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RTCWeb Working Group G. Mandyam
Internet Draft Qualcomm Innovation Center
Intended status: Informational Vijay Suryavanshi
Expires: January 30, 2013 Qualcomm
July 30, 2012

RTCWeb Data Stream and RTP Synchronization
draft-mandyam-rtcweb-data-synch-00.txt


Abstract


The RTCWeb working group in the IETF is tasked with developing
standards that will ensure interoperability between web browsers
establishing rich communications sessions. This working group is
tasked with delivering the specifications necessary to establish
real-time transport sessions between browsers (e.g. those based on
real-time protocol, i.e. RTP). Moreover, the group is also tasked
with providing a means for application data streaming between
browsers (i.e. opaque data streaming). Much like RTP
synchronization sources (SSRC's) can be temporally synchronized,
there are use cases that require opaque data stream synchronization
with the real-time communications stream between browsers in an
RTCWeb session. This document provides some options for temporally
associating an opaque data stream with a voice/video stream as part
of RTCWeb communications.


Status of this Memo


This Internet-Draft is submitted in full conformance with the
provisions of BCP 78 and BCP 79. This document may not be modified,
and derivative works of it may not be created, and it may not be
published except as an Internet-Draft.


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This Internet-Draft will expire on January 30, 2013.


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Copyright Notice


Copyright (c) 2012 IETF Trust and the persons identified as the document authors. All rights reserved.


This document is subject to BCP 78 and the IETF Trust's Legal
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warranty as described in the Simplified BSD License.


Table of Contents


1. Introduction...................................................2
2. SCTP-Based Data Streaming......................................3
2.1. SCTP Multi-Channel Impacts................................4
3. Opaque Data Synchronization and In-band RTP Signaling..........5
4. Security Considerations........................................7
5. IANA Considerations............................................7
6. Conclusions....................................................7
7. References.....................................................7
7.1. Normative References......................................7
7.2. Informative References....................................8
8. Acknowledgments................................................8
1. Introduction
The RTCWeb effort seeks to define the necessary interoperability
specifications required for real-time peer-to-peer communications
sessions between browsers. These communications sessions normally
involve multimedia data transmission (audio, video, or both).
However, RTCWeb will also include the ability for web applications
to initiate data streaming sessions between browsers.


One of the recommended transports for audio or video in RTCWeb
sessions is real-time protocol (RTP) [I-D.-rtcweb-rtp-usage]. An
RCTWeb session can include one or more RTP streams, each stream
identified by an SSRC (synchronization source) included in the RTP
frame header.


There has been some concern about whether existing mechanisms in RTP
standards allow an RTP session endpoint to be able to render


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multiple SSRC's in a time-synchronized manner [I-D.-draftalvestrand-
rtcweb-msid]. As a result, several mechanisms have been
proposed that would allow an RTCWeb endpoint to definitively
determine which SSRC's are temporally synchronized and must be
rendered as such.


Assuming the problem of associating temporally-synchronized SSRC's
will be solved by one of the proposed mechanisms, there still can be
cases where an SSRC may have a temporal relationship with
application-generated data that would also be streamed as part of
the RTCWeb session. An example is video overlay based on web touch
events during a video telephony session. In this case, a web
application detects an animation over the video preview window
(based on the end user drawing an image using the device touch
surface), and is required to send such information to the RTCWeb
endpoint so that the animation can be rendered.


This document discusses three approaches to synchronization of data
streams, along with associated recommendations.


2. SCTP-Based Data Streaming
[I-D.-jesup-rtcweb-data-protocol] describes an approach that could
be adopted in RTCWeb for data streaming, leveraging the Stream
Control Transmission Protocol (SCTP), and [I-D.ietf-mmusic-sctp-sdp]
provides the necessary extensions to Session Description Protocol
(DSP) to describe an SCTP stream. SDP is the mechanism by which
multimedia sessions are described RTCWeb, usually as part of the
invite or call announce.


The m-line in the SDP message (as per [I-D.ietf-mmusic-sctp-sdp])
should include sufficient information to describe the SCTP session


(e.g. plain SCTP, SCTP over DTLS, etc.). For example, an SDP
message from an offerer at address xxx.xx.xx.xx using port yyyyy for
SCTP communication, then a possible SDP offer would include
m=application yyyyy SCTP *
c=IN IP4 xxx.xx.xx.xx


If there is an additional RTP-based media source sent by the offerer
that needs synchronization with the SCTP stream, the ideal case
would be to leverage existing SDP grouping mechanisms. The mid
attribute of RFC 5888 could potentially be leveraged:


c=IN IP4 xxx.xx.xx.xx


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a=group:LS 1 2
m=application yyyyy SCTP *
a=mid:1
m=video zzzzz RTP/AVP
a=mid:2


There are some issues with this approach, as highlighted in [I-D.draft-
alvestrand-rtcweb-msid] (e.g. multiple SSRC's in each RTP
stream). Nevertheless, SDP grouping can provide a sufficient
solution to synchronizing the SCTP stream to an RTP stream as long
as there is one SSRC per RTP stream. SDP grouping should also be
applicable in the case where multiple SSRC's are part of the offer
and are associated with a canonical name (CNAME), using the
attribute guidelines of RFC 5576 (e.g. "a=ssrc:<ssrc-id>
cname:<cname>" along with "a=mid:...").


2.1. SCTP Multi-Channel Impacts
[I-D.-jesup-rtcweb-data-protocol] provides an SCTP-encapsulated
control protocol for the RTCWeb data channel that takes advantage of
the multistreaming capabilities of SCTP. SCTP allows for individual
stream identifiers and associated sequence numbers for any given
data chunk. This allows for flow control on individual streams
within an SCTP session. Streams are also further identified by a
label attribute as defined in [I-D.-jesup-rtcweb-data-protocol] as
part of the logical channel request. Since the streams are dynamic,
to associate an SCTP stream at any given instant in time with an RTP
session is not straightforward. In addition, SCTP can be
multihomed, i.e. endpoints can be associated with more than one IP
address. Some of the current unresolved issues are:


a. Should the SDP attribute describing the data channel stream be
based on logical channel label or SCTP stream ID?
b. What is the required receiver behavior if the data channel stream
identifier provided in the SDP offer does not match with
information sent in-band? Note that a comparable issue also
exists for RTP streams using CNAME and SSRC.
In order to address these issues in a simpler manner, the following
guideline is proposed for RTCWeb: the SDP grouping mechanism should
not address individual streams within an SCTP session. In other
words, once a temporal relationship is established between an RTP
stream and an SCTP session, that relationship will apply to all
streams in the SCTP session.


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3. Opaque Data Synchronization and In-band RTP Signaling
Web application generated data may have a temporal relationship with
an RTP-based media stream, but if is relatively infrequent and
therefore requires much less throughput than the media stream itself
it could make more sense to multiplex the application-specific data
into the RTP stream. Sec. 5.3.1 of RFC 3550 describes the RFC
Header Extension mechanism, by which an application-specific payload
can be inserted into an existing RTP stream without affecting the
media flow.


In the RTP header, and extension bit X can be set. This indicates


the existence of an extension header.


01234567890123456789012345678901


+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+


| defined by profile | length |


+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+


| header extension |


| .... |


Figure 1 : RTP Extension Header


Referring to Figure 1, the value of 16-bit profile field in the
extension header is implementation specific. This field could be
used in place of the channel label in the SCTP-based data channel.
Otherwise, this field can be ignored by the receiver.


The signaling of the use of an extension header as the means of
opaque data transfer could be agreed upon by the two RTCWeb
endpoints by means of an offer/answer protocol like SDP. The outof-
band signaling channel can be used to indicate to the receiver to
create a data channel based on the RTP extension header. RFC 5576
can also be leveraged in this case using a new source-specific
attribute 'data':


a=ssrc:<ssrc-id> data


The SDP exchange is not strictly required, however. This is because
the SSRC of the RTP stream has already been negotiated, and the
extension header is in fact really part of the RTP media stream
data.


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Ideally, a message-based Data Channel API from the WebRTC
specification (see [W3C.WD-webrtc-20120530]) would be leveraged by
the web application in such a way that the underlying user agent
would multiplex application data onto an existing RTP stream using
the RTP extension header. Borrowing from the JSEP messaging flow [ID.-
rtcweb-jsep], the PeerConnection setup will proceed as normal
from the offerer perspective:


OffererJS->OffererUA: var pc = new PeerConnection(config, null);
OffererJS->OffererUA: pc.onicecandidate = onIceCandidate;
OffererJS->OffererUA: pc.addStream(stream);
OffererJS->OffererUA: var offer = pc.createOffer(null);
OffererJS->OffererUA: pc.setLocalDescription("offer", offer);


... Answerer creates PeerConnection and sends answer


AnswererUA->OffererUA: <media>


// Send opaque data from Offerer to Answerer


OffererJS->OffererUA: var chan = pc.createDataChannel(10);
// Numeric label means opaque data to be sent with extension
header


OffererJS->OffererUA: chan.send("Some Payload");


AnswererUA->OffererUA: <media> with extension header


AnswererUA->AnswererJS: pc.ondatachannel = function({...});
// Answerer creates DataChannel listener on existing
PeerConnection based upon firing of onDataChannel event


OffererUA->OffererJS: datachannellistener.onmessage({});


Note that in the approach above, the creation of a data channel with
a numeric label is what triggers the OffererUA to use the extension
header. The numeric label can be directly sent as part of the
profile field in the extension header (provided that the numeric
label does not exceed 16 bits) The initial receipt of RTP data with
an extension header triggers the onDataChannel event to fire from
the AnswererUA.


3.1. Other Uses of the RTP Extension Header
Section 5.2 of [I-D.-rtcweb-rtp-usage] clearly discusses (but does
not call for requiring) additional uses of the RTP extension header.
These uses include a rapid synchronization feature (which allows
timing metadata to be inserted into the RTP stream), client-to-mixer
audio level, and mixer-to-client audio level. The profile space


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that may be consumed by these uses of the header extension can be
avoided for RTCWeb logical data channels that also use the header
extension.


4. Security Considerations
TBD.


5. IANA Considerations
TBD.


6. Conclusions
The ability to send and receive opaque data streams that are
syncronized to existing RTP media sessions will greatly enhance
RTCWeb. It will open up a several new possibilities for user
interactions around telephony sessions (video or voice). The
existing specifications in both the W3C and IETF do not address how
such a feature would be implemented. This document provided two
methods for achieving this feature that leveraged as much as
possible the specifications currently under consideration in RTCWeb
and the W3C.


7. References
7.1. Normative References
[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
Requirement Levels", BCP 14, RFC 2119, March 1997.


[RFC2234] Crocker, D. and Overell, P.(Editors), "Augmented BNF for
Syntax Specifications: ABNF", RFC 2234, Internet Mail
Consortium and Demon Internet Ltd., November 1997.


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


[RFC5576] Lennox, J., Ott, J., and T. Schierl, "Source-Specific
Media Attributes in the Session Description Protocol
(SDP)", RFC 5576, June 2009.


[RFC5888] Camarillo, G. and H. Schulzrinne, "The Session
DescriptionProtocol (SDP) Grouping Framework", RFC 5888,
June 2010.


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[RFC6222] Begen, A.,Perkins, C. and D. Wing, "Guidelines for
Choosing RTP Control Protocol (RTCP) Canonical Names", RFC
6222, April 2011.


[RFC5576] Lennox, J., Ott, J., and T. Schierl, "Source-Specific
Media Attributes in the Session Description Protocol
(SDP)", RFC 5576, June 2009.


[RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V.
Jacobson, "RTP: A Transport Protocol for Real Time
Communications", RFC 3550, July 2003.


[I-D.-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-03, June 2012.


[W3C.WD-webrtc-20120530] Bergkvist, A., Burnett, D., Narayanan, A.,
and C. Jennings, "WebRTC 1.0: Real-time Communication
Between Browsers", World Wide Web Consortium WD WD-webrtc20120209,
Editor's Draft, 30 May 2012.


[I-D.-rtcweb-jsep] Uberti, J. and C. Jennings, "Javascript Session
Establishment Protocol", draft-ietf-rtcweb-jsep-01, June
2012.


7.2. Informative References
[I-D.-jesup-rtcweb-data-protocol] Jesup, R., Loreto, S. and M.
Tuexen, "WebRTC Data Channel Protocol", draft-jesuprtcweb-
data-protocol-01, June 2012.


[I-D.-draft-alvestrand-rtcweb-msid] Alverstand, H., "Cross Session
Stream Identification in the Session Description
Protocol", draft-alvestrand-rtcweb-msid-02, May 2012.


[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-ietfmmusic-
sctp-sdp-01(work in progress), March 2012.


8. Acknowledgments
This document was prepared using 2-Word-v2.0.template.dot.


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Authors' Addresses


Giridhar Mandyam
Qualcomm Innovation Center
5775 Morehouse Drive
San Diego, CA 92121
USA


Email: mandyam@quicinc.com


Vijay Suryavanshi
Qualcomm Inc.
5775 Morehouse Drive
San Diego, CA 92121
USA


Email: vsuryava@qualcomm.com


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