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Versions: (draft-lennox-avtcore-rtp-multi-stream) 00 01 02 03 04 05 06 07 08 09 10 11 RFC 8108

AVTCORE                                                        J. Lennox
Internet-Draft                                                     Vidyo
Updates: 3550, 4585 (if approved)                          M. Westerlund
Intended status: Standards Track                                Ericsson
Expires: January 04, 2015                                          Q. Wu
                                                                  Huawei
                                                              C. Perkins
                                                   University of Glasgow
                                                           July 03, 2014


         Sending Multiple Media Streams in a Single RTP Session
                 draft-ietf-avtcore-rtp-multi-stream-05

Abstract

   This memo expands and clarifies the behaviour of Real-time Transport
   Protocol (RTP) endpoints that use multiple synchronization sources
   (SSRCs).  This occurs, for example, when an endpoint sends multiple
   media streams in a single RTP session.  This memo updates RFC 3550
   with regards to handling multiple SSRCs per endpoint in RTP sessions,
   with a particular focus on RTCP behaviour.  It also updates RFC 4585
   to update and clarify the calculation of the timeout of SSRCs and the
   inclusion of feedback messages.

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
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   Drafts is at http://datatracker.ietf.org/drafts/current/.

   Internet-Drafts are draft documents valid for a maximum of six months
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   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 04, 2015.

Copyright Notice

   Copyright (c) 2014 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
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   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  . . . . . . . . . . . . . . . . . . . . . . . .   2
   2.  Terminology . . . . . . . . . . . . . . . . . . . . . . . . .   3
   3.  Use Cases For Multi-Stream Endpoints  . . . . . . . . . . . .   3
     3.1.  Endpoints with Multiple Capture Devices . . . . . . . . .   3
     3.2.  Multiple Media Types in a Single RTP Session  . . . . . .   3
     3.3.  Multiple Stream Mixers  . . . . . . . . . . . . . . . . .   4
     3.4.  Multiple SSRCs for a Single Media Source  . . . . . . . .   4
   4.  Use of RTP by endpoints that send multiple media streams  . .   5
   5.  Use of RTCP by Endpoints that send multiple media streams . .   5
     5.1.  RTCP Reporting Requirement  . . . . . . . . . . . . . . .   5
     5.2.  Initial Reporting Interval  . . . . . . . . . . . . . . .   5
     5.3.  Aggregation of Reports into Compound RTCP Packets . . . .   6
       5.3.1.  Maintaining AVG_RTCP_SIZE . . . . . . . . . . . . . .   7
       5.3.2.  Scheduling RTCP with Multiple Reporting SSRCs . . . .   8
     5.4.  Use of RTP/AVPF Feedback  . . . . . . . . . . . . . . . .  10
       5.4.1.  Choice of SSRC for Feedback Packets . . . . . . . . .  10
       5.4.2.  Scheduling an RTCP Feedback Packet  . . . . . . . . .  11
   6.  RTCP Considerations for Streams with Disparate Rates  . . . .  12
     6.1.  Timing out SSRCs  . . . . . . . . . . . . . . . . . . . .  13
       6.1.1.  AVPF T_rr_interval Behaviour  . . . . . . . . . . . .  13
       6.1.2.  Avoiding Premature Timeout  . . . . . . . . . . . . .  14
       6.1.3.  RTP/AVP and RTP/AVPF Interoperability . . . . . . . .  15
       6.1.4.  Specified Behaviour . . . . . . . . . . . . . . . . .  16
     6.2.  Tuning RTCP transmissions . . . . . . . . . . . . . . . .  16
       6.2.1.  RTP/AVP and RTP/SAVP  . . . . . . . . . . . . . . . .  16
       6.2.2.  RTP/AVPF and RTP/SAVPF  . . . . . . . . . . . . . . .  18
   7.  Security Considerations . . . . . . . . . . . . . . . . . . .  19
   8.  IANA Considerations . . . . . . . . . . . . . . . . . . . . .  19
   9.  Open Issues . . . . . . . . . . . . . . . . . . . . . . . . .  19
   10. References  . . . . . . . . . . . . . . . . . . . . . . . . .  20
     10.1.  Normative References . . . . . . . . . . . . . . . . . .  20
     10.2.  Informative References . . . . . . . . . . . . . . . . .  20
   Authors' Addresses  . . . . . . . . . . . . . . . . . . . . . . .  22

1.  Introduction




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   At the time the Real-Time Transport Protocol (RTP) [RFC3550] was
   originally designed, and for quite some time after, endpoints in RTP
   sessions typically only transmitted a single media stream, and thus
   used a single synchronization source (SSRC) per RTP session, where
   separate RTP sessions were typically used for each distinct media
   type.  Recently, however, a number of scenarios have emerged in which
   endpoints wish to send multiple RTP media streams, distinguished by
   distinct RTP synchronization source (SSRC) identifiers, in a single
   RTP session.  These are outlined in Section 3.  Although the initial
   design of RTP did consider such scenarios, the specification was not
   consistently written with such use cases in mind.  The specifications
   are thus somewhat unclear.

   This memo updates [RFC3550] to clarify behaviour in use cases where
   endpoints use multiple SSRCs.  It also updates [RFC4585] in regards
   to the timeout of inactive SSRCs to resolve problematic behaviour as
   well as clarifying the inclusion of feedback messages.

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 sends RTP data using multiple SSRCs in
   a single RTP session.

3.1.  Endpoints with Multiple Capture Devices

   The most straightforward motivation for an endpoint to send multiple
   simultaneous RTP streams in a session is the scenario where an
   endpoint has multiple capture devices, and thus media sources, 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], often have multiple cameras or microphones
   covering various areas of a room, and hence send several RTP streams.

3.2.  Multiple Media Types in a Single RTP Session

   Recent work has updated RTP
   [I-D.ietf-avtcore-multi-media-rtp-session] and SDP
   [I-D.ietf-mmusic-sdp-bundle-negotiation] to remove the historical
   assumption in RTP that media sources of different media types would



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   always be sent on different RTP sessions.  In this work, a single
   endpoint's audio and video RTP media streams (for example) are
   instead sent in a single RTP session to reduce the number of
   transport layer flows used.

3.3.  Multiple Stream Mixers

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

   A more complex example is the selective forwarding middlebox,
   described in Section 3.7 of [I-D.ietf-avtcore-rtp-topologies-update].
   This is a middlebox that receives media streams from several
   endpoints, and then selectively forwards modified versions of some
   RTP streams toward the other endpoints to which it is connected.  For
   each connected endpoint, a separate media source appears in the
   session for every other source connected to the middlebox,
   "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.

3.4.  Multiple SSRCs for a Single Media Source

   There are also several cases where a single media source results in
   the usage of multiple SSRCs within the same RTP session.  Transport
   robustness tools like RTP Retransmission [RFC4588] result in multiple
   SSRCs, one with source data, and another with the repair data.
   Scalable encoders and their RTP payload formats, like H.264's
   extension for Scalable Video Coding(SVC) [RFC6190] can be transmitted
   in a configuration where the scalable layers are distributed over
   multiple SSRCs within the same session, to enable RTP packet stream
   level (SSRC) selection and routing in conferencing middleboxes.









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4.  Use of RTP by endpoints that send multiple media streams

   Every RTP endpoint will have an allocated share of the available
   session bandwidth, as determined by signalling and congestion
   control.  The endpoint MUST keep its total media sending rate within
   this share.  However, endpoints that send multiple media streams do
   not necessarily need to subdivide their share of the available
   bandwidth independently or uniformly to each media stream and its
   SSRCs.  In particular, an endpoint can vary the allocation to
   different streams depending on their needs, and can dynamically
   change the bandwidth allocated to different SSRCs (for example, by
   using a variable rate codec), provided the total sending rate does
   not exceed its allocated share.  This includes enabling or disabling
   media streams and their redundancy streams as more or less bandwidth
   becomes available.

5.  Use of RTCP by Endpoints that send multiple media streams

   The RTP Control Protocol (RTCP) is defined in Section 6 of [RFC3550].
   The description of the protocol is phrased in terms of the behaviour
   of "participants" in an RTP session, under the assumption that each
   endpoint is a participant with a single SSRC.  However, for correct
   operation in cases where endpoints can send multiple media streams,
   the specification needs to be interpreted with each SSRC counting as
   a participant in the session, so that an endpoint that has multiple
   SSRCs counts as multiple participants.  The following describes
   several concrete cases where this applies.

5.1.  RTCP Reporting Requirement

   An RTP endpoint that has multiple SSRCs MUST treat each SSRC as a
   separate participant in the RTP session, sending RTCP reports for
   each of its SSRCs in every RTCP reporting interval.  If the mechanism
   in [I-D.ietf-avtcore-rtp-multi-stream-optimisation] is not used, then
   each SSRC will send RTCP reports for all other SSRCs, including those
   co-located at the same endpoint.

   If the endpoint has some SSRCs that are sending data and some that
   are only receivers, then they will receive different shares of the
   RTCP bandwidth and calculate different base RTCP reporting intervals.
   Otherwise, all SSRCs at an endpoint will calculate the same base RTCP
   reporting interval.  The actual reporting intervals for each SSRC are
   randomised in the usual way, but reports can be aggregated as
   described in Section 5.3.

5.2.  Initial Reporting Interval





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   When a participant joins a unicast session, the following text from
   Section 6.2 of [RFC3550] applies: "For unicast sessions... the delay
   before sending the initial compound RTCP packet MAY be zero."  This
   also applies to the individual SSRCs of an endpoint that has multiple
   SSRCs, and such endpoints MAY send an initial RTCP packet for each of
   their SSRCs immediately upon joining a unicast session.

   Caution has to be exercised, however, when an endpoint (or middlebox)
   with a large number of SSRCs joins a unicast session, since immediate
   transmission of many RTCP reports can create a significant burst of
   traffic, leading to transient congestion and packet loss due to queue
   overflows.  Implementers are advised to consider sending immediate
   RTCP packets for only a small number of SSRCs (e.g., the one or two
   SSRCs they consider most important), with the initial RTCP packets
   for their other SSRCs being sent after the calculated initial RTCP
   reporting interval, to avoid self congestion.

   (tbd: is this recommendation sufficiently strong?)

5.3.  Aggregation of Reports into Compound RTCP Packets

   As outlined in Section 5.1, an endpoint with multiple SSRCs has to
   treat each SSRC as a separate participant when it comes to sending
   RTCP reports.  This will lead to each SSRC sending a compound RTCP
   packet in each reporting interval.  Since these packets are coming
   from the same endpoint, it might reasonably be expected that they can
   be aggregated to reduce overheads.  Indeed, Section 6.1 of [RFC3550]
   allows RTP translators and mixers to aggregate packets in similar
   circumstances:

      "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."

   The allows RTP translators and mixers to generate compound RTCP
   packets that contain multiple SR or RR packets from different SSRCs,
   as well as any of the other packet types.  There are no restrictions
   on the order in which the RTCP packets can occur within the compound
   packet, except the regular rule that the compound RTCP packet starts



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   with an SR or RR packet.  Due to this rule, correctly implemented RTP
   endpoints will be able to handle compound RTCP packets that contain
   RTCP packets relating to multiple SSRCs.

   Accordingly, endpoints that use multiple SSRCs MAY aggregate the RTCP
   packets sent by their different SSRCs into compound RTCP packets,
   provided they maintain the average RTCP packet size as described in
   Section 5.3.1, and schedule packet transmission and aggregation as
   described in Section 5.3.2.

5.3.1.  Maintaining AVG_RTCP_SIZE

   The RTCP scheduling algorithm in [RFC3550] works on a per-SSRC basis.
   Each SSRC sends a single compound RTCP packet in each RTCP reporting
   interval.  When an endpoint uses multiple SSRCs, it is desirable to
   aggregate the compound RTCP packets sent by its SSRCs, reducing the
   overhead by forming a larger compound RTCP packet.  This aggregation
   can be done as described in Section 5.3.2, provided the average RTCP
   packet size calculation is updated as follows.

   Participants in an RTP session update their estimate of the average
   RTCP packet size (avg_rtcp_size) each time they send or receive an
   RTCP packet (see Section 6.3.3 of [RFC3550]).  When a compound RTCP
   packet that contains RTCP packets from several SSRCs is sent or
   received, the avg_rtcp_size estimate for each SSRC that is reported
   upon is updated using div_packet_size rather than the actual packet
   size:

      avg_rtcp_size = (1/16) * div_packet_size + (15/16) * avg_rtcp_size


   where div_packet_size is packet_size divided by the number of SSRCs
   reporting in that compound packet.  The number of SSRCs reporting in
   a compound packet is determined by counting the number of different
   SSRCs that are the source of Sender Report (SR) or Receiver Report
   (RR) RTCP packets within the compound RTCP packet.  Non-compound RTCP
   packets (i.e., RTCP packets that do not contain an SR or RR packet
   [RFC5506]) are considered report on a single SSRC.

   An SSRC doesn't follow the above rule, and instead uses the full RTCP
   compound packet size to calculate avg_rtcp_size, will derive an RTCP
   reporting interval that is overly large by a factor that is
   proportional to the number of SSRCs aggregated into compound RTCP
   packets and the size of set of SSRCs being aggregated relative to the
   total number of participants.  This increased RTCP reporting interval
   can cause premature timeouts if it is more than five times the
   interval chosen by the SSRCs that understand compound RTCP that
   aggregate reports from many SSRCs.  A 1500 octet MTU can fit six



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   typical size reports into a compound RTCP packet, so this is a real
   concern if endpoints aggregate RTCP reports from multiple SSRCs.  If
   compatibility with non-updated endpoints is a concern, the number of
   reports from different SSRCs aggregated into a single compound RTCP
   packet SHOULD be limited.

5.3.2.  Scheduling RTCP with Multiple Reporting SSRCs

   When implementing RTCP packet scheduling for cases where multiple
   reporting SSRCs are aggregating their RTCP packets in the same
   compound packet there are a number of challenges.  First of all, we
   have the goal of not changing the general properties of the RTCP
   packet transmissions, which include the general inter-packet
   distribution, and the behaviour for dealing with flash joins as well
   as other dynamic events.

   The below specified mechanism deals with:

   o  That one can't have a-priori knowledge about which RTCP packets
      are to be sent, or their size, prior to generating the packets.
      In which case, the time from generation to transmission ought to
      be as short as possible to minimize the information that becomes
      stale.

   o  That one has an MTU limit, that one ought to avoid exceeding, as
      that requires lower-layer fragmentation (e.g., IP fragmentation)
      which impacts the packets' probability of reaching the
      receiver(s).

   Schedule all the endpoint's local SSRCs individually for transmission
   using the regular calculation of Tn for the profile being used.  Each
   time a SSRC's Tn timer expires, do the regular reconsideration.  If
   the reconsideration indicates that an RTCP packet is to be sent:

   1.  Consider if an additional SSRC can be added.  That consideration
       is done by picking the SSRC which has the Tn value closest in
       time to now (Tc).

   2.  Calculate how much space for RTCP packets would be needed to add
       that SSRC.

   3.  If the considered SSRC's RTCP Packets fit within the lower layer
       datagram's Maximum Transmission Unit, taking the necessary
       protocol headers into account and the consumed space by prior
       SSRCs, then add that SSRC's RTCP packets to the compound packet
       and go again to Step 1.





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   4.  If the considered SSRC's RTCP Packets will not fit within the
       compound packet, then transmit the generated compound packet.

   5.  Update the RTCP Parameters for each SSRC that has been included
       in the sent RTCP packet.  The Tp value for each SSRC MUST be
       updated as follows:

       For the first SSRC:  As this SSRC was the one that was
             reconsidered the tp value is set to the tc as defined in
             RTP [RFC3550].

       For any additional SSRC:  The tp value SHALL be set to the
             transmission time this SSRC would have had it not been
             aggregated and given the current existing session context.
             This value is derived by taking this SSRC's Tn value and
             performing reconsideration and updating tn until tp + T <=
             tn.  Then set tp to this tn value.

   6.  For the sent SSRCs calculate new tn values based on the updated
       parameters and reschedule the timers.

   Reverse reconsideration needs to be performed as specified in RTP
   [RFC3550].  It is important to note that under the above algorithm
   when performing reconsideration, the value of tp can actually be
   larger than tc.  However, that still has the desired effect of
   proportionally pulling the tp value towards tc (as well as tn) as the
   group size shrinks in direct proportion the reduced group size.

   The above algorithm has been shown in simulations to maintain the
   inter-RTCP-packet transmission distribution for the SSRCs and consume
   the same amount of bandwidth as non-aggregated packets in RTP
   sessions with static sets of participants.  With this algorithm the
   actual transmission interval for any SSRC triggering an RTCP compound
   packet transmission is following the regular transmission rules.  It
   also handles the cases where the number of SSRCs that can be included
   in an aggregated packet varies.  An SSRC that previously was
   aggregated and fails to fit in a packet still has its own
   transmission scheduled according to normal rules.  Thus, it will
   trigger a transmission in due time, or the SSRC will be included in
   another aggregate.

   The algorithm's behaviour under SSRC group size changes is under
   investigation.  However, it is expected to be well behaved based on
   the following analyses.

   RTP sessions where the number of SSRC are growing:  When the group
      size is growing, the Td values grow in proportion to the number of
      new SSRCs in the group.  The reconsideration when the timer for



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      the tn expires, that SSRC will reconsider the transmission and
      with a certain probability reschedule the tn timer.  This part of
      the reconsideration algorithm is only impacted by the above
      algorithm by having tp values that are in the future instead of
      set to the time of the actual last transmission at the time of
      updating tp.  Thus the scheduling causes in worst case a plateau
      effect for that SSRC.  That effect depends on how far into the
      future tp can advance.

   RTP sessions where the number of SSRC are shrinking:  When the group
      shrinks, reverse reconsideration moves the tp and tn values
      towards tc proportionally to the number of SSRCs that leave the
      session compared to the total number of participants when they
      left.  Thus the also group size reductions need to be handled.

   In general the potential issue that might exist depends on how far
   into the future the tp value can drift compared to the actual packet
   transmissions that occur.  That drift can only occur for an SSRC that
   never is the trigger for RTCP packet transmission and always gets
   aggregated and where the calculated packet transmission interval
   randomly occurs so that tn - tp for this SSRC is on average larger
   than the ones that gets transmitted.

5.4.  Use of RTP/AVPF Feedback

   This section discusses the transmission of RTP/AVPF feedback packets
   when the transmitting endpoint has multiple SSRCs.

5.4.1.  Choice of SSRC for Feedback Packets

   When an RTP/AVPF endpoint has multiple SSRCs, it can choose what SSRC
   to use as the source for the RTCP feedback packets it sends.  Several
   factors can affect that choice:

   o  RTCP feedback packets relating to a particular media type SHOULD
      be sent by an SSRC that receives that media type.  For example,
      when audio and video are multiplexed onto a single RTP session,
      endpoints will use their audio SSRC to send feedback on the audio
      received from other participants.

   o  RTCP feedback packets and RTCP codec control messages that are
      notifications or indications regarding RTP data processed by an
      endpoint MUST be sent from the SSRC used by that RTP data.  This
      includes notifications that relate to a previously received
      request or command.

   o  If separate SSRCs are used to send and receive media, then the
      corresponding SSRC SHOULD be used for feedback, since they have



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      differing RTCP bandwidth fractions.  This can also effect the
      consideration if the SSRC can be used in immediate mode or not.

   o  Some RTCP feedback packet types requires consistency in the SSRC
      used.  For example, if one sets a TMMBR limitation, the same SSRC
      needs to be used to remove the limitation.

   When an RTCP feedback packet is sent as part of a compound RTCP
   packet that aggregates reports from multiple SSRCs, there is no
   requirement that the compound packet contains an SR or RR packet
   generated by the sender of the RTCP feedback packet.  For reduced-
   size RTCP packets, aggregation of RTCP feedback packets from multiple
   sources is not limited further than Section 4.2.2 of [RFC5506].

5.4.2.  Scheduling an RTCP Feedback Packet

   When an SSRC has a need to transmit a feedback packet in early mode
   it follows the scheduling rules defined in Section 3.5 in RTP/AVPF
   [RFC4585].  When following these rules the following clarifications
   need to be taken into account:

   o  That a session is considered to be point-to-point or multiparty
      not based on the number of SSRCs, but the number of endpoints
      directly seen in the RTP session by the endpoint.  tbd: Clarify
      what is considered to "see" an endpoint?

   o  Note that when checking if there is already a scheduled compound
      RTCP packet containing feedback messages (Step 2 in
      Section 3.5.2), that check is done considering all local SSRCs.

   TBD: The above does not allow an SSRC that is unable to send either
   an early or regular RTCP packet with the feedback message within the
   T_max_fb_delay to trigger another SSRC to send an early packet to
   which it could piggyback.  Nor does it allow feedback to piggyback on
   even regular RTCP packet transmissions that occur within
   T_max_fb_delay.  A question is if either of these behaviours ought to
   be allowed.  The latter appears simple and straight forward.  Instead
   of discarding a FB message in step 4a: alternative 2, one could place
   such messages in a cache with a discard time equal to T_max_fb_delay,
   and in case any of the SSRCs schedule an RTCP packet for transmission
   within that time, it includes this message.  The former case can have
   more widespread impact on the application, and possibly also on the
   RTCP bandwidth consumption as it allows for more massive bursts of
   RTCP packets.  Still, on a time scale of a regular reporting
   interval, it ought to have no effect on the RTCP bandwidth as the
   extra feedback messages increase the avg_rtcp_size.





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6.  RTCP Considerations for Streams with Disparate Rates

   It is possible for a single RTP session to carry streams of greatly
   differing bandwidth.  There are two scenarios where this can occur.
   The first is when a single RTP session carries multiple flows of the
   same media type, but with very different quality; for example a video
   switching multi-point conference unit might send a full rate high-
   definition video stream of the active speaker but only thumbnails for
   the other participants, all sent in a single RTP session.  The second
   scenarios occurs when audio and video flows are sent in a single RTP
   session, as discussed in [I-D.ietf-avtcore-multi-media-rtp-session].

   An RTP session has a single set of parameters that configure the
   session bandwidth, the RTCP sender and receiver fractions (e.g., via
   the SDP "b=RR:" and "b=RS:" lines), and the parameters of the RTP/
   AVPF profile [RFC4585] (e.g., trr-int) if that profile (or its secure
   extension, RTP/SAVPF [RFC5124]) is used.  As a consequence, the RTCP
   reporting interval will be the same for every SSRC in an RTP session.
   This uniform RTCP reporting interval can result in RTCP reports being
   sent more often than is considered desirable for a particular media
   type.  For example, if an audio flow is multiplexed with a high
   quality video flow where the session bandwidth is configured to match
   the video bandwidth, this can result in the RTCP packets having a
   greater bandwidth allocation than the audio data rate.  If the
   reduced minimum RTCP interval described in Section 6.2 of [RFC3550]
   is used in the session, which might be appropriate for video where
   rapid feedback is wanted, the audio sources could be expected to send
   RTCP packets more often than they send audio data packets.  This is
   most likely undesirable, and while the mismatch can be reduced
   through careful tuning of the RTCP parameters, particularly trr_int
   in RTP/AVPF sessions, it is inherent in the design of the RTCP timing
   rules, and affects all RTP sessions containing flows with mismatched
   bandwidth.

   Having multiple media types in one RTP session also results in more
   SSRCs being present in this RTP session.  This increasing the amount
   of cross reporting between the SSRCs.  From an RTCP perspective, two
   RTP sessions with half the number of SSRCs in each will be slightly
   more efficient.  If someone needs either the higher efficiency due to
   the lesser number of SSRCs or the fact that one can't tailor RTCP
   usage per media type, they need to use independent RTP sessions.










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   When it comes to configuring RTCP the need for regular periodic
   reporting needs to be weighted against any feedback or control
   messages being sent.  Applications using RTP/AVPF or RTP/SAVPF are
   RECOMMENDED to consider setting the trr-int parameter to a value
   suitable for the application's needs, thus potentially reducing the
   need for regular reporting and thus releasing more bandwidth for use
   for feedback or control.

   Another aspect of an RTP session with multiple media types is that
   the RTCP packets, RTCP Feedback Messages, or RTCP XR metrics used
   might not be applicable to all media types.  Instead, all RTP/RTCP
   endpoints need to correlate the media type of the SSRC being
   referenced in a message or packet and only use those that apply to
   that particular SSRC and its media type.  Signalling solutions might
   have shortcomings when it comes to indicating that a particular set
   of RTCP reports or feedback messages only apply to a particular media
   type within an RTP session.

6.1.  Timing out SSRCs

   This section discusses issues around timing out SSRCs.  After the
   discussion, clarified and mandated behaviour for SSRC timeout is
   specified.

6.1.1.  AVPF T_rr_interval Behaviour

   The RTP/AVPF profile includes a mechanism for suppressing regular
   RTCP reporting from being sent unnecessarily frequently if sufficient
   RTCP bandwidth is configured.  This mechanism is defined in
   Section 3.5.3 of [RFC4585], and can be summarized as follows: if less
   than a randomized T_rr_interval value has passed since the last
   regular report, and no feedback messages need to be sent, then the
   RTCP regular report is suppressed.  The randomization is done
   linearly in the interval 0.5 to 1.5 times T_rr_interval.  The
   randomized T_rr_interval is recalculated after every transmitted
   regular packet, i.e when t_rr_last was updated.  The benefit of the
   suppression mechanism is that it avoids wasting bandwidth when there
   is nothing requiring frequent RTCP transmissions, but still allows
   utilization of the configured bandwidth when feedback is needed.

   Unfortunately this suppression mechanism has some behaviour that is
   less than ideal.  First of all, the randomized T_rr_interval is
   distributed over a larger range than the actual transmission interval
   for RTCP would be if T_rr_interval and Td had the same value.  The
   reconsideration mechanism and its compensation factor result in the
   actual RTCP transmission intervals for a Td having a distribution
   that is exponentially growing more likely with higher values, and is
   bounded to the interval [0.5/1.21828, 1.5/1.21828]*Td, i.e.  with a



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   Td value of 5 s [2.052, 6.156].  In comparison, the suppression acts
   in an interval that is 0.5 to 1.5 times the T_rr_interval, i.e.  for
   T_rr_interval = 5 s this is [2.5, 7.5].

   The effect of the above is that the time period between two RTCP
   packets when using T_rr_interval suppression can become very long
   compared to the average input values.  The longest time interval
   between one transmitted regular RTCP compound packet and the next
   when T_rr_interval suppression is being used are: First the maximum
   T_rr_interval, i.e.  1.5*T_rr_interval.  Assuming that the last
   suppressed packet would have been sent at 1.5*T_rr_interval, the
   maximum interval until a packet can be sent under the regular
   scheduling is 1.5/1.21828*Td.  Thus, the maximum time in total is
   1.5*T_rr_interval + 1.5/1.21828*Td.

   If Td and T_rr_interval have the same value, i.e.  the minimal
   interval desired (T_rr_interval) and the actual actual average
   interval specified by the RTCP scheduling algorithm (Td) are the
   same, one might expect that RTCP packets would be sent according to
   the regular mechanism.  Instead, this algorithm results in the RTCP
   packets being sent anywhere from 0.5*Td to ~2.731*Td.  The
   probability distribution over that time is also non-trivial in its
   shape, somewhat similar to a saw tooth.

   Thus, we recommend that the AVPF regular transmission mechanism is
   revised in the future.  This issue also has further implications as
   discussed in the next section.

6.1.2.  Avoiding Premature Timeout

   In RTP/AVP [RFC3550] the timeout behaviour is simple and is 5 times
   Td, where Td is calculated with a Tmin value of 5 seconds.  In other
   words, if the RTCP bandwidth allowed for an RTCP interval more
   frequent than every 5 seconds on average, then timeout happened after
   5*Td = 25 seconds of no activity from the SSRC (RTP or RTCP),
   otherwise it was 5 average reporting intervals.

   RTP/AVPF [RFC4585] introduced two different behaviours depending on
   the value of T_rr_interval.  When T_rr_interval was 0, it defaulted
   to the same Td calculation in RTP/AVP [RFC3550].  However, when
   T_rr_interval is non-zero the Tmin value become T_rr_interval in that
   calculation, most likely to enable speed up the detection of timed
   out SSRCs.  However, using a non-zero T_rr_interval has two
   consequences for RTP behaviour.

   First, the number of actually sent RTCP packets for an SSRC that
   currently is not an active RTP sender can become very low due to the
   issue discussed above in Section 6.1.1.  As the RTCP packet interval



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   can be as long as 2.73*Td, then during a 5*Td time period an endpoint
   may in fact transmit only a single RTCP packet.  The long intervals
   result in fewer RTCP packets, to a point where a one or two packet
   losses in RTCP result in timing out an SSRC.

   Second, the change also increased RTP/AVPF's brittleness to both
   packet loss and configuration errors.  In many cases, when one
   desires to use RTP/AVPF for its feedback, one will ensure that RTCP
   is configured for more frequent transmissions on average than every 5
   seconds.  Thus, many more RTP and RTCP packets can be transmitted
   during the time interval.  Lets consider an implementation that would
   follow the RTP/AVP or RTP/AVPF with T_rr_interval = 0 rules for
   timeout, also when T_rr_interval is not zero.  In such a case when
   the configured value of the T_rr_interval is significantly smaller
   than 5 seconds, e.g.  less than 1 second, then a difference between
   using 0.1 seconds and 0.6 seconds has no significant impact on when
   an SSRC will be timed out.  However, such a configuration difference
   between two endpoints following RFC 4585 will result in that the
   endpoint configured with T_rr_interval = 0.1 will frequently timeout
   SSRCs currently not sending RTP, from the endpoint configured with
   0.6, as that is six times the Td value used by the endpoint
   configured with T_rr_interval=0.1, assuming sufficient bandwidth.
   For this reason such a change is implemented below in Section 6.1.4.

6.1.3.  RTP/AVP and RTP/AVPF Interoperability

   If endpoints implementing the RTP/AVP and RTP/AVPF profiles (or their
   secure variants) are combined in a single RTP session, and the RTP/
   AVPF endpoints use a non-zero T_rr_interval that is significantly
   lower than 5 seconds, then there is a risk that the RTP/AVPF
   endpoints will prematurely timeout the RTP/AVP SSRCs due to their
   different RTCP timeout intervals.  Conversely, if the RTP/AVPF
   endpoints use a T_rr_interval that is significant larger than 5
   seconds, there is a risk that the RTP/AVP endpoints will timeout the
   RTP/AVPF SSRCs.

   If such mixed RTP profiles are used, (though this is NOT
   RECOMMENDED), and the AVPF endpoint is not updated to follow this
   specification, then the RTP/AVPF session SHOULD use a non-zero
   T_rr_interval that is 4 seconds.

   It might appear strange to use a T_rr_interval of 4 seconds.  It
   might be intuitive that this value ought to be 5 seconds, as then
   both the RTP/AVP and RTP/AVPF would use the same timeout period.
   However, considering regular RTCP transmission and their packet
   intervals for RTP/AVPF its mean value will (with non-zero
   T_rr_interval) be larger than T_rr_interval due to the scheduling
   algorithm's behaviour as discussed in Section 6.1.1.  Thus, to enable



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   an equal amount of regular RTCP transmissions in each directions
   between RTP/AVP and RTP/AVPF endpoints, taking the altered timeout
   intervals into account, the optimal value is around four (4), where
   almost four transmissions will on average occur in each direction
   between the different profile types given an otherwise good
   configuration of parameters in regards to T_rr_interval.  If the RTCP
   bandwidth parameters are selected so that Td based on bandwidth is
   close to 4, i.e.  close to T_rr_interval the risk increases that RTP/
   AVPF SSRCs will be timed out by RTP/AVP endpoints, as the RTP/AVPF
   SSRC might only manage two transmissions in the timeout period.

6.1.4.  Specified Behaviour

   The above considerations result in the following clarification and
   RTP/AVPF specification change.

   All SSRCs used in an RTP session MUST use the same timeout behaviour
   to avoid premature timeouts.  This will depend on the RTP profile and
   its configuration.  The RTP specification provides several options
   that can influence the values used when calculating the time
   interval.  To avoid interoperability issues when using this
   specification, this document makes several clarifications to the
   calculations.

   For RTP/AVP, RTP/SAVP, RTP/AVPF, and RTP/SAVPF, the timeout interval
   SHALL be calculated using a multiplier of 5, i.e.  the timeout
   interval becomes 5*Td.  The Td calculation SHALL be done using a Tmin
   value of 5 seconds, not the reduced minimal interval even if used to
   calculate RTCP packet transmission intervals.  This changes the
   behaviour for the RTP/AVPF or RTP/SAVPF profiles when T_rr_interval
   != 0, a behaviour defined in Section 3.5.4 of RFC 4585, i.e.  Tmin in
   the Td calculation is the T_rr_interval.

6.2.  Tuning RTCP transmissions

   This sub-section discusses what tuning can be done to reduce the
   downsides of the shared RTCP packet intervals.  First, it is
   considered what possibilities exist for the RTP/AVP [RFC3551]
   profile, then what additional tools are provided by RTP/AVPF
   [RFC4585].

6.2.1.  RTP/AVP and RTP/SAVP

   When using the RTP/AVP or RTP/SAVP profiles the tuning one can do is
   very limited.  The controls one has are limited to the RTCP bandwidth
   values and whether the minimum RTCP interval is scaled according to
   the bandwidth.  As the scheduling algorithm includes both random
   factors and reconsideration, one can't simply calculate the expected



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   average transmission interval using the formula for Td.  But it does
   indicate the important factors affecting the transmission interval,
   namely the RTCP bandwidth available for the role (Active Sender or
   Participant), the average RTCP packet size, and the number of SSRCs
   classified in the relevant role.  Note that if the ratio of senders
   to total number of session participants is larger than the ratio of
   RTCP bandwidth for senders in relation to the total RTCP bandwidth,
   then senders and receivers are treated together.

   Let's start with some basic observations:

   a.  Unless the scaled minimum RTCP interval is used, then Td prior to
       randomization and reconsideration can never be less than 5
       seconds (assuming default Tmin of 5 seconds).

   b.  If the scaled minimum RTCP interval is used, Td can become as low
       as 360 divided by RTP Session bandwidth in kilobits.  In SDP the
       RTP session bandwidth is signalled using b=AS.  An RTP Session
       bandwidth of 72 kbps results in Tmin being 5 seconds.  An RTP
       session bandwidth of 360 kbps of course gives a Tmin of 1 second,
       and to achieve a Tmin equal to once every frame for a 25 Hz video
       stream requires an RTP session bandwidth of 9 Mbps!  (The use of
       the RTP/AVPF or RTP/SAVPF profile allows a smaller Tmin, and
       hence more frequent RTCP reports, as discussed below).

   c.  Let's calculate the number (n) of SSRCs in the RTP session that
       5% of the session bandwidth can support to yield a Td value equal
       to Tmin with minimal scaling.  For this calculation we have to
       make two assumptions.  The first is that we will consider most or
       all SSRC being senders, resulting in everyone sharing the
       available bandwidth.  Secondly we will select an average RTCP
       packet size.  This packet will consist of an SR, containing (n-1)
       report blocks up to 31 report blocks, and an SDES item with at
       least a CNAME (17 bytes in size) in it.  Such a basic packet will
       be 800 bytes for n>=32.  With these parameters, and as the
       bandwidth goes up the time interval is proportionally decreased
       (due to minimal scaling), thus all the example bandwidths 72
       kbps, 360 kbps and 9 Mbps all support 9 SSRCs.

   d.  The actual transmission interval for a Td value is [0.5*Td/
       1.21828,1.5*Td/1.21828], which means that for Td = 5 seconds, the
       interval is actually [2.052,6.156] and the distribution is not
       uniform, but rather exponentially-increasing.  The probability
       for sending at time X, given it is within the interval, is
       probability of picking X in the interval times the probability to
       randomly picking a number that is <=X within the interval with an
       uniform probability distribution.  This results in that the
       majority of the probability mass is above the Td value.



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   To conclude, with RTP/AVP and RTP/SAVP the key limitation for small
   unicast sessions is going to be the Tmin value.  Thus the RTP session
   bandwidth configured in RTCP has to be sufficiently high to reach the
   reporting goals the application has following the rules for the
   scaled minimal RTCP interval.

6.2.2.  RTP/AVPF and RTP/SAVPF

   When using RTP/AVPF or RTP/SAVPF we get a quite powerful additional
   tool, the setting of the T_rr_interval which has several effects on
   the RTCP reporting.  First of all as Tmin is set to 0 after the
   initial transmission, the regular reporting interval is instead
   determined by the regular bandwidth based calculation and the
   T_rr_interval.  This has the effect that we are no longer restricted
   by the minimal interval or even the scaling rule for the minimal
   rule.  Instead the RTCP bandwidth and the T_rr_interval are the
   governing factors.

   Now it also becomes important to separate between the application's
   need for regular reports and RTCP feedback packet types.  In both
   regular RTCP mode, as in Early RTCP Mode, the usage of the
   T_rr_interval prevents regular RTCP packets, i.e.  packets without
   any Feedback packets, to be sent more often than T_rr_interval.  This
   value is applied to prevent any regular RTCP packet to be sent less
   than T_rr_interval times a uniformly distributed random value from
   the interval [0.5,1.5] after the previous regular packet packet.  The
   random value recalculated after each regular RTCP packet
   transmission.

   So applications that have a use for feedback packets for some media
   streams, for example video streams, but don't want frequent regular
   reporting for audio, could configure the T_rr_interval to a value so
   that the regular reporting for both audio and video is at a level
   that is considered acceptable for the audio.  They could then use
   feedback packets, which will include RTCP SR/RR packets, unless
   reduced-size RTCP feedback packets [RFC5506] are used, and can
   include other report information in addition to the feedback packet
   that needs to be sent.  That way the available RTCP bandwidth can be
   focused for the use which provides the most utility for the
   application.

   Using T_rr_interval still requires one to determine suitable values
   for the RTCP bandwidth value, in fact it might make it even more
   important, as this is more likely to affect the RTCP behaviour and
   performance than when using RTP/AVP, as there are fewer limitations
   affecting the RTCP transmission.





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   When using T_rr_interval, i.e.  having it be non zero, there are
   configurations that have to be avoided.  If the resulting Td value is
   smaller but close to T_rr_interval then the interval in which the
   actual regular RTCP packet transmission falls into becomes very
   large, from 0.5 times T_rr_interval up to 2.73 times the
   T_rr_interval.  Therefore for configuration where one intends to have
   Td smaller than T_rr_interval, then Td is RECOMMENDED to be targeted
   at values less than 1/4th of T_rr_interval which results in that the
   range becomes [0.5*T_rr_interval, 1.81*T_rr_interval].

   With RTP/AVPF, using a T_rr_interval of 0 or with another low value
   significantly lower than Td still has utility, and different
   behaviour compared to RTP/AVP.  This avoids the Tmin limitations of
   RTP/AVP, thus allowing more frequent regular RTCP reporting.  In fact
   this will result that the RTCP traffic becomes as high as the
   configured values.

   (tbd: a future version of this memo will include examples of how to
   choose RTCP parameters for common scenarios)

   There exists no method within the specification for using different
   regular RTCP reporting intervals depending on the media type or
   individual media stream.

7.  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 allow use of 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.

8.  IANA Considerations

   No IANA actions needed.

9.  Open Issues

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






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   1.  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?

   2.  RTCP parameters for common scenarios in Section 6.2?

   3.  Is scheduling algorithm working well with dynamic changes?

   4.  Are the scheduling algorithm changes impacting previous
       implementations in such a way that the report aggregation has to
       be agreed on, and thus needs to be considered as an optimization?

   5.  An open question is if any improvements or clarifications ought
       to be allowed regarding FB message scheduling in multi-SSRC
       endpoints.

10.  References

10.1.  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.

10.2.  Informative References

   [I-D.ietf-avtcore-multi-media-rtp-session]




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              Westerlund, M., Perkins, C., and J. Lennox, "Sending
              Multiple Types of Media in a Single RTP Session", draft-
              ietf-avtcore-multi-media-rtp-session-05 (work in
              progress), February 2014.

   [I-D.ietf-avtcore-rtp-multi-stream-optimisation]
              Lennox, J., Westerlund, M., Wu, Q., and C. Perkins,
              "Sending Multiple Media Streams in a Single RTP Session:
              Grouping RTCP Reception Statistics and Other Feedback ",
              draft-ietf-avtcore-rtp-multi-stream-optimisation-00 (work
              in progress), July 2013.

   [I-D.ietf-avtcore-rtp-topologies-update]
              Westerlund, M. and S. Wenger, "RTP Topologies", draft-
              ietf-avtcore-rtp-topologies-update-02 (work in progress),
              May 2014.

   [I-D.ietf-clue-framework]
              Duckworth, M., Pepperell, A., and S. Wenger, "Framework
              for Telepresence Multi-Streams", draft-ietf-clue-
              framework-16 (work in progress), June 2014.

   [I-D.ietf-mmusic-sdp-bundle-negotiation]
              Holmberg, C., Alvestrand, H., and C. Jennings,
              "Negotiating Media Multiplexing Using the Session
              Description Protocol (SDP)", draft-ietf-mmusic-sdp-bundle-
              negotiation-07 (work in progress), April 2014.

   [RFC3551]  Schulzrinne, H. and S. Casner, "RTP Profile for Audio and
              Video Conferences with Minimal Control", STD 65, RFC 3551,
              July 2003.

   [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.

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

   [RFC6190]  Wenger, S., Wang, Y.-K., Schierl, T., and A.
              Eleftheriadis, "RTP Payload Format for Scalable Video
              Coding", RFC 6190, May 2011.




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Authors' Addresses

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

   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


   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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