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Versions: 00                                                            
RTCWEB                                                      J. Rosenberg
Internet-Draft                                                     Skype
Intended status: Informational                               C. Jennings
Expires: January 5, 2012                                           Cisco
                                                             J. Peterson
                                                                 Neustar
                                                              M. Kaufman
                                                                   Skype
                                                             E. Rescorla
                                                                    RTFM
                                                           T. Terriberry
                                                                 Mozilla
                                                            July 4, 2011


 Multiplexing of Real-Time Transport Protocol (RTP) Traffic for Browser
                  based Real-Time Communications (RTC)
                    draft-rosenberg-rtcweb-rtpmux-00

Abstract

   This document argues that multiplexing of voice and video traffic
   over a single RTP session should be specified as the baseline mode of
   operation for multimedia traffic in RTC web.

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/.

   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 January 5, 2012.

Copyright Notice

   Copyright (c) 2011 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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   Provisions Relating to IETF Documents
   (http://trustee.ietf.org/license-info) in effect on the date of
   publication of this document.  Please review these documents
   carefully, as they describe your rights and restrictions with respect
   to this document.  Code Components extracted from this document must
   include Simplified BSD License text as described in Section 4.e of
   the Trust Legal Provisions and are provided without warranty as
   described in the Simplified BSD License.


Table of Contents

   1.  Introduction . . . . . . . . . . . . . . . . . . . . . . . . .  3
   2.  RTP Muxing with SSRC . . . . . . . . . . . . . . . . . . . . .  3
   3.  Arguments in Favor of Multiplexing . . . . . . . . . . . . . .  4
     3.1.  NAT Resource Preservation  . . . . . . . . . . . . . . . .  4
     3.2.  Improved Failure Modes . . . . . . . . . . . . . . . . . .  5
     3.3.  Setup Time . . . . . . . . . . . . . . . . . . . . . . . .  5
     3.4.  Complexity . . . . . . . . . . . . . . . . . . . . . . . .  5
   4.  Responding to draft-perkins-rtcweb-rtp-usage . . . . . . . . .  5
     4.1.  Requires Additional Signaling  . . . . . . . . . . . . . .  6
     4.2.  QoS and Traffic Engineering  . . . . . . . . . . . . . . .  6
     4.3.  Scalability  . . . . . . . . . . . . . . . . . . . . . . .  7
     4.4.  RTP Retransmission . . . . . . . . . . . . . . . . . . . .  7
     4.5.  Forward Error Correction . . . . . . . . . . . . . . . . .  8
     4.6.  RTCP Issues  . . . . . . . . . . . . . . . . . . . . . . .  8
   5.  Arguing Against a Shim . . . . . . . . . . . . . . . . . . . .  9
   6.  Conclusion . . . . . . . . . . . . . . . . . . . . . . . . . . 10
   7.  Informative References . . . . . . . . . . . . . . . . . . . . 10
   Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 11





















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1.  Introduction

   The RTCweb working group is chartered to specify a framework and
   protocols for enabling real-time communications services within a
   browser, without the need for plugins
   [I-D.rosenberg-rtcweb-framework].  It is envisioned that this will
   enable many use cases [I-D.ietf-rtcweb-use-cases-and-requirements],
   the most basic of which is a video call between two users on the web.

   In order to enable this functionality, the specifications produced by
   the IETF will mandate a specific set of protocols that must be
   implemented within the browser.  It is anticipated that these
   protocols will include the Real-Time Transport Protocol [RFC3550],
   and either in full or in part, Interactive Connectivity Establishment
   (ICE) [RFC5245].

   The usage of RTP raises the question of multiplexing - whether or not
   RTCP and RTP should run on the same port, and furthermore, whether or
   not voice, video, and possibly data, should also run on the same
   port.  To provide guidance on this, Perkins et. al. produced
   [I-D.perkins-rtcweb-rtp-usage], which recommends that voice and video
   utilize different RTP sessions, and thus different UDP ports.

   This document argues against this conclusion, and advocates that a
   single transport session (i.e., a single UDP port) is used to carry
   voice and video traffic, using the SSRC for demux.


2.  RTP Muxing with SSRC

   This document recommends that all of the associated media content of
   the call - the voice, video, and RTCP traffic for both the voice and
   video sessions, utilize a single transport session (i.e., single UDP
   port).  In cases where there are multiple video streams (for example,
   screen sharing), the single transport session would carry all of the
   video.  Furthemore, that demultiplexing voice and video traffic is
   done by assigning a different SSRC to each.  This recommendation
   applies to the case of a single unicast communications session
   between a pair of endpoints (e.g., this document does not consider
   the case of running a multi-user service like a gateway).

   To enable multiplexing, we propose that the 32-bit SSRC value in the
   RTP header be broken up into the following sub-fields:


     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
     |          Magic Cookie         |Type |     StreamID          |x|
     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+



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                                SSRC Field

   The Magic Cookie is two bytes, with a value of 0xf7b3.  It is meant
   to facilitate DPI applications which can use its value to - with high
   confidence - determine that this RTP packet uses the encoding format
   defined here.  The type is a 3 bit value, corresponding to the top-
   level MIME type of the media (mapping table TBD).  It too is meant to
   facilitate DPI applications which want to separate voice and video.
   The streamID is a 12 bit field which represents the unique ID for
   this stream.  It is signaled between participants out of band.  The
   final bit, 'x' is set to zero and is reserved for future usage.


3.  Arguments in Favor of Multiplexing

   This section outlines several arguments in favor of multiplexing.

3.1.  NAT Resource Preservation

   Today's Internet is full of Network Address Translators (NAT), a
   situation which is likely to get worse as IPv4 address exhaustion
   continues.  When NAT is in use, the constraint on the number of
   endpoints behind the NAT is based on the number of parallel transport
   sessions that need to be supported.  If, for example, a NAT has a
   single external IP address, it can support 64k UDP sessions while
   having an endpoint-independent mapping behavior [RFC4787].  Thus, in
   the presence of NAT, parallel transport sessions becomes the scarce
   resource.

   If rtcweb specifies that audio and video run on a separate port, this
   will double the number of transport session resources consumed in
   intervening NATs.  While the usage of port as an application layer
   demux point made sense when RTP was designed back in 1992 (the year
   the first RTP draft was published), the Internet has changed
   substantially since then.  Continuing to perpetuate this design
   optimizes preseveration of legacy against protection of resources in
   the modern Internet.  We feel that this optimizes in the wrong
   direction.

   Given that we anticipate widespread usage of rtcweb, this design
   choice may create a non-trivial load on the transport session
   capacity of the Internet at large.  Real-time video communications on
   the Internet has seen huge growth in recent years.  For Skype,
   approximately 40% of its Skype-to-Skype calls are video based.  A
   recent report by Sandvine reports that Skype alone is the third
   largest source of upload traffic on the Internet as a whole, largely
   attributed to Skype video calling.  <http://www.sandvine.com/
   downloads/documents/05-17-2011_phenomena/



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   Sandvine%20Global%20Internet%20Phenomena%20Spotlight%20-%20Netflix%
   20Rising.pdf>.  The conclusion from this is that the costs of a
   separate voice and video port cannot be ignored.

   Simply put, the usage of transport ports for application
   demultiplexing should be considered harmful for the Internet.

3.2.  Improved Failure Modes

   The usage of separate transport sessions for the audio, video or
   other content of the call introduces a variety of partial failure
   modes.  The transport session for one type of media might get
   established; but a NAT capacity problem might cause the transport
   session for another type of media to fail.  Usage of a single
   transport session means that the conversation succeeds or fails
   atomically.  We consider this a feature.

3.3.  Setup Time

   The rtcweb group is considering the usage of ICE to create p2p
   sessions.  ICE provides firewall and NAT traversal in addition to
   providing a handshake necessary to assure mutual consent for
   communications.

   Unfortunately, ICE requires time to perform its setup operations.
   This time grows in proportion to the number of transport sessions
   which must be opened in order to support the call.  By using a
   different port for video traffic, call setup times will increase.
   The precise amount of this increase depends on the type of NAT and
   varies depending on packet loss.  However, in a simple, ideal case of
   no packet loss and direct connectivity between endpoints, this value
   is XXX [[fill in]].

3.4.  Complexity

   ICE is not a simple protocol.  One of its significant complexities is
   its requirement to support calls for multiple media streams, each of
   which runs on a separate port, and multiple components for each
   stream (e.g., RTCP).  If the concept of streams and components were
   eliminated, ICE would be a simpler protocol.

   If, within rtcweb, a single transport connection was utilized,
   browsers could implement a simplified version of the ICE protocol.


4.  Responding to draft-perkins-rtcweb-rtp-usage

   [I-D.perkins-rtcweb-rtp-usage] outlines several arguments for



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   continuing to use a separate port for audio and video.  In this
   section, we respond to those arguments.

4.1.  Requires Additional Signaling

   [I-D.perkins-rtcweb-rtp-usage] argues that multiplexing of voice and
   video on the same RTP session would require a demux point to be
   specified (for example, the SSRC), and require additional signaling
   to be specified to accomplish this.

   Firstly, this conclusion is only partly true.  For communications
   sessions between rtcweb users within the same domain, no signaling
   specifications are required.  This is true in general with rtcweb;
   one of its benefits is that it does not require standardized
   signaling.

   Secondly, it is not yet clear that rtcweb will be able to
   interoperate with existing VoIP endpoitns without a media
   intermediary to terminate ICE traffic.  It is our position that
   interoperability without media intermediary only be provided for
   basic voice services, and even then, only when RTCP is supported.  In
   the case of basic voice endpoints, where there is no video, RTP
   multiplexing of voice and video is irrelevant, and thus no signaling
   complexity is introduced.

   Thirdly, the primary place where there will be a need for signaling
   enhancements is for inter-domain calling between rtcweb endpoints in
   different domains.  In such a case, an SDP extension is required, and
   one can be specified.  It is trivial to do so.

   Finally, this document does recommend that it be possible to utilize
   a separate transport session for voice and for video, and that, in
   the worst case, this mode can be used for calls between an rtcweb
   endpoint and a legacy endpoint.

4.2.  QoS and Traffic Engineering

   [I-D.perkins-rtcweb-rtp-usage] argues that multiplexing of voice and
   video on the same RTP session would mean that it would not be
   possible to apply QoS techniques separately for voice and video which
   rely on the 5-tuple.

   Firstly, the public Internet lacks any QoS mechanism, so this
   argument is moot on the public Internet.

   Secondly, private enterprise networks which do provide QoS most often
   use diffserv.  Diffserv is compatible with utilization of a common
   port for voice and video traffic.  Typically, different DSCPs are



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   used for voice and video (Cisco recommends EF for audio and AF41 for
   video in enterprise telephony deployments), and this practice is
   compatible with usage of the same port - each packet would be marked
   appropriately.  It is also possible to use the same DSCP for voice
   and video.

   Carrier networks, such as mobile operator networks, typically provide
   QoS through traffic engineering, using a combination of MPLS tunnels
   and diffserv markings.  MPLS tunnels do use 5-tuples as classifiers
   to determine which traffic to put in what kind of tunnel.  If there
   is a need for using separate MPLS tunnels for voice and video, the
   DSCP codepoint itself can be used as a differentiator.

   It is true that it would not be possible to utilize RSVP to
   separately establish QoS treatment for the voice and the video
   traffic.  However, there is very little real deployment of RSVP.
   None within the public Internet and relatively little within
   corporate networks.  As such, this argument is mostly theoretical.

   Finally, DPI is used within some operator networks to perform traffic
   classification.  It would always be possible to use DPI to assign
   different treatment to voice and video traffic.

4.3.  Scalability

   [I-D.perkins-rtcweb-rtp-usage] argues that multiplexing of voice and
   video on the same RTP session would mean that layered coding using
   multicast for each layer would not be possible.

   Firstly, most layered coding today uses unicast and a switch or mixer
   of some sort to discard layers.  That architecture is completely
   compatible with the usage of a single transport session for voice and
   video.  The limitation applies only to the use of IP multicast for
   real-time communications.  The usage of multicast on the Internet has
   substantially diminished over time.  There is some usage today in
   private networks but primarily for streaming media distribution.  The
   usage for real-time communications is quite rare.  As such, we find
   this to be a theoretical corner case.

4.4.  RTP Retransmission

   [I-D.perkins-rtcweb-rtp-usage] argues that multiplexing of voice and
   video on the same RTP session would not be interoperable with
   endpoints doing RTP retransmission per [RFC4588].

   As pointed out above, interoperability with existing endpoints
   without the usage of a media intermediary is not a given at this
   point, and we argue it should only be supported for the common case -



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   a basic, voice-only RTP-capable endpoint.  There is, to our
   knowledge, relatively little deployment of RFC4588, at least for
   real-time communications.  It is certainly not a common feature in
   basic RTP endpoints and never a baseline requirement for
   interoperability.  Consequently, if there is a need to interoperate
   with an endpoint supporting RFC4588, and it is desired to avoid a
   media intermediary, RFC4588 can just be turned off for the session.

   As such, we find the interoperability argument here not compelling.

4.5.  Forward Error Correction

   [I-D.perkins-rtcweb-rtp-usage] argues that multiplexing of voice and
   video on the same RTP session will limit the applicability of FEC
   [RFC5109] to when the RTP packets are half of the path MTU.

   There are two cases to consider - interoperability with existing
   endpoints and usage for calls between rtcweb endpoints.

   For interoperability with existing endpoints, we argue the same thing
   here as for retransmits.  FEC is not commonly used in legacy voice
   endpoints, and if it is supported, is never a required feature.
   Consequently, if present, its usage can be disabled when
   interoperating with an rtcweb endpoint.  If FEC is included as part
   of the rtcweb specifications, the lower bandwidth of voice means that
   FEC packets could be sent on the same port, using [RFC2198], without
   approaching the path MTU.

   For communications between rtcweb endpoints, this is only an issue if
   FEC is included as part of the rtcweb specification.  If the group
   decides to do that (there is some value for real-time video), it
   should define a mechanism which allows for FEC packets to be sent
   using a separate SSRC.

4.6.  RTCP Issues

   [I-D.perkins-rtcweb-rtp-usage] argues that multiplexing of voice and
   video on the same RTP session will introduce complications in the
   usage of RTCP, primarily when considering RTCP extensions.

   It is our belief that normal RTCP operation as defined in the RTCP
   specification will work fine with multiplexed voice and video
   traffic.  SRs and RRs are already generated per SSRC to handle
   multiple senders, and RTCP in general supports feedback for multiple
   SSRC within a session.  These mechanisms work as defined when each
   SSRC happens to represent a different media stream instead of a
   different user.




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   The only complication that arises is for RTCP extensions which are
   defined to be media dependent.  [I-D.perkins-rtcweb-rtp-usage] points
   out, as an example, the usage of RTCP extended report blocks (XR)
   [RFC3611].  However, XR works fine in conjunction with multiplexing
   of voice and video within the same port.  Each of the seven report
   blocks defined in [RFC3611] include the SSRC of the source as part of
   the block, and thus will work.  [I-D.perkins-rtcweb-rtp-usage]
   indicates that "SSRC purpose tagging needs not only to be one the
   media side, but also on the RTCP reporting".  However, we do not
   believe this to be accurate.  Since the XR blocks report the SSRC
   source already, the specifications provide all that is needed.  The
   XR report is merely included when it is relevant.

   Furthermore, the discussion around XR assumes that we need to support
   them for interoperability with existing VoIP endpoints, or we are
   utilizing it for rtcweb itself.  As with FEC and retransmissions, in
   the case of interoperability, if there is an issue, XR can simply be
   disabled in these cases.  [RFC3611] does specify that XR can be sent
   without prior signaling.  In the worst case XR are received by an
   rtcweb endpoint which are then discarded.  In terms of usage of RTCP
   XR for communications between rtcweb endpoints, we would argue that a
   much more flexible solution would be to provide Javascript APis which
   allow the application to have access to the same data used to
   generate the XR, and then the application itself can use this data as
   it sees fit, including sending it back to the sender through some
   kind of application data packet.


5.  Arguing Against a Shim

   It has been proposed on the mailing list that an alternative approach
   for multiplexing on the same port would be to specify a new
   multiplexing protocol that has a small shim, which could then be used
   to separate voice and video traffic as a layer between UDP and RTP.
   Such a shim could then also be used to enable non-RTP data traffic as
   well.

   We believe that such a shim would be a mistake, for the same reason
   that shims have been avoided in the multiplexing of RTCP, STUN, and
   DTLS on the same port as RTP:

   o  The shim would break interoperability with a great deal of
      existing network inspection gear - firewalls, packet sniffers,
      traffic analyzers, and so on - which know how to extract, parse,
      and process RTP packets.

   o  The shim would add complexity through yet another layer of
      multiplexing.



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   o  The shim would increase packet overhead further.

   o  A shim is a mistake which cannot be undone later.  If multiplexing
      on a single port truly causes interoperability issues, clients can
      fall back to using multiple ports, possibly even in the
      preponderance of cases.  However, once a shim is inserted,
      interoperability will always require an intermediary to strip it
      out, forever.


6.  Conclusion

   In conclusion, we feel that benefits of multiplexing of voice and
   video on a single RTP session (and thus single transport connection),
   outweight the drawbacks.  The primary benefit is the impact on NAT
   capacity, which is becoming an important issue in the modern
   Internet.  Furthermore, the unique nature of backwards compatibility
   for rtcweb lessens many of the interoperability concerns, and the
   traditional arguments around multicast and RSVP are simply no longer
   relevant and those technologies have faded from use.


7.  Informative References

   [I-D.perkins-rtcweb-rtp-usage]
              Perkins, C., Westerlund, M., and J. Ott, "RTP Requirements
              for RTC-Web", draft-perkins-rtcweb-rtp-usage-01 (work in
              progress), June 2011.

   [I-D.rosenberg-rtcweb-framework]
              Rosenberg, J., Kaufman, M., Hiie, M., and F. Audet, "An
              Architectural Framework for Browser based Real-Time
              Communications (RTC)", draft-rosenberg-rtcweb-framework-00
              (work in progress), February 2011.

   [I-D.ietf-rtcweb-use-cases-and-requirements]
              Holmberg, C., Hakansson, S., and G. Eriksson, "Web Real-
              Time Communication Use-cases and Requirements",
              draft-ietf-rtcweb-use-cases-and-requirements-01 (work in
              progress), July 2011.

   [RFC5245]  Rosenberg, J., "Interactive Connectivity Establishment
              (ICE): A Protocol for Network Address Translator (NAT)
              Traversal for Offer/Answer Protocols", RFC 5245,
              April 2010.

   [RFC3550]  Schulzrinne, H., Casner, S., Frederick, R., and V.
              Jacobson, "RTP: A Transport Protocol for Real-Time



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              Applications", STD 64, RFC 3550, July 2003.

   [RFC4787]  Audet, F. and C. Jennings, "Network Address Translation
              (NAT) Behavioral Requirements for Unicast UDP", BCP 127,
              RFC 4787, January 2007.

   [RFC4588]  Rey, J., Leon, D., Miyazaki, A., Varsa, V., and R.
              Hakenberg, "RTP Retransmission Payload Format", RFC 4588,
              July 2006.

   [RFC5109]  Li, A., "RTP Payload Format for Generic Forward Error
              Correction", RFC 5109, December 2007.

   [RFC3611]  Friedman, T., Caceres, R., and A. Clark, "RTP Control
              Protocol Extended Reports (RTCP XR)", RFC 3611,
              November 2003.

   [RFC2198]  Perkins, C., Kouvelas, I., Hodson, O., Hardman, V.,
              Handley, M., Bolot, J., Vega-Garcia, A., and S. Fosse-
              Parisis, "RTP Payload for Redundant Audio Data", RFC 2198,
              September 1997.


Authors' Addresses

   Jonathan Rosenberg
   Skype

   Email: jdrosen@skype.net
   URI:   http://www.jdrosen.net


   Cullen Jennings
   Cisco

   Email: fluffy@cisco.com


   Jon Peterson
   Neustar

   Email: jon.peterson@neustar.biz









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   Matthew Kaufman
   Skype

   Email: matthew.kaufman@skype.net


   Eric Rescorla
   RTFM

   Email: ekr@rtfm.com


   Tim Terriberry
   Mozilla

   Email: tterriberry@mozilla.com



































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