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Versions: 00 01 02 draft-ietf-avtcore-rtp-multi-stream

AVTCORE                                                        J. Lennox
Internet-Draft                                                     Vidyo
Updates: 3550 (if approved)                                M. Westerlund
Intended status: Standards Track                                Ericsson
Expires: August 29, 2013                                           Q. Wu
                                                                  Huawei
                                                              C. Perkins
                                                   University of Glasgow
                                                       February 25, 2013


    RTP Considerations for Endpoints Sending Multiple Media Streams
                draft-lennox-avtcore-rtp-multi-stream-02

Abstract

   This document expands and clarifies the behavior of the Real-Time
   Transport Protocol (RTP) endpoints when they are sending multiple
   media streams in a single RTP session.  In particular, issues
   involving Real-Time Transport Control Protocol (RTCP) messages are
   described.

   This document updates RFC 3550 in regards to handling of multiple
   SSRCs per endpoint in RTP sessions.

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 August 29, 2013.

Copyright Notice

   Copyright (c) 2013 IETF Trust and the persons identified as the
   document authors.  All rights reserved.





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   This document is subject to BCP 78 and the IETF Trust's Legal
   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.  Terminology . . . . . . . . . . . . . . . . . . . . . . . . .   3
   3.  Use Cases For Multi-Stream Endpoints  . . . . . . . . . . . .   3
     3.1.  Multiple-Capturer Endpoints . . . . . . . . . . . . . . .   3
     3.2.  Multi-Media Sessions  . . . . . . . . . . . . . . . . . .   4
     3.3.  Multi-Stream Mixers . . . . . . . . . . . . . . . . . . .   4
   4.  Issue Cases . . . . . . . . . . . . . . . . . . . . . . . . .   4
     4.1.  Cascaded Multi-party Conference with Source Projecting
           Mixers  . . . . . . . . . . . . . . . . . . . . . . . . .   5
   5.  Multi-Stream Endpoint RTP Media Recommendations . . . . . . .   5
   6.  Multi-Stream Endpoint RTCP Recommendations  . . . . . . . . .   5
     6.1.  RTCP Reporting Requirement  . . . . . . . . . . . . . . .   6
     6.2.  Initial Reporting Interval  . . . . . . . . . . . . . . .   6
     6.3.  Compound RTCP Packets . . . . . . . . . . . . . . . . . .   6
   7.  RTCP Bandwidth Considerations for Sources with Disparate
       Rates . . . . . . . . . . . . . . . . . . . . . . . . . . . .   7
   8.  Grouping of RTCP Reception Statistics and Other Feedback  . .   7
     8.1.  Semantics and Behavior of Reporting Groups  . . . . . . .   8
     8.2.  Determine the Report Group  . . . . . . . . . . . . . . .   9
     8.3.  RTCP Packet Reporting Group's Reporting Sources . . . . .   9
     8.4.  RTCP Source Description (SDES) item for Reporting Groups   11
     8.5.  Middlebox Considerations  . . . . . . . . . . . . . . . .  11
     8.6.  SDP signaling for Reporting Groups  . . . . . . . . . . .  11
     8.7.  Bandwidth Benefits of RTCP Reporting Groups . . . . . . .  11
     8.8.  Consequences of RTCP Reporting Groups . . . . . . . . . .  12
   9.  Security Considerations . . . . . . . . . . . . . . . . . . .  13
   10. Open Issues . . . . . . . . . . . . . . . . . . . . . . . . .  13
   11. IANA Considerations . . . . . . . . . . . . . . . . . . . . .  13
     11.1.  RTCP SDES Item . . . . . . . . . . . . . . . . . . . . .  13
     11.2.  RTCP Packet Type . . . . . . . . . . . . . . . . . . . .  14
   12. References  . . . . . . . . . . . . . . . . . . . . . . . . .  14
     12.1.  Normative References . . . . . . . . . . . . . . . . . .  14
     12.2.  Informative References . . . . . . . . . . . . . . . . .  14
   Appendix A.  Changes From Earlier Versions  . . . . . . . . . . .  15
     A.1.  Changes From Draft -01  . . . . . . . . . . . . . . . . .  15
     A.2.  Changes From Draft -00  . . . . . . . . . . . . . . . . .  16



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   Authors' Addresses  . . . . . . . . . . . . . . . . . . . . . . .  16

1.  Introduction

   At the time The Real-Time Transport Protocol (RTP) [RFC3550] was
   originally written, and for quite some time after, endpoints in RTP
   sessions typically only transmitted a single media stream per RTP
   session, where separate RTP sessions were typically used for each
   distinct media type.

   Recently, however, a number of scenarios have emerged (discussed
   further in Section 3) in which endpoints wish to send multiple RTP
   media streams, distinguished by distinct RTP synchronization source
   (SSRC) identifiers, in a single RTP session.  Although RTP's initial
   design did consider such scenarios, the specification was not
   consistently written with such use cases in mind.  The specifications
   are thus somewhat unclear.

   The purpose of this document is to expand and clarify [RFC3550]'s
   language for these use cases.  The authors believe this does not
   result in any major normative changes to the RTP specification,
   however this document defines how the RTP specification is to be
   interpreted.  In these cases, this document updates RFC3550.

   The document starts with terminology and some use cases where
   multiple sources will occur.  This is followed by some case studies
   to try to identify issues that exist and need considerations.  This
   is followed by RTP and RTCP recommendations to resolve issues.  Next
   are security considerations and remaining open issues.

2.  Terminology

   The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
   "SHOULD", "SHOULD NOT", "RECOMMENDED", "NOT RECOMMENDED", "MAY", and
   "OPTIONAL" in this document are to be interpreted as described in RFC
   2119 [RFC2119] and indicate requirement levels for compliant
   implementations.

3.  Use Cases For Multi-Stream Endpoints

   This section discusses several use cases that have motivated the
   development of endpoints that send multiple streams in a single RTP
   session.

3.1.  Multiple-Capturer Endpoints

   The most straightforward motivation for an endpoint to send multiple
   media streams in a session is the scenario where an endpoint has



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   multiple capture devices of the same media type and characteristics.
   For example, telepresence endpoints, of the type described by the
   CLUE Telepresence Framework [I-D.ietf-clue-framework] is designed,
   often have multiple cameras or microphones covering various areas of
   a room.

3.2.  Multi-Media Sessions

   Recent work has been done in RTP
   [I-D.ietf-avtcore-multi-media-rtp-session] and SDP
   [I-D.ietf-mmusic-sdp-bundle-negotiation] to update RTP's historical
   assumption that media streams of different media types would always
   be sent on different RTP sessions.  In this work, a single endpoint's
   audio and video media streams (for example) are instead sent in a
   single RTP session.

3.3.  Multi-Stream Mixers

   There are several RTP topologies which can involve a central device
   that itself generates multiple media streams in a session.

   One example is a mixer providing centralized compositing for a multi-
   capture scenario like that described in Section 3.1.  In this case,
   the centralized node is behaving much like a multi-capturer endpoint,
   generating several similar and related sources.

   More complicated is the Source Projecting Mixer, see Section 3.6 of
   [I-D.westerlund-avtcore-rtp-topologies-update].  This is a central
   box that receives media streams from several endpoints, and then
   selectively forwards modified versions of some of the streams toward
   the other endpoints it is connected to.  Toward one destination, a
   separate media source appears in the session for every other source
   connected to the mixer, "projected" from the original streams, but at
   any given time many of them can appear to be inactive (and thus are
   receivers, not senders, in RTP).  This sort of device is closer to
   being an RTP mixer than an RTP translator, in that it terminates RTCP
   reporting about the mixed streams, and it can re-write SSRCs,
   timestamps, and sequence numbers, as well as the contents of the RTP
   payloads, and can turn sources on and off at will without appearing
   to be generating packet loss.  Each projected stream will typically
   preserve its original RTCP source description (SDES) information.

4.  Issue Cases

   This section illustrates some scenarios that have shown areas where
   the RTP specification is unclear.





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4.1.  Cascaded Multi-party Conference with Source Projecting Mixers

   This issue case tries to illustrate the effect of having multiple
   SSRCs sent by an endpoint, by considering the traffic between two
   source-projecting mixers in a large multi-party conference.

   For concreteness, consider a 200-person conference, where 16 sources
   are viewed at any given time.  Assuming participants are distributed
   evenly among the mixers, each mixer would have 100 sources "behind"
   (projected through) it, of which at any given time eight are active
   senders.  Thus, the RTP session between the mixers consists of two
   endpoints, but 200 sources.

   The RTCP bandwidth implications of this scenario are discussed
   further in Section 8.7.

   (TBD: Other examples?  Can this section be removed?)

5.  Multi-Stream Endpoint RTP Media Recommendations

   While an endpoint MUST (of course) stay within its share of the
   available session bandwidth, as determined by signalling and
   congestion control, this need not be applied independently or
   uniformly to each media stream.  In particular, session bandwidth MAY
   be reallocated among an endpoint's media streams, for example by
   varying the bandwidth use of a variable-rate codec, or changing the
   codec used by the media stream, up to the constraints of the
   session's negotiated (or declared) codecs.  This includes enabling or
   disabling media streams as more or less bandwidth becomes available.

6.  Multi-Stream Endpoint RTCP Recommendations

   This section contains a number of different RTCP clarifications or
   recommendations that enables more efficient and simpler behavior
   without loss of functionality.

   The RTP Control Protocol (RTCP) is defined in Section 6 of [RFC3550],
   but it is largely documented in terms of "participants".  In many
   cases, the specification's recommendations for "participants" are to
   be interpreted as applying to individual media streams, rather than
   to endpoints.  This section describes several concrete cases where
   this applies.

   (tbd: rather than think in terms of media streams, it might be
   clearer to refer to SSRC values, where a participant with multiple
   active SSRC values counts as multiple participants, once per SSRC)





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6.1.  RTCP Reporting Requirement

   For each of an endpoint's media streams, whether or not it is
   currently sending media, SR/RR and SDES packets MUST be sent at least
   once per RTCP report interval.  (For discussion of the content of SR
   or RR packets' reception statistic reports, see Section 8.)

6.2.  Initial Reporting Interval

   When a new media stream is added to a unicast session, the sentence
   in [RFC3550]'s Section 6.2 applies: "For unicast sessions ... the
   delay before sending the initial compound RTCP packet MAY be zero."
   This applies to individual media sources as well.  Thus, endpoints
   MAY send an initial RTCP packet for an SSRC immediately upon adding
   it to a unicast session.

   This allowance also applies, as written, when initially joining a
   unicast session.  However, in this case some caution needs to be
   exercised if the end-point or mixer has a large number of sources
   (SSRCs) as this can create a significant burst.  How big an issue
   this depends on the number of source to send initial SR or RR and
   Session Description CNAME items for in relation to the RTCP
   bandwidth.

   (tbd: Maybe some recommendation here?  The aim in restricting this to
   unicast sessions was to avoid this burst of traffic, which the usual
   RTCP timing and reconsideration rules will prevent)

6.3.  Compound RTCP Packets

   Section 6.1 gives the following advice to RTP translators and mixers:

      It is RECOMMENDED that translators and mixers combine individual
      RTCP packets from the multiple sources they are forwarding into
      one compound packet whenever feasible in order to amortize the
      packet overhead (see Section 7).  An example RTCP compound packet
      as might be produced by a mixer is shown in Fig.  1.  If the
      overall length of a compound packet would exceed the MTU of the
      network path, it SHOULD be segmented into multiple shorter
      compound packets to be transmitted in separate packets of the
      underlying protocol.  This does not impair the RTCP bandwidth
      estimation because each compound packet represents at least one
      distinct participant.  Note that each of the compound packets MUST
      begin with an SR or RR packet.

   Note: To avoid confusion, an RTCP packet is an individual item, such
   as a Sender Report (SR), Receiver Report (RR), Source Description
   (SDES), Goodbye (BYE), Application Defined (APP), Feedback [RFC4585]



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   or Extended Report (XR) [RFC3611] packet.  A compound packet is the
   combination of two or more such RTCP packets where the first packet
   has to be an SR or an RR packet, and which contains a SDES packet
   containing an CNAME item.  Thus the above results in compound RTCP
   packets that contain multiple SR or RR packets from different sources
   as well as any of the other packet types.  There are no restrictions
   on the order in which the packets can occur within the compound
   packet, except the regular compound rule, i.e., starting with an SR
   or RR.

   This advice applies to multi-media-stream endpoints as well, with the
   same restrictions and considerations.  (Note, however, that the last
   sentence does not apply to AVPF [RFC4585] or SAVPF [RFC5124] feedback
   packets if Reduced-Size RTCP [RFC5506] is in use.)

   Due to RTCP's randomization of reporting times, there is a fair bit
   of tolerance in precisely when an endpoint schedules RTCP to be sent.
   Thus, one potential way of implementing this recommendation would be
   to randomize all of an endpoint's sources together, with a single
   randomization schedule, so an MTU's worth of RTCP all comes out
   simultaneously.

   (tbd: Multiplexing RTCP packets from multiple different sources might
   require some adjustment to the calculation of RTCP's avg_rtcp_size,
   as the RTCP group interval is proportional to avg_rtcp_size times the
   group size).

7.  RTCP Bandwidth Considerations for Sources with Disparate Rates

   It is possible for an RTP session to carry sources of greatly
   differing bandwidths.  One example is the scenario of
   [I-D.ietf-avtcore-multi-media-rtp-session], when audio and video are
   sent in the same session.  However, this can occur even within a
   single media type, for example a video session carrying both 5 fps
   QCIF and 60 fps 1080p HD video, or an audio session carrying both
   G.729 and L24/48000/6 audio.

   (tbd: recommend how RTCP bandwidths are to be chosen in these
   scenarios.  Likely, these recommendations will be different for
   sessions using AVPF-based profiles (where the trr-int parameter is
   available) than for those using AVP.)

8.  Grouping of RTCP Reception Statistics and Other Feedback

   As specified by [RFC3550], an endpoint MUST send reception reports
   about every active media stream it is receiving, from at least one
   local source.




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   However, a naive application of the RTP specification's rules could
   be quite inefficient.  In particular, if a session has N SSRCs
   (active and inactive, i.e., participant SSRCs), and the session has S
   active senders in each reporting interval, there will either be N*S
   report blocks per reporting interval, or (per the round-robin
   recommendations of [RFC3550] Section 6.1) reception sources would be
   unnecessarily round-robinned.  In a session where most media sources
   become senders reasonably frequently, this results in quadratically
   many reception report blocks in the conference, or reporting delays
   proportional to the number of session members.

   Since traffic is received by endpoints, however, rather than by media
   sources, there is not actually any need for this quadratic expansion.
   All that is needed is for each endpoint to report all the remote
   sources it is receiving.

   Thus, this document defines a new RTCP mechanism, Reporting Groups,
   to indicate sources which originate from the same endpoint, and which
   therefore would have identical recption reports.

8.1.  Semantics and Behavior of Reporting Groups

   An RTCP Reporting Group indicates that a set of sources (SSRCs) that
   originate from a single entity (endpoint or middlebox) in an RTP
   session, and therefore all the sources in the group's view of the
   network is identical.  If reporting groups are in use, two sources
   SHOULD be put into the same reporting group if their view of the
   network is identical; i.e., if they report on traffic received at the
   same interface of an RTP endpoint.  Sources with different views of
   the network MUST NOT be put into the same reporting group.

   If reporting groups are in use, an endpoint MUST NOT send reception
   reports from one source in a reporting group about another one in the
   same group ("self-reports").  Similarly, an endpoint MUST NOT send
   reception reports about a remote media source from more than one
   source in a reporting group ("cross-reports").  Instead, it MUST pick
   one of its local media sources as the "reporting" source for each
   remote media source, and use it to send reception reports about that
   remote source; all the other media sources in the reporting group
   MUST NOT send any reception reports about that remote media source.

   This reporting source MUST also be the source for any RTP/AVPF
   Feedback [RFC4585] or Extended Report (XR) [RFC3611] packets about
   the corresponding remote sources as well.  If a reporting source
   leaves the session (i.e., if it sends a BYE, or leaves the group
   without sending BYE under the rules of [RFC3550] section 6.3.7),
   another reporting source MUST be chosen if any members of the group
   still exist.



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   An endpoint or middlebox MAY use multiple sources as reporting
   sources; however, each reporting source MUST have non-overlapping
   sets of remote SSRCs it reports on.  This is primarily to be done
   when the reporting source's number of reception report blocks is so
   large that it would be forced to round robin around the sources.
   Thus, by splitting the reports among several reporting SSRCs more
   consistent reporting can be achieved.

   If RTP/AVPF feedback is in use, a reporting source MAY send immediate
   or early feedback at any point when any member of the reporting group
   could validly do so.

   An endpoint SHOULD NOT create single-source reporting groups, unless
   it is anticipated that the group might have additional sources added
   to it in the future.

8.2.  Determine the Report Group

   A remote RTP entity, such as an endpoint or a middlebox needs to be
   able to determine the report group used by another RTP entity.  To
   achieve this goal two RTCP extensions has been defined.  For the
   SSRCs that are reporting on behalf of the reporting group an SDES
   item RGRP has been defined for providing the report group with an
   identifier.  For SSRCs that aren't reporting on any peer SSRC a new
   RTCP packet type is defined.  This RTCP packet type "Reporting
   Sources", lists the SSRC that are reporting on this SSRC's behalf.

   This divided approach has been selected for the following reasons:

   1.  Enable an explicit indication of who reports on this SSRC's
       behalf.  Being explicit prevents the remote entity from detecting
       that is missing the reports if there issues with the reporting
       SSRC's RTCP packets.

   2.  Enable explicit identification of the SSRCs that are actively
       reporting as one entity.

8.3.  RTCP Packet Reporting Group's Reporting Sources

   This section defines a new RTCP packet type called "Reporting Group's
   Reporting Sources" (RGRS).  It identifies the SSRC(s) that report on
   behalf of the SSRC that is the sender of the RGRS packet.

   This packet consists of the fixed RTCP packet header which indicates
   the packet type, the number of reporting sources included and the
   SSRC which behalf the reporting SSRCs report on.  This is followed by
   the list of reporting SSRCs.




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    0                   1                   2                   3
    0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |V=2|P|    SC   | PT=RGRS(TBA)  |             length            |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                     SSRC of packet sender                     |
   +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
   :                     SSRC for Reporting Source                 :
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+


   The RTCP Packets field has the following definition.

   version (V):  This field identifies the RTP version.  The current
      version is 2.

   padding (P):  1 bit If set, the padding bit indicates that the packet
      contains additional padding octets at the end that are not part of
      the control information but are included in the length field.  See
      [RFC3550].

   Source Count (SC):  5 bits Indicating the number of reporting SSRCs
      (1-31) that are include in this RTCP packet type.

   Payload type (PT):  8 bits This is the RTCP packet type that
      identifies the packet as being an RTCP FB message.  The RGRS RTCP
      packet has the value [TBA].

   Length:  16 bits The length of this packet in 32-bit words minus one,
      including the header and any padding.  This is in line with the
      definition of the length field used in RTCP sender and receiver
      reports [RFC3550].

   SSRC of packet sender:  32 bits.  The SSRC of the sender of this
      packet which indicates which SSRCs that reports on its behalf,
      instead of reporting itself.

   SSRC for Reporting Source:  A variable number (as indicated by Source
      Count) of 32-bit SSRC values.  Each SSRC is an reporting SSRC
      belonging to the same Report Group.

   Each RGRS packet MUST contain at least one reporting SSRC.  In case
   the reporting SSRC field is insufficient to list all the SSRCs that
   is reporting in this report group, the SSRC SHALL round robin around
   the reporting sources.






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   Any RTP mixer or translator which forwards SR or RR packets from
   members of a reporting group MUST forward the corresponding RGRS RTCP
   packet as well.

8.4.  RTCP Source Description (SDES) item for Reporting Groups

   A new RTCP Source Description (SDES) item is defined for the purpose
   of identifying reporting groups.

   The Source Description (SDES) item "RGRP" is sent by a reporting
   group's reporting SSRC.  Syntactically, its format is the same as the
   RTCP SDES CNAME item [RFC6222], and MUST be chosen with the same
   global-uniqueness and privacy considerations as CNAME.  This name
   MUST be stable across the lifetime of the reporting group, even if
   the SSRC of a reporting source changes.

   Every source which belongs to a reporting group MUST either include
   an RGRP SDES item in an SDES packet (if it is a reporting source), or
   an RGRS packet (if it is not), in every compound RTCP packet in which
   it sends an RR or SR packet (i.e., in every RTCP packet it sends,
   unless Reduced-Size RTCP [RFC5506] is in use).

   Any RTP mixer or translator which forwards SR or RR packets from
   members of a reporting group MUST forward the corresponding RGRP SDES
   items as well, even if it otherwise strips SDES items other than
   CNAME.

8.5.  Middlebox Considerations

   This section discusses middlebox considerations for Reporting groups.

   To be expanded.

8.6.  SDP signaling for Reporting Groups

   TBD

8.7.  Bandwidth Benefits of RTCP Reporting Groups

   To understand the benefits of RTCP reporting groups, consider the
   scenario described in Section 4.1.  This scenario describes an
   environment in which the two endpoints in a session each have a
   hundred sources, of which eight each are sending within any given
   reporting interval.

   For ease of analysis, we can make the simplifying approximation that
   the duration of the RTCP reporting interval is equal to the total
   size of the RTCP packets sent during an RTCP interval, divided by the



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   RTCP bandwidth.  (This will be approximately true in scenarios where
   the bandwidth is not so high that the minimum RTCP interval is
   reached.)  For further simplification, we can assume RTCP senders are
   following the recommendations of Section 6.3; thus, the per-packet
   transport-layer overhead will be small relative to the RTCP data.
   Thus, only the actual RTCP data itself need be considered.

   In a report interval in this scenario, there will, as a baseline, be
   200 SDES packets, 184 RR packets, and 16 SR packets.  This amounts to
   approximately 6.5 kB of RTCP per report interval, assuming 16-byte
   CNAMEs and no other SDES information.

   Using the original [RFC3550] everyone-reports-on-every-sender
   feedback rules, each of the 184 receivers will send 16 report blocks,
   and each of the 16 senders will send 15.  This amounts to
   approximately 76 kB of report block traffic per interval; 92% of RTCP
   traffic consists of report blocks.

   If reporting groups are used, however, there is only 0.4 kB of
   reports per interval, with no loss of useful information.
   Additionally, there will be (assuming 16-byte RGRPs, and a single
   reporting source per reporting group) an additional 2.4 kB per cycle
   of RGRP SDES items and RGRS packets.  Put another way, the unmodified
   [RFC3550] reporting interval is approximately 8.9 times longer than
   if reporting groups are in use.

8.8.  Consequences of RTCP Reporting Groups

   The RTCP traffic generated by receivers using RTCP Reporting Groups
   might appear, to observers unaware of these semantics, to be
   generated by receivers who are experiencing a network disconnection,
   as the non-reporting sources appear not to be receiving a given
   sender at all.

   This could be a potentially critical problem for such a sender using
   RTCP for congestion control, as such a sender might think that it is
   sending so much traffic that it is causing complete congestion
   collapse.

   However, such an interpretation of the session statistics would
   require a fairly sophisticated RTCP analysis.  Any receiver of RTCP
   statistics which is just interested in information about itself needs
   to be prepared that any given reception report might not contain
   information about a specific media source, because reception reports
   in large conferences can be round-robined.

   Thus, it is unclear to what extent such backward compatibility issues
   would actually cause trouble in practice.



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9.  Security Considerations

   In the secure RTP protocol (SRTP) [RFC3711], the cryptographic
   context of a compound SRTCP packet is the SSRC of the sender of the
   first RTCP (sub-)packet.  This could matter in some cases, especially
   for keying mechanisms such as Mikey [RFC3830] which use per-SSRC
   keying.

   Other than that, the standard security considerations of RTP apply;
   sending multiple media streams from a single endpoint does not appear
   to have different security consequences than sending the same number
   of streams.

10.  Open Issues

   At this stage this document contains a number of open issues.  The
   below list tries to summarize the issues:

   1.  Further clarifications on how to handle the RTCP scheduler when
       sending multiple sources in one compound packet.

   2.  How is the use of reporting groups be signaled in SDP?

   3.  How is the RTCP avg_rtcp_size be calculated when RTCP packets are
       routinely multiplexed among multiple RTCP senders?

   4.  Do we need to provide a recommendation for unicast session
       joiners with many sources to not use 0 initial minimal interval
       from bit-rate burst perspective?

11.  IANA Considerations

   This document make several requests to IANA for registering new RTP/
   RTCP identifiers.

   (Note to the RFC-Editor: please replace "TBA" with the IANA-assigned
   value, and "XXXX" with the number of this document, prior to
   publication as an RFC.)

11.1.  RTCP SDES Item

   This document adds an additional SDES types to the IANA "RTCP SDES
   Item Types" Registry, as follows:

   Value    Abbrev      Name              Reference
   TBA      RGRP        Reporting Group   [RFCXXXX]

           Figure 1: Item for the IANA Source Attribute Registry



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11.2.  RTCP Packet Type

   This document defines one new RTCP Control Packet types (PT) to be
   registered as follows:

   Value    Abbrev      Name                                Reference
   TBA      RGRR        Reporting Group Reporting Sources   [RFCXXXX]

    Figure 2: Item for the IANA RTCP Control Packet Types (PT) Registry

12.  References

12.1.  Normative References

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

   [RFC3550]  Schulzrinne, H., Casner, S., Frederick, R., and V.
              Jacobson, "RTP: A Transport Protocol for Real-Time
              Applications", STD 64, RFC 3550, July 2003.

   [RFC3711]  Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K.
              Norrman, "The Secure Real-time Transport Protocol (SRTP)",
              RFC 3711, March 2004.

   [RFC4585]  Ott, J., Wenger, S., Sato, N., Burmeister, C., and J. Rey,
              "Extended RTP Profile for Real-time Transport Control
              Protocol (RTCP)-Based Feedback (RTP/AVPF)", RFC 4585, July
              2006.

   [RFC5124]  Ott, J. and E. Carrara, "Extended Secure RTP Profile for
              Real-time Transport Control Protocol (RTCP)-Based Feedback
              (RTP/SAVPF)", RFC 5124, February 2008.

   [RFC5506]  Johansson, I. and M. Westerlund, "Support for Reduced-Size
              Real-Time Transport Control Protocol (RTCP): Opportunities
              and Consequences", RFC 5506, April 2009.

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

12.2.  Informative References

   [I-D.ietf-avtcore-multi-media-rtp-session]
              Westerlund, M., Perkins, C., and J. Lennox, "Multiple
              Media Types in an RTP Session", draft-ietf-avtcore-multi-
              media-rtp-session-01 (work in progress), October 2012.



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   [I-D.ietf-clue-framework]
              Duckworth, M., Pepperell, A., and S. Wenger, "Framework
              for Telepresence Multi-Streams", draft-ietf-clue-
              framework-09 (work in progress), February 2013.

   [I-D.ietf-mmusic-sdp-bundle-negotiation]
              Holmberg, C., Alvestrand, H., and C. Jennings,
              "Multiplexing Negotiation Using Session Description
              Protocol (SDP) Port Numbers", draft-ietf-mmusic-sdp-
              bundle-negotiation-03 (work in progress), February 2013.

   [I-D.westerlund-avtcore-rtp-topologies-update]
              Westerlund, M. and S. Wenger, "RTP Topologies", draft-
              westerlund-avtcore-rtp-topologies-update-01 (work in
              progress), October 2012.

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

   [RFC3830]  Arkko, J., Carrara, E., Lindholm, F., Naslund, M., and K.
              Norrman, "MIKEY: Multimedia Internet KEYing", RFC 3830,
              August 2004.

Appendix A.  Changes From Earlier Versions

   Note to the RFC-Editor: please remove this section prior to
   publication as an RFC.

A.1.  Changes From Draft -01

   o  Merged with draft-wu-avtcore-multisrc-endpoint-adver.

   o  Changed how Reporting Groups are indicated in RTCP, to make it
      clear which source(s) is the group's reporting sources.

   o  Clarified the rules for when sources can be placed in the same
      reporting group.

   o  Clarified that mixers and translators need to pass reporting group
      SDES information if they are forwarding RR and SR traffic from
      members of a reporting group.









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A.2.  Changes From Draft -00

   o  Added the Reporting Group semantic to explicitly indicate which
      sources come from a single endpoint, rather than leaving it
      implicit.

   o  Specified that Reporting Group semantics (as they now are) apply
      to AVPF and XR, as well as to RR/SR report blocks.

   o  Added a description of the cascaded source-projecting mixer, along
      with a calculation of its RTCP overhead if reporting groups are
      not in use.

   o  Gave some guidance on how the flexibility of RTCP randomization
      allows some freedom in RTCP multiplexing.

   o  Clarified the language of several of the recommendations.

   o  Added an open issue discussing how avg_rtcp_size ought to be
      calculated for multiplexed RTCP.

   o  Added an open issue discussing how RTCP bandwidths are to be
      chosen for sessions where source bandwidths greatly differ.

Authors' Addresses

   Jonathan Lennox
   Vidyo, Inc.
   433 Hackensack Avenue
   Seventh Floor
   Hackensack, NJ  07601
   US

   Email: jonathan@vidyo.com


   Magnus Westerlund
   Ericsson
   Farogatan 6
   SE-164 80 Kista
   Sweden

   Phone: +46 10 714 82 87
   Email: magnus.westerlund@ericsson.com







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

   Email: sunseawq@huawei.com


   Colin Perkins
   University of Glasgow
   School of Computing Science
   Glasgow  G12 8QQ
   United Kingdom

   Email: csp@csperkins.org


































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