value they have in terms of interoperability with legacy
or other non-WebRTC devices.
Status of This Memo
This Internet-Draft is submitted in full conformance with the
provisions of BCP 78 and BCP 79.
Internet-Drafts are working documents of the Internet Engineering
Task Force (IETF). Note that other groups may also distribute
working documents as Internet-Drafts. The list of current Internet-
Drafts is at http://datatracker.ietf.org/drafts/current/.
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Internet-Drafts are draft documents valid for a maximum of six months
and may be updated, replaced, or obsoleted by other documents at any
time. It is inappropriate to use Internet-Drafts as reference
material or to cite them other than as "work in progress."
This Internet-Draft will expire on December 19, 2013.
Copyright Notice
Copyright (c) 2013 IETF Trust and the persons identified as the
document authors. All rights reserved.
This document is subject to BCP 78 and the IETF Trust's Legal
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the Trust Legal Provisions and are provided without warranty as
described in the Simplified BSD License.
Table of Contents
1. Background . . . . . . . . . . . . . . . . . . . . . . . . . 2
2. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 4
3. Reliance on Offer/Answer . . . . . . . . . . . . . . . . . . 5
3.1. Interoperability with Legacy . . . . . . . . . . . . . . 6
4. Additional Session Control and Signalling . . . . . . . . . . 8
5. Demultiplexing and Identifying Streams
(Use of Bundle) . . . . . . . . . . . . . . . . . . . . . . . 9
6. Simulcasting, FEC, Layering and RTX (Open Issue) . . . . . . 10
7. WebRTC API Requirements . . . . . . . . . . . . . . . . . . . 11
7.1. Suggested WebRTC API Using TrackSendParams . . . . . . . 12
7.1.1. Example 2 . . . . . . . . . . . . . . . . . . . . . . 15
8. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 18
9. Informative References . . . . . . . . . . . . . . . . . . . 18
Appendix A. Acknowledgements . . . . . . . . . . . . . . . . . . 19
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 20
1. Background
In its early stages the RTCWEB working group chose to use the Session
Description Protocol (SDP) and the Offer/Answer model [RFC3264] when
establishing and negotiating sessions. This choice was also
accompanied by the decision not to mandate a specific signalling
protocol so that, once interoperability has been achieved, web
applications can choose the semantics that best fit their
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requirements. In some scenarios however, such as those involving the
use of multiple media sources, these choices have left open the issue
of exactly which operations should be handled by SDP Offer/Answer and
which of them should be left to application-specific signalling.
At the time of writing of this document, the RTCWEB working group is
considering two approaches to addressing the issue, that are often
referred to as Plan A [PlanA] and Plan B [PlanB]. Both of them
describe semantics that require Offer/Answer exchanges in a number of
situations where this could be avoided, particularly when adding or
removing media sources to a session. This requirement applies
equally to cases where a client adds the stream of a newly activated
web cam, a simulcast flow or upon the arrival or departure of a
conference participant.
Plan A handles such notifications with the addition or removal of
independent m= lines [PlanA], while Plan B relies on the use of
multiplexed m= lines but still depends on the Offer/Answer exchanges
for the addition or removal of media stream identifiers [MSID].
By taking the Offer/Answer approach, both Plan A and Plan B take away
from the application the opportunity to handle such events in a way
that is most fitting for the use case, which, among other things,
also goes against the working group's decision to not to define a
specific signalling protocol. (It could be argued that it is
therefore only natural how proponents of each plan, having different
use cases in mind, are remarkably far from reaching consensus).
Reliance on preliminary announcement of SSRC identifiers is another
issue. While this could be perceived as relatively straightforward
in one-to-one sessions or even conference calls within controlled
environments, it can be a problem in the following cases:
o interoperability with legacy/non-WebRTC endpoints
o use within non-controlled and potentially federated conference
environments where new RTP streams may appear relatively often.
In such cases the signalling required to describe all of them
through Offer/Answer may represent substantial overhead while none
or only a part of it (e.g. the description of a main, active
speaker stream) may be required by the application.
By increasing the number of Offer/Answer exchanges Both Plan A and
Plan B also increase the risk of encountering glare situations (i.e.
cases where both parties attempt to modify a session at the same
time). While glare is also possible with basic Offer/Answer and
resolution of such situations must be implemented anyway, the need to
frequently resort to such code may either negatively impact user
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experience (e.g. when "back off" resolution is used) or require
substantial modifications in the Offer/Answer model and/or further
venturing into the land of signalling protocols
[ROACH-GLARELESS-ADD].
2. Introduction
The goal of this document is to provide directions for use of the SDP
Offer/Answer model in a way that satisfies the following
requirements:
o the addition and removal of media sources (e.g. conference
participants, multiple web cams or "slides" ) must be possible
without the need of Offer/Answer exchanges;
o the addition or removal of simulcast or layered streams must be
possible without the need for Offer/Answer exchanges beyond the
initial declaration of such capabilities for either direction.
o call establishment must not require preliminary announcement or
even knowledge of all potentially participating media sources;
o application specific signalling should be used to cover most
semantics following call establishment, such as adding, removing
or identifying SSRCs;
o straightforward interoperability with widely deployed legacy
endpoints with rudimentary support for Offer/Answer. This
includes devices that allow for one audio and potentially one
video m= line and that expect to only ever be required to render a
single RTP stream at a time for any of them. (Note that this does
NOT include devices that expect to see multiple "m=video" lines
for different SSRCs as they can hardly be viewed as "widely
deployed legacy").
To achieve the above requirements this specification expects that
browsers and WebRTC endpoints in general will only use SDP Offer/
Answer to establish transport channels and initialize an RTP stack
and codec/processing chains. This also includes any renegotiation
that requires the re-initialisation of these chains. For example,
adding VP8 to a session that was setup with only H.264, would
obviously still require an Offer/Answer exchange.
All other session control and signalling are to be left to
applications.
The actual Offer/Answer semantics presented here do not differ
fundamentally from those proposed by Plan A and Plan B. The main
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differentiation point of this approach is the fact that the exact
protocol mechanism is left to WebRTC applications. Such applications
or lightweight signalling gateways can then implement either Plan A,
or Plan B, or an entirely different signalling protocol, depending on
what best matches their use cases and topology.
3. Reliance on Offer/Answer
The model presented in this specification relies on use of SDP and
Offer/Answer in quite the same way as many of the pre-WebRTC (and
most of the legacy) endpoints do: negotiating formats, establishing
transport channels and exchanging, in a declarative way, media and
transport parameters that are then used for the initialization of the
corresponding stacks.
The following is an example presenting what this specification views
as a typical offer sent by a WebRTC endpoint:
v=0
o=- 0 0 IN IP4 198.51.100.33
s=
t=0 0
a=group:BUNDLE audio video // declaring BUNDLE Support
c=IN IP4 198.51.100.33
a=ice-ufrag:Qq8o/jZwknkmXpIh // initializing ICE
a=ice-pwd:gTMACiJcZv1xdPrjfbTHL5qo
a=ice-options:trickle
a=fingerprint:sha-1 // DTLS-SRTP keying
a4:b1:97:ab:c7:12:9b:02:12:b8:47:45:df:d8:3a:97:54:08:3f:16
m=audio 5000 RTP/SAVPF 96 0 8
a=mid:audio
a=rtcp-mux
a=rtpmap:96 opus/48000/2 // PT mappings
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=extmap:1 urn:ietf:params:rtp-hdrext:csrc-audio-level //5825 header
a=extmap:2 urn:ietf:params:rtp-hdrext:ssrc-audio-level //extensions
[ICE Candidates]
m=video 5002 RTP/SAVPF 97 98
a=mid:video
a=rtcp-mux
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a=rtpmap:97 VP8/90000 // PT mappings and resolutions capabilities
a=imageattr:97 \
send [x=[480:16:800],y=[320:16:640],par=[1.2-1.3],q=0.6] \
[x=[176:8:208],y=[144:8:176],par=[1.2-1.3]] \
recv *
a=rtpmap:98 H264/90000
a=imageattr:98 send [x=800,y=640,sar=1.1,q=0.6] [x=480,y=320] \
recv [x=330,y=250]
a=extmap:3 urn:ietf:params:rtp-hdrext:fec-source-ssrc //5825 header
a=extmap:4 urn:ietf:params:rtp-hdrext:rtx-source-ssrc //extensions
a=max-send-ssrc:{*:1} // declaring maximum
a=max-recv-ssrc:{*:4} // number of SSRCs
[ICE Candidates]
The answer to the offer above would have roughly the same structure
and content. The most important aspects here are:
o Preserves interoperability with most kinds of legacy or non-WebRTC
endpoints.
o Allows the negotiation of most parameters that concern the media/
RTP stack (typically the browser).
o Only a single Offer/Answer exchange is required for session
establishment and, in most cases, for the entire duraftion of a
session.
o Leaves complete freedom to applications as to the way that they
are going to signal any other information such as SSRC
identification information or the addition or removal of RTP
streams.
3.1. Interoperability with Legacy
Interoperating with the "widely deployed legacy endpoints" is one of
the main reasons for the RTCWEB working group to choose the SDP Offer
/Answer model as basis for media negotiation. It is hence important
to clarify the compatibility claims that this specification makes.
A "widely deployed legacy endpoint" is considered to have the
following characteristics:
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o Likely to use the SIP protocol.
o Capability to gracefully handle one audio and potentially one
video m= line in an SDP Offer.
o Capability to render one SSRC per m=line at any given moment but
multiple, consecutive SSRCs over a period of time. This would be
the case with transferred session replacements for example. While
the capability to handle multiple SSRCs simultaneously is not
uncommon it cannot be relied upon and should first be confirmed by
signalling.
o Possibly have features such as ICE, BUNDLE, RTCP-MUX, etc. Just
as likely not to.
o Very unlikely to announce in SDP the SSRCs that they intend to use
for a given session.
o Exact set of features and capabilities: Guaranteed to be wildly
and widely diverse.
While it is relatively simple for RTCWEB to accommodate some of the
above, it is obviously impossible to design a model that could simply
be labeled as "compatible with legacy". It is reasonable to assume
that use cases involving use of such endpoints will be designed for a
relatively specific set of devices and applications. The role of the
WebRTC framework is to hence provide a least-common-denominator model
that can then be extended by applications.
It is just as important not to make choices or assumptions that will
render interoperability for some applications or topologies difficult
or even impossible.
This is exactly what the use of Offer/Answer discussed here strives
to achieve. Audio/Video offers originating from WebRTC endpoints
will always have a maximum of one audio and one video m= line. It
will be up to applications to determine exactly how many streams they
can afford to send once such a session has been established. The
exact mechanism to do this is outside the scope of this document (or
WebRTC in general).
Note that it is still possible for WebRTC endpoints to indicate
support for a maximum number of incoming or outgoing streams for
reasons such as processing constraints. Use of the "max-send-ssrc"
and "max-recv-ssrc" attributes [MAX-SSRC] could be one way of doing
this, although that mechanism would need to be extended to provide
ways of distinguishing between independent flows and complementary
ones such as layered FEC and RTX. Even with this in mind it is still
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important, not to rely on the presence of that indication in incoming
descriptions as well as to provide applications with a way of
retrieving such capabilities from the WebRTC stack (e.g. the
browser).
Determining whether a peer has the ability to seamlessly switch from
one SSRC to another is also left to application specific signalling.
It is worth noting that protocols such as SIP for example, often
accompany SSRC replacements with extra signalling (re-INVITEs with a
"replaces" header) that can easily be reused by applications or
mapped to something that they deem more convenient.
For the sake of interoperability this specification strongly advises
against the use of multiple m= lines for a single media type. Not
only would such use be meaningless to a large number of legacy
endpoints but it is also likely to be mishandled by many of them and
to cause unexpected behaviour.
Finally, it is also worth pointing out that there is a significant
number of feature rich non-WebRTC applications and devices that have
relatively advanced, modern sets of capabilities. Such endpoints
hardly fit the "legacy" qualification. Yet, as is often the case
with novel and/or proprietary applications, they too have adopted
diverse signalling mechanisms and the requirements described in this
section fully apply when it comes to interoperating with them.
4. Additional Session Control and Signalling
o Adding and removing RTP streams to an existing session.
o Accepting and refusing some of them.
o Identifying SSRCs and obtaining additional metadata for them (e.g.
the user corresponding to a specific SSRC).
All of the above semantics are best handled and hence should be left
to applications. There are numerous existing or emerging solutions,
some of them developed by the IETF, that already cover this. This
includes CLUE channels [CLUE], the SIP Event Package For Conference
State [RFC4575] and its XMPP variant [COIN] as well as the protocols
defined within the Centralised Conferencing IETF working group [XCON]
. Additional mechanisms, undoubtedly many based on JSON, are very
likely to emerge in the future as WebRTC applications address varying
use cases, scenarios and topologies.
The most important part of this specification is hence to prevent
certain assumptions or topologies from being imposed on applications.
One example of this is the need to know and include in the Offer/
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Answer exchange, all the SSRCs that can show up in a session. This
can be particularly problematic for scenarios that involve non-WebRTC
endpoints.
Large scale conference calls, potentially federated through RTP
translator-like bridges, would be another problematic scenario.
Being able to always pre-announce SSRCs in such situations could of
course be made to work but it would come at a price. It would either
require a very high number of Offer/Answer updates that propagate the
information through the entire topology, or use of tricks such as
pre-allocating a range of "fake" SSRCs, announcing them to
participants and then overwriting the actual SSRCs with them.
Depending on the scenario both options could prove inappropriate or
inefficient while some applications may not even need such
information. Others could be retrieving it through simplistic means
such as access to a centralized resource (e.g. an URL pointing to a
JSON description of the conference).
5. Demultiplexing and Identifying Streams (Use of Bundle)
This document assumes use of BUNDLE in WebRTC endpoints. This
implies that all RTP streams are likely to end up being received on
the same port. A demuxing mechanism is therefore necessary in order
for these packets to then be fed into the appropriate processing
chain (i.e. matched to an m= line).
Note: it is important to distinguish between the demultiplexing
and the identification of incoming flows. Throughout this
specification the former is used to refer to the process of
choosing selecting a depacketizing/decoding/processing chain to
feed incoming packets to. Such decisions depend solely on the
format that is used to encode the content of incoming packets.
The above is not to be confused with the process of making
rendering decision about a processed flow. Such decisions include
showing a "current speaker" flow at a specific location, window or
video tag, while choosing a different one for a second, "slides"
flow. Another example would be the possibility to attach "Alice",
"Bob" and "Carol" labels on top of the appropriate UI components.
This specification leaves such rendering choices entirely to
application-specific signalling as described in Section 4.
This specification uses demuxing based on RTP payload types. When
creating offers and answers WebRTC applications MUST therefore
allocate RTP payload types only once per bundle group. In cases
where rtcp-mux is in use this would mean a maximum of 96 payload
types per bundle [RFC5761]. It has been pointed out that some legacy
devices may have unpredictable behaviour with payload types that are
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outside the 96-127 range reserved by [RFC3551] for dynamic use. Some
applications or implementations may therefore choose not to use
values outside this range. Whatever the reason, offerers that find
they need more than the available payload type numbers, will simply
need to either use a second bundle group or not use BUNDLE at all
(which in the case of a single audio and a single video m= line
amounts to roughly the same thing). This would also imply building a
dynamic table, mapping SSRCs to PTs and m= lines, in order to then
also allow for RTCP demuxing.
While not desirable, the implications of such a decision would be
relatively limited. Use of trickle ICE [TRICKLE-ICE] is going to
lessen the impact on call establishment latency. Also, the fact that
this would only occur in a limited number of cases makes it unlikely
to have a significant effect on port consumption.
An additional requirement that has been expressed toward demuxing is
the ability to assign incoming packets with the same payload type to
different processing chains depending on their SSRCs. A possible
example for this is a scenario where two video streams are being
rendered on different video screens that each have their own decoding
hardware.
While the above may appear as a demuxing and a decoding related
problem it is really mostly a rendering policy specific to an
application. As such it should be handled by app. specific
signalling that could involve custom-formatted, per-SSRC information
that accompanies SDP offers and answers.
6. Simulcasting, FEC, Layering and RTX (Open Issue)
From a WebRTC perspective, repair flows such as layering, FEC, RTX
and to some extent simulcasting, present an interesting challenge,
which is why they are considered an open issue by this specification.
On the one hand they are transport utilities that need to be
understood, supported and used by browsers in a way that is mostly
transparent to applications. On the other, some applications may
need to be made aware of them and given the option to control their
use. This could be necessary in cases where their use needs to be
signalled to non-WebRTC endpoints in an application specific way.
Another example is the possibility for an application to choose to
disable some or all repair flows because it has been made aware by
application-specific signalling that they are temporarily not being
used/rendered by the remote end (e.g. because it is only displaying a
thumbnail or because a corresponding video tag is not currently
visible).
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One way of handling such flows would be to advertise them in the way
suggested by [RFC5956] and to then control them through application
specific signalling. This options has the merit of already existing
but it also implies the pre-announcement and propagation of SSRCs and
the bloated signalling that this incurs. Also, relying solely on
Offer/Answer here would expose an offerer to the typical race
condition of repair SSRCs arriving before the answer and the
processing ambiguity that this would imply.
Another approach could be a combination of RTCP and RTP header
extensions [RFC5285] in a way similar to the one employed by the
Rapid Synchronisation of RTP Flows [RFC6051]. While such a mechanism
is not currently defined by the IETF, specifying it could be
relatively straightforward:
Every packet belonging to a repair flow could carry an RTP header
extension [RFC5285] that points to the source stream (or source layer
in case of layered mechanisms).
Again, these are just some possibilities. Different mechanisms may
and probably will require different extensions or signalling
([SRCNAME] will likely be an option for some). In some cases, where
layering information is provided by the codec, an extensions is not
going to be necessary at all.
In cases where FEC or simulcast relations are not immediately needed
by the recipient, this information could also be delayed until the
reception of the first RTCP packet.
7. WebRTC API Requirements
One of the main characteristics of this specification is the use of
SDP for transport channel setup and media stack initialisation only.
In order for applications to be able to cover everything else it is
important that WebRTC APIs actually allow for it. Given the initial
directions taken by early implementations and specification work,
this is currently almost but not entirely possible.
The following is a list of requirements that the WebRTC APIs would
need to satisfy in order for this specification to be usable. (Note:
some of the items are already possible and are only included for the
sake of completeness.)
1. Expose the SSRCs of all local MediaStreamTrack-s that the
application attaches to a PeerConnection.
2. Expose the SSRCs of all remote MediaStreamTrack-s that are
received on a PeerConnection
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3. Expose to applications all locally generated repair flows that
exist for a source (e.g. FEC and RTX flows that will be generated
for a webcam) their types relations and SSRCs.
4. Expose information about the maximum number of incoming streams
that can be decoded and rendered.
5. Applications should be able to pause and resume (disable and
enable) any MediaStreamTrack. This should also include the
possibility to do so for specific repair flows.
6. Information about how certain MediaStreamTrack-s relate to each
other (e.g. a given audio flow is related to a specific video
flow) may be exchanged by applications after media has started
arriving. At that point the corresponding MediaStreamTrack-s may
have been announced to the application within independent
MediaStream-s. It should therefore be possible for applications
to join such tracks within a single MediaStream.
The following section Section 7.1 provides suggestions for addressing
the above requirements.
7.1. Suggested WebRTC API Using TrackSendParams
This document proposes that the following methods and dictionaries be
added to the WebRTC API. The changes follow the model of
createDataChannel, which has a JS method on PeerConnection that makes
it possible to add data channels without going through SDP.
Furthermore, just like createDataChannel allows 2 ways to handle
neogitation (the "I know what I'm doing; Here's what I want to send;
Let me signal everything" mode and the "please take care of it for
me; send an OPEN message" mode), this also has 2 ways to handle
negotiation (the "I know what I'm doing; Here's what I want to send;
Let me signal everything" mode and the "please take care of it for
me; send SDP back and forth" mode).
Following the success of createDataChannel, this allows simple
applications to Just Work and more advanced applications to easily
control what they need to. In particular, it's possible to use this
API to implement either Plan A or Plan B.
// The following two method are added to RTCPeerConnection
partial interface RTCPeerConnection {
// Create a stream that is used to send a source stream.
// The MediaSendStream.description can be used for signalling.
// No media is sent until addStream(MediaSendStream) is called.
LocalMediaStream createLocalStream(MediaStream sourceStream);
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// Create a stream that is used to receive media from the remote side,
// given the parameters signalled from MedaiSendStream.description.
MediaStream createRemoteStream(MediaStreamDescription description);
}
interface LocalMediaStream implements MediaStream {
// This can be changed at any time, but especially before calling
// PeerConnection.addStream
attribute MediaStreamDescription description;
}
// Represents the parameters used to either send or receive a stream
// over a PeerConnection.
dictionary MediaStreamDescription {
MediaStreamTrackDescription[] tracks;
}
// Represents the parameters used to either send or receive a track over
// a PeerConnection. A track has many "flows", which can be grouped
// together.
dictionary MediaStreamTrackDescription {
// Same as the MediaStreamTrack.id
DOMString id;
// Same as the MediaStreamTrack.kind
DOMString kind;
// A track can have many "flows", such as for Simulcast, FEC, etc.
// And they can be grouped in arbitrary ways.
MediaFlowDescription[] flows;
MediaFlowGroup[] flowGroups;
}
// Represents the parameters used to either send or receive a "flow"
// over a PeerConnection. A "flow" is a media that arrives with a
// single, unique SSRC. One to many flows together make up the media
// for a track. For example, there may be Simulcast, FEC, and RTX
// flows.
dictionay MediaFlowDescription {
// The "flow id" must be unique to the track, but need not be unique
// outside of the track (two tracks could both have a flow with the
// same flow ID).
DOMString id;
// Each flow can go over its own transport. If the JS sets this to a
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// transportId that doesn't have a transport setup already, the
// browser will use SDP negotiation to setup a transport to back that
// transportId. If This is set to an MID in the SDP, then that MID's
// transport is used.
DOMString transportId;
// The SSRC used to send the flow.
unsigned int ssrc;
// When used as receive parameters, this indicates the possible list
// of codecs that might come in for this flow. For exmample, a given
// receive flow could be setup to receive any of OPUS, ISAC, or PCMU.
// When used as send parameters, this indicates that the first codec
// should be used, but the browser can use send other codecs if it
// needs to because of either bandwidth or CPU constraints.
MediaCodecDescription[] codecs;
}
dictionary MediaFlowGroup {
DOMString type; // "SIM" for Simulcast, "FEC" for FEC, etc
DOMString[] flowids;
}
dictionary MediaCodecDescription {
unsigned byte payloadType;
DOMString name;
unsigned int? clockRate;
unsigned int? bitRate;
// A grab bag of other fmtp that will need to be further defined.
MediaCodecParam[] params;
}
dictionary MediaCodecParam {
DOMString key;
DOMString value;
}
}
Some additional notes:
o When LocalMediaStreams are added using addStream,
onnegotiatedneeded is not called, and those streams are never
reflected in future SDP exchanges. Indeed, it would be impossible
to put them in the SDP without first resolving if that would be
Plan A SDP or Plan B SDP.
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o Just like piles of attributes would need to be defined for Plan A
and for Plan B, similar attributes would need to be defined here
(Luckily, much work has already been done figuring out what those
parameters are :).
API Pros:
o Either Plan A or Plan B or could be implemented in Javascript
using this API
o It exposes all the same functionality to the Javascript as SDP,
but in a much nicer format that is much easier to work with.
o Any other signalling mechanism, such as Jingle or CLUE could be
implemented using this API.
o There is almost no risk of signalling glare.
o Debugging errors with misconfigured descriptions should be much
easier with this than with large SDP blobs.
API Cons:
o Now there are two slightly different ways to add streams: by
creating a LocalMediaStream first, and not. This is, however,
analogous to setting "negotiated: true" in createDataChannel. On
way is "Just Work", and the other is more advanced control.
o All the options in MediaCodecDescription are a bit complicated.
Really, this is only necessary because Plan A requires being able
to specify codec parameters per SSRC, and set each flow on
different transports. If we did not have this requirement, we
could simplify.
7.1.1. Example 2
Following is an example of how these API additions would be used:
// Imagine I have MyApp, handles creating a PeerConnection,
// signalling, and rendering streams. This is how the new API could be
// used.
var peerConnection = MyApp.createPeerConnection();
// On sender side:
var stream = MyApp.getMediaStream();
var localStream = peerConnection.createSendStream(stream);
sendStream.description = MyApp.modifyStream(localStream.description)
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MyApp.signalAddStream(localStream.description, function(response)) {
if (!response.rejected) {
// Media will not be sent.
peerConnection.addStream(localStream);
}
}
// On receiver side:
MyApp.onAddStreamSignalled = function(streamDescription) {
var stream = peerConnection.createReceiveStream(streamDescription);
MyApp.renderStream(stream);
}
// In this exchange, the MediaStreamDescription signalled from the
// sender to the receiver may have looked something like this:
{
tracks: [
{
id: "audio1",
kind: "audio",
flows: [
{
id: "main",
transportId: "transport1",
ssrc: 1111,
codecs: [
{
payloadType: 111,
name: "opus",
// ... more codec details
},
{
payloadType: 112,
name: "pcmu",
// ... more codec details
}]
}]
},
{
id: "video1",
kind: "video",
flows: [
{
id: "sim0",
transportId: "transport2",
ssrc: 2222,
codecs: [
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{
payloadType: 122,
name: "vp8"
// ... more codec details
}]
},
{
id: "sim1",
transportId: "transport2",
ssrc: 2223,
codecs: [
{
payloadType: 122,
name: "vp8",
// ... more codec details
}]
},
{
id: "sim2",
transportId: "transport2",
ssrc: 2224,
codecs: [
{
payloadType: 122,
name: "vp8",
// ... more codec details
}]
},
{
id: "sim0fec",
transportId: "transport2",
ssrc: 2225,
codecs: [
{
payloadType: 122,
name: "vp8",
// ...
}]
}],
flowGroups: [
{
semantics: "SIM",
ssrcs: [2222, 2223, 2224]
},
{
semantics: "FEC",
ssrcs: [2222, 2225]
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}]
}]
}
8. IANA Considerations
None.
9. Informative References
[CLUE] Duckworth, M., Pepperell, A., and S. Wenger, "Framework
for Telepresence Multi-Streams", reference.I-D.ietf-clue-
framework (work in progress), May 2013, <reference.I-D
.ietf-clue-framework>.
[COIN] Ivov, E. and E. Marocco, "XEP-0298: Delivering Conference
Information to Jingle Participants (Coin)", XSF XEP 0298,
June 2011, <reference.I-D.ietf-coin-framework>.
[MAX-SSRC]
Westerlund, M., Burman, B., and F. Jansson, "Multiple
Synchronization sources (SSRC) in RTP Session Signaling ",
reference.I-D.westerlund-avtcore-max-ssrc (work in
progress), July 2012, <reference.I-D.westerlund-avtcore-
max-ssrc>.
[MSID] Alvestrand, H., "Cross Session Stream Identification in
the Session Description Protocol", reference.I-D.ietf-
mmusic-msid (work in progress), February 2013,
<reference.I-D.ietf-mmusic-msid>.
[PlanA] Roach, A. and M. Thomson, "Using SDP with Large Numbers of
Media Flows", reference.I-D.roach-rtcweb-plan-a (work in
progress), May 2013, <reference.I-D.roach-rtcweb-plan-a>.
[PlanB] Uberti, J., "Plan B: a proposal for signaling multiple
media sources in WebRTC.", reference.I-D.uberti-rtcweb-
plan (work in progress), May 2013, <reference.I-D.uberti-
rtcweb-plan>.
[RFC3264] Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model
with Session Description Protocol (SDP)", RFC 3264, June
2002.
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[RFC3551] Schulzrinne, H. and S. Casner, "RTP Profile for Audio and
Video Conferences with Minimal Control", STD 65, RFC 3551,
July 2003.
[RFC4575] Rosenberg, J., Schulzrinne, H., and O. Levin, "A Session
Initiation Protocol (SIP) Event Package for Conference
State", RFC 4575, August 2006.
[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.
[RFC5956] Begen, A., "Forward Error Correction Grouping Semantics in
the Session Description Protocol", RFC 5956, September
2010.
[RFC6015] Begen, A., "RTP Payload Format for 1-D Interleaved Parity
Forward Error Correction (FEC)", RFC 6015, October 2010.
[RFC6051] Perkins, C. and T. Schierl, "Rapid Synchronisation of RTP
Flows", RFC 6051, November 2010.
[ROACH-GLARELESS-ADD]
Roach, A., "An Approach for Adding RTCWEB Media Streams
without Glare", reference.I-D.roach-rtcweb-glareless-add
(work in progress), May 2013, <reference.I-D.roach-rtcweb-
glareless-add>.
[SRCNAME] Westerlund, M., Burman, B., and P. Sandgren, "RTCP SDES
Item SRCNAME to Label Individual Sources ", reference.I-D
.westerlund-avtext-rtcp-sdes-srcname (work in progress),
October 2012, <reference.I-D.westerlund-avtext-rtcp-sdes-
srcname>.
[TRICKLE-ICE]
Ivov, E., Rescorla, E., and J. Uberti, "Trickle ICE:
Incremental Provisioning of Candidates for the Interactive
Connectivity Establishment (ICE) Protocol ", reference.I-D
.ivov-mmusic-trickle-ice (work in progress), March 2013,
<reference.I-D.ivov-mmusic-trickle-ice>.
[XCON] , "Centralized Conferencing (XCON) Status Pages", ,
<http://tools.ietf.org/wg/xcon/>.
Appendix A. Acknowledgements
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Many thanks to Bernard Aboba and Mary Barnes, for reviewing this
document and providing numerous comments and substantial input.
Authors' Addresses
Emil Ivov
Jitsi
Strasbourg 67000
France
Phone: +33-177-624-330
Email: emcho@jitsi.org
Enrico Marocco
Telecom Italia
Via G. Reiss Romoli, 274
Turin 10148
Italy
Email: enrico.marocco@telecomitalia.it
Peter Thatcher
Google
747 6th St S
Kirkland, WA 98033
USA
Phone: +1 857 288 8888
Email: pthatcher@google.com
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