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Audio/Video Transport Working Group                                Q. Wu
Internet-Draft                                                    Huawei
Intended status: Standards Track                        October 12, 2012
Expires: April 15, 2013


       Bandwidth and RTCP timing issues for multi-source endpoint
         draft-wu-avtcore-multiplex-multisource-endpoint-00.txt

Abstract

   This document discusses bandwidth issues and RTCP timing rule issues
   that arise when the multi-source endpoint multiplexing all the media
   type in one RTP session and follows RFC3550 timing rules.  It
   provides recommendations for multi-source host sending multiple media
   types in the same session.

Status of this Memo

   This Internet-Draft is submitted in full conformance with the
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   This Internet-Draft will expire on April 15, 2013.

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   Copyright (c) 2012 IETF Trust and the persons identified as the
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   described in the Simplified BSD License.



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Table of Contents

   1.  Introduction . . . . . . . . . . . . . . . . . . . . . . . . .  3
   2.  Terminology  . . . . . . . . . . . . . . . . . . . . . . . . .  4
     2.1.  Standards Language . . . . . . . . . . . . . . . . . . . .  4
   3.  Use Cases for multi-source host communications . . . . . . . .  5
     3.1.  One to one Communication between multi-source endpoints  .  5
     3.2.  Multi-party Communication between multi-source
           endpoints  . . . . . . . . . . . . . . . . . . . . . . . .  6
     3.3.  Communication between multi-source endpoint
           multiplexing each medium in separate RTP session and
           multiple-source endpoint multiplexing all mediums in
           one RTP session  . . . . . . . . . . . . . . . . . . . . .  7
   4.  Discuss  . . . . . . . . . . . . . . . . . . . . . . . . . . .  9
   5.  Recommendations  . . . . . . . . . . . . . . . . . . . . . . . 10
     5.1.  Choosing RTCP Bandwidth  . . . . . . . . . . . . . . . . . 10
     5.2.  Maintaining the number of session members and senders  . . 11
     5.3.  RTCP Reporting Interval calculation  . . . . . . . . . . . 12
     5.4.  RTCP reception report  . . . . . . . . . . . . . . . . . . 12
   6.  Security Considerations  . . . . . . . . . . . . . . . . . . . 13
   7.  IANA Considerations  . . . . . . . . . . . . . . . . . . . . . 14
   8.  References . . . . . . . . . . . . . . . . . . . . . . . . . . 15
     8.1.  Normative References . . . . . . . . . . . . . . . . . . . 15
     8.2.  Informative References . . . . . . . . . . . . . . . . . . 15
   Author's Address . . . . . . . . . . . . . . . . . . . . . . . . . 16


























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1.  Introduction

   Multiplexing is a method by which multiple streams are combined into
   one stream over a shared medium.  In RTP, multiplexing is provided by
   the destination transport address (network address and port number)
   which is different for each RTP session.  Given each media type
   having different requirement for bandwidth, multiple media types
   (e.g., Separate audio and video streams) are usually not carried in a
   single RTP session.

   However for some applications that use unicast transport, e.g., in
   RTCWeb application, multiple-source hosts may use a different SSRC
   for each medium but sending them in the same RTP session, which
   reduces communication failure due to NAT and firewall when using
   multiple RTP sessions or multiple transport flow.  If these multi-
   source hosts still follow RFC3550 timing rules, audio and video are
   multiplexed onto a single RTP session and share a common session
   bandwidth, the audio flows sending at much lower rates will waste a
   large amount of bandwidth.

   This document provides recommendations for multi-source host sending
   multiple media types in the same session.





























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2.  Terminology

2.1.  Standards Language

   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 RFC 2119 [RFC2119].

   Multi-Source Endpoint

      End system with multiple sources generated from one host and
      running on that host.  One example of multi-source endpoint is an
      RTP endpoint which has multiple capture devices of the same media
      type and characteristics.

   Single-Source Endpoint
      End system with single source generated from one host described in
      RFC3550.

































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3.  Use Cases for multi-source host communications

3.1.  One to one Communication between multi-source endpoints

          Multiple-Source              Multiple-Source
              Endpoint1                    Endpoint2
             +--------+                   +--------+
             |        |  Common Transport |        |
             |        |                   |        |
             |        |Transport flow1(v1,a1)      |
   SSRC-V1---|        |------------------>|        |--- SSRC-V2
             |        |                   |        |
             |        |                   |        |
             |        |                   |        |
             |        |                   |        |
   SSRC-A1---|        |Transport flow1(v2,a2)      |
             |        |<------------------|        +----SSRC-A2
             |        |                   |        |
             |        |                   |        |
             |        |                   |        |
             +--------+                   +--------+

                                 Figure 1

   In the figure 1, one to one communication between multiple-source
   endpoint1 and multiple-source endpoint2 takes place.  In order to
   reduce unicast transport flow to facilitate NAT/FW traversal,
   Multiple-Source endpoint 1 and multiple-source endpoint2 in one to
   one communication share the same transport.  Both video stream v1 and
   audio stream a1 from multiple-source endpoint 1 are multiplexed in
   the RTP session1 and transmitted over the transport flow1 to
   multiple-source endpoint 2.  Similarly, both video stream v2 and
   audio stream a2 from multiple-source endpoint2 are multiplexed in the
   RTP session 2 and transmitted over the same transmitted over the same
   transport flow 1 as multiple source endpoint1.

   Multiple-source endpoint1 sends RTCP SR and RR packets for every
   active media stream it's receiving (i.e.,v2,a2) from every local
   source (i.e.,v1,a1), which wastes a lot of bandwidth for redundant
   statistics reports.  So does multiple-source endpoint2.

   Also the bandwidth usage for each stream is limited by its media data
   rate and audio stream usually consumes lower bandwidth than video
   stream (e.g.,high quality audio with 80kbps and high quality video
   with 350 kbps).  When audio stream and video stream share the same
   transport, i.e., sharing the common session bandwidth between audio
   and video, audio stream will still be sent at its own media data rate
   far below the common session bandwidth and therefore waste lots of



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   bandwidth provided by the transport.  Further when reception reports
   are used for audio stream and follows RFC3550 timing rule, RTCP SR
   and RR packets for active audio stream will be reported more frequent
   than when audio and video are carried in separate transports since
   more RTCP bandwidth are allocated to audio stream reporting when more
   session bandwidth are allocated to audio stream.

3.2.  Multi-party Communication between multi-source endpoints

                                +---+
                                |   |
                       Transport flow 1(v1,a1)-----------+
                       Transport flow 1(v2,a2)           |
                             +--+ c +-------->           |
                             |  | e |        | Multiple  |-----SSRC-V2
                             |  | n |        |  Source   |
             +-----------+   |  | t |        | Endpoint2 |
             |           <---|  | e |        |           |
   SSRC-V1---|           |      | r |        |           +-----SSRC-A2
             | Multiple  |      |   |        +-----------+
             |  Source   |      | s |
             | Endpoint1 |      | e |
   SSRC-A1---|           <---|  | r |        +-----------+
             |           |   |  | v |        |           |
             +-----------+   |  | e |        |           |
                             |  | r |        | Multiple  |-----SSRC-V3
                             |  |               Source   |
                       Transport flow 2(v1,a1) Endpoint3 |
                       Transport flow 2(v3,a3)           |
                             ---|   +-------->           +-----SSRC-A3
                                |   |        +-----------+
                                |   |
                                |   |
                                +---+

                                 Figure 2

   In figure 2, multiple-source endpoint1 communicates with multiple-
   source endpoint2 and multiple-source endpoint3 in the same session.
   In order to reduce unicast transport flow to facilitate NAT/FW
   traversal, multiple-source endpoint 1 shares the same transport flow
   1 with multiple-source endpoint2 to send video stream v1 and audio
   stream a1 and receive video stream v2 and audio stream v2
   simultaneously.  Similarly, multiple-source endpoint 1 shares the
   same transport flow 2 with multiple-source endpoint3 to send video
   stream v1 and audio stream a1 and receive video stream v3 and audio
   stream a3 simultaneously.




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   Multiple-source endpoint1 sends RTCP SR and RR packets for every
   active media stream it's receiving (i.e.,v2,a2,v3,a3) from every
   local source (i.e.,v1,a1), which wastes a lot of bandwidth for
   redundant statistics reports.  So does multiple-source endpoint2.

   Also the bandwidth usage for each stream is limited by its media data
   rate and audio stream usually consumes lower bandwidth than video
   stream (e.g.,high quality audio with 80kbps and high quality video
   with 350 kbps).  When audio stream and video stream share the same
   transport, i.e., sharing the common session bandwidth between audio
   and video, audio stream will still be sent at its own media data rate
   far below the common session bandwidth and therefore waste lots of
   bandwidth provided by the transport.  Further when reception reports
   are used for audio stream and follows RFC3550 timing rule, RTCP SR
   and RR packets for active audio stream will be reported more frequent
   than when audio and video are carried in separate transports since
   more RTCP bandwidth are allocated to audio stream reporting when more
   session bandwidth are allocated to audio stream.

3.3.  Communication between multi-source endpoint multiplexing each
      medium in separate RTP session and multiple-source endpoint
      multiplexing all mediums in one RTP session

          Multiple-Source              Multiple-Source
              Endpoint1                    Endpoint2
             +--------+                   +--------+
             |        | Various Transports|        |
             |        |                   |        |
             |        |Transport flow1(v1,a1)      |
   SSRC-V1---|        |------------------>|        |--- SSRC-V2
             |        |                   |        |
             |        |Transport flow2(v2)|        |
             |        |<------------------|        |
             |        |                   |        |
   SSRC-A1---|        |Transport flow3(a2)|        |
             |        |<------------------|        +----SSRC-A2
             |        |                   |        |
             |        |                   |        |
             |        |                   |        |
             +--------+                   +--------+

   In the figure 3, multiple-source endpoint1 residing outside of the
   firewall communicates with multiple-source endpoint2 residing inside
   of the firewall.  Unlike one to one communication described in
   section 3.1, multiple-source endpoint2 multiplexes each medium
   (i.e.,v2,a2) in separate transports (i.e.,transport flow2,transport
   flow3)to multiple-source endpoint1 while multiple-source endpoint1
   multiplexes all mediums(ie.,v1,a1) in one unicast



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   transport(i.e.,transport flow1)to multiple-source endpoint2.


















































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4.  Discuss

   The RTCP bandwidth fraction is derived from the media data rate.  If
   audio and video are sent as two separate RTP sessions, they would
   naturally have different RTCP bandwidth fractions, since the two
   media types have different rates.

   However, if audio and video are multiplexed onto a single RTP
   session, a common session bandwidth would have to be chosen.  This
   common bandwidth will likely be inappropriate for one of the media
   types, leading to the situation where some media flows use their
   allocation, while some flows send at a rate that is quite different
   to the session bandwidth.  In the common case, the video sends at the
   session bandwidth, while the audio flows send at much lower rates.
   The RTCP bandwidth is derived from the session bandwidth, which means
   it's appropriate for the video, but is too high for the audio.  In
   the worst case, the audio flows can have more RTCP than video flows,
   which is very wasteful.  Fixing this is potentially difficult, since
   the session bandwidth concept is baked into all the RTCP timing
   rules.































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5.  Recommendations

   Senders and receivers which are not multi-source endpoints are not
   affected by bandwidth issues associated with multi-source endpoint
   and should follow RFC3550 timing rules[RFC3550], no special
   accommodation is required.

   Senders and receivers which are multi-source endpoints and sending or
   receiving multiple media types over different transport should be
   treated in the same way as single source endpoints dealing with
   multiple streams in the same media type.

   Senders and receivers which are multi-source endpoints and sending or
   receiving multiple media types MAY multiplex/demultiplex each media
   type in separate RTP session or multiplex/demultiplex all media types
   in one RTP session.  The multiplexing/demultiplexing mode to be
   employed in two directions between senders and receivers should be
   configurable.  It is RECOMMENDED that

   1.  As the default behavior, Senders and receivers use the media
       bundling mechanism [BUNDLE] in two directions between senders and
       receivers, i.e., multiplexing/demulplexing separate media types
       in one RTP session over the same lower layer transport.

   2.  Configuration or local policy on the senders or receivers can
       override the default Mechanism specified in Option 1 above in one
       or two direction.  Therefore senders and receivers MAY be
       configured with the different multiplexing mode.

   3.  Dynamic multiplexing negotiation mechanism [BUNDLE] can be used
       to signal which multiplexing mechanism is used between senders
       and receivers and override the default mechanism specified in
       Option 1 and 2 above.  The employed multiplexing negotiation
       mechanism is outside the scope of this document.

   Multi-source endpoints multiplexing each media type in separate RTP
   session SHOULD be treated as multiple endpoints for each media type.

5.1.  Choosing RTCP Bandwidth

   Multi-source endpoint multiplexing multiple media types in the same
   RTP session SHOULD specify RTCP bandwidth using SDP, in a "b=RS" line
   or a "b=RR" line rather than choosing 5% of session bandwidth for
   RTCP bandwidth, especially when audio stream and video stream are
   sent over the same transport and the media data rate of audio stream
   is far less than video stream.

   RTCP bandwidth for video stream MAY still follow RFC3550



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   recommendation, i.e., the fraction of the session bandwidth added for
   RTCP be fixed at 5%.

   RTCP bandwidth for video = session bandwidth *5%

   However given lower bandwidth usage of audio stream, RTCP bandwidth
   for audio stream is RECOMMENDED to be specified using SDP in a "b=RS"
   line or a "b=RR" line.

   RTCP bandwidth for audio stream SHOULD be calculated based on the
   following formula:

   RTCP bandwidth for audio = ((audio codec maximum bitrate*20%)+ audio
   codec maximum bitrate)*5%

   Here a 20% RTP packet overhead is added to the data rate to calculate
   the required RTCP bandwidth.

5.2.  Maintaining the number of session members and senders

   When multi-Source endpoints multiplexing all media types in the same
   RTP session join the session, they SHOULD be counted as only one
   active sender.  To achieve this, the sender which is multi-source
   endpoint SHOULD not send Reception reports (e.g., SR/RR RTCP packets)
   from all local sources to each remote source in the receiving side,
   Similarly the Receiver which is multi-source endpoint SHOULD not
   receive reception reports(e.g., SR/RR RTCP packets from each remote
   source in the sending side to all local sources.  Instead, one
   designated local source from the senders within multi-source endpoint
   should be chosen as reporting source for other local sources within
   the sender.

   When multi-Source endpoints multiplexing each media type in separated
   RTP session join the session, they SHOULD treated as multiple single-
   source endpoint multiplexing for each media type and SHOULD be
   counted as multiple active senders.  The number of active senders
   within multi-source endpoint is decided by the number of local source
   which is sending reception reports.

   For the number of session members, it depends on the number of
   multiple-source endpoints E and the number of local source in each
   multiple-source endpoint S. It can be estimated or calculated
   according to the rules in RFC 3550 section 6.2.1, based on received
   valid SSRC in the SDES packet or RTP packets.







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5.3.  RTCP Reporting Interval calculation

   The RTCP packet interval calculation SHOULD consider RTCP bandwidth
   estimation and signaling in section 5.1 and active sender estimation
   in section 5.2.  The RTCP reporting interval for audio stream and
   video stream should be calculated according to the rules in RFC3550
   section 6.2 respectively.  When RTCP reception reports for audio
   stream and video stream are sent in different intervals, in order to
   decrease the number of RTCP packet to be sent, it is more desirable
   to transmit reception report for audio stream and reception report
   for video stream in the same compound RTCP packet and set RTCP
   reporting interval for audio stream as a multiple of RTCP reporting
   intervals for video stream.

   For example, if the RTCP Reporting Interval for audio stream is more
   than one RTCP Reporting Interval for video but less than two RTCP
   Reporting Intervals for video, it is RECOMMENDED the RTCP Reporting
   Interval for audio stream be chosen as one RTCP Reporting Interval
   for video.

   If the RTCP Reporting Interval for audio stream is more than two RTCP
   Reporting Intervals for video but less than three RTCP Reporting
   Intervals for video, it is RECOMMENDED the RTCP Reporting Interval
   for audio stream be chosen as two RTCP Reporting Intervals for video.

   If the RTCP Reporting Interval for audio stream is more than three
   RTCP Reporting Intervals for video but less than four RTCP Reporting
   Intervals for video, it is RECOMMENDED the RTCP Reporting Interval
   for audio stream be chosen as three RTCP Reporting Intervals for
   video.

5.4.  RTCP reception report

   Multiple-Source Endpoint SHOULD NOT send reception reports from one
   of its source about all the other local sources of its own.  RTP
   application SHOULD provide a means to identify multiple-source
   endpoint as in fact being sources from the same RTP node.

   Multiple-Source Endpoint SHOULD combine RTCP reception reports into a
   single compound RTCP packet without exceeding the maximum
   transmission unit (MTU) of the network path.










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6.  Security Considerations

   TBC.
















































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7.  IANA Considerations

   This document has no actions for IANA.
















































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8.  References

8.1.  Normative References

   [BUNDLE]   Holmberg, C. and H. Alvestrand, "Extended RTP Profile for
              Real-time Transport Control Protocol (RTCP)-Based Feedback
              (RTP/AVPF)",
              ID draft-ietf-mmusic-sdp-bundle-negotiation-01,
              August 2012.

   [RFC2119]  Bradner, S., "Key words for use in RFCs to Indicate
              Requirement Levels", March 1997.

   [RFC3550]  Schulzrinne, H., "RTP: A Transport Protocol for Real-Time
              Applications", RFC 3550, July 2003.

   [RFC3556]  Casner, S., "Session Description Protocol (SDP) Bandwidth
              Modifiers for RTP Control Protocol (RTCP) Bandwidth",
              RFC 3556, July 2003.

8.2.  Informative References

   [I-D.westerlund-avtcore-multiplex-architecture]
              Westerlund, M., "Guidelines for using the Multiplexing
              Features of RTP",
              ID draft-westerlund-avtcore-multiplex-architecture-02,
              July 2012.
























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Author's Address

   Qin Wu
   Huawei
   101 Software Avenue, Yuhua District
   Nanjing, Jiangsu  210012
   China

   Email: sunseawq@huawei.com










































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