AVTCORE                                                        J. Lennox
Internet-Draft                                                     Vidyo
Updates: 3550, 4585 (if approved)                          M. Westerlund
Intended status: Standards Track                                Ericsson
Expires: September 10, 2015                                        Q. Wu
                                                              C. Perkins
                                                   University of Glasgow
                                                           March 9, 2015

         Sending Multiple Media Streams in a Single RTP Session


   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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   material or to cite them other than as "work in progress."

   This Internet-Draft will expire on September 10, 2015.

Copyright Notice

   Copyright (c) 2015 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
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   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.  Endpoints with Multiple Capture Devices . . . . . . . . .   3
     3.2.  Multiple Media Types in a Single RTP Session  . . . . . .   4
     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  . . . . . . . . . . . . . . .   6
     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  . . . . . . . . . . . . . . . .  11
       5.4.1.  Choice of SSRC for Feedback Packets . . . . . . . . .  11
       5.4.2.  Scheduling an RTCP Feedback Packet  . . . . . . . . .  12
   6.  RTCP Considerations for Streams with Disparate Rates  . . . .  14
     6.1.  Timing out SSRCs  . . . . . . . . . . . . . . . . . . . .  16
       6.1.1.  Problems with RTP/AVPF the T_rr_interval Parameter  .  16
       6.1.2.  Avoiding Premature Timeout  . . . . . . . . . . . . .  17
       6.1.3.  Interoperability Between RTP/AVP and RTP/AVPF . . . .  18
       6.1.4.  Updated SSRC Timeout Rules  . . . . . . . . . . . . .  18
     6.2.  Tuning RTCP transmissions . . . . . . . . . . . . . . . .  19
       6.2.1.  RTP/AVP and RTP/SAVP  . . . . . . . . . . . . . . . .  19
       6.2.2.  RTP/AVPF and RTP/SAVPF  . . . . . . . . . . . . . . .  20
   7.  Security Considerations . . . . . . . . . . . . . . . . . . .  22
   8.  IANA Considerations . . . . . . . . . . . . . . . . . . . . .  22
   9.  References  . . . . . . . . . . . . . . . . . . . . . . . . .  22
     9.1.  Normative References  . . . . . . . . . . . . . . . . . .  22
     9.2.  Informative References  . . . . . . . . . . . . . . . . .  23
   Authors' Addresses  . . . . . . . . . . . . . . . . . . . . . . .  24

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1.  Introduction

   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",
   "OPTIONAL" in this document are to be interpreted as described in RFC
   2119 [RFC2119] and indicate requirement levels for compliant

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.

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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
   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.

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5.2.  Initial Reporting Interval

   When a participant joins a unicast session, the following text from
   Section 6.2 of [RFC3550] is relevant: "For unicast sessions... the
   delay before sending the initial compound RTCP packet MAY be zero."
   The basic assumption is that this also ought to apply in the case of
   multiple SSRCs.  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.

   To ensure that the initial burst of traffic generated by an RTP
   endpoint is no larger than would be generated by a TCP connection, an
   RTP endpoint MUST NOT send more than four compound RTCP packets with
   zero initial delay when it joins a session.  Each of those initial
   compound RTCP packets MAY include aggregated reports from multiple
   SSRCs, provided the total compound RTCP packet size does not exceed
   the MTU, and the avg_rtcp_packet_size is maintained as in
   Section 5.3.1.  Aggregating reports from several SSRCs in the initial
   compound RTCP packets allows a substantial number of SSRCs to report
   immediately.  Endpoints SHOULD prioritize reports on SSRCs that are
   likely to be most immediately useful, e.g., for SSRCs that are
   initially senders.

   An endpoint that needs to report on more SSRCs than will fit into the
   four compound RTCP reports that can be sent immediately MUST send the
   other reports later, following the usual RTCP timing rules including
   timer reconsideration.  Those reports MAY be aggregated as described
   in Section 5.3.

      Note: The above is based on an TCP initial window of 4 packets,
      not the larger initial windows which there is an ongoing
      experiment with.  The reason for this is a desire to be
      conservative as an RTP endpoint will also in many cases commence
      RTP transmission at the same time as these initial RTCP packets
      are sent.

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

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      "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."

   This 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
   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 1) the resulting compound RTCP packets begin with an SR or
   RR packet; 2) they maintain the average RTCP packet size as described
   in Section 5.3.1; and 3) they schedule packet transmission and manage
   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

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

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   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 to report on a single SSRC.

   An SSRC that 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 five
   typical size reports into a compound RTCP packet, so this is a real
   concern if endpoints aggregate RTCP reports from multiple SSRCs.

   The issue raised in the previous paragraph is mitigated by the
   modification in timeout behaviour specified in Section 6.1.2.  This
   mitigation is in place in those cases where the RTCP bandwidth is
   sufficiently high that an endpoint, using an avg_rtcp_size calculated
   without taking into account the number of reporting SSRCs, can
   transmit more frequently than approximately every 5 seconds.  Note,
   however, that the non-modified endpoint's RTCP reporting is still
   negatively impacted even if the premature timeout of its SSRCs are
   avoided.  If compatibility with non-updated endpoints is a concern,
   the number of reports from different SSRCs aggregated into a single
   compound RTCP packet SHOULD either be limited to two reports, or
   aggregation ought not used at all.  This will limit the non-updated
   endpoint's RTCP reporting interval to be no larger than twice the
   RTCP reporting interval that would be chosen by an endpoint following
   this specification.

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:

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   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

   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

   The below text modifies and extends the behavior defined in
   Section 6.3 of [RFC3550], and in Section 3.5.3 of [RFC4585] if the
   AVPF or SAVPF profile is used, regarding actions to take when
   scheduling and sending an RTCP packet.  It uses the variable names
   tn, tp, tc, T and Td defined in Section 6.3 of [RFC3550].  The
   variable T_rr_last is defined in [RFC4585].

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

   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 the current time (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 and the space consumed by prior SSRCs into
       account, then add that SSRC's RTCP packets to the compound packet
       and go again to Step 1.

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

   5.  Update the RTCP Parameters for each SSRC that has been included
       in the sent RTCP packet.  The previous RTCP transmit time (tp)
       value for each SSRC MUST be updated as follows:

       A.  For the first SSRC set the transmission time (tt) to tc.

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       B.  For any additional SSRC calculate the transmission time that
           each of these SSRCs 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
           tt = tn.  If AVPF or SAVPF is being used, then T_rr_int based
           suppression MUST NOT be used in this calcualtion.

       C.  Calculate average transmission time (tt_avg) using the tt of
           all the SSRCs included in the packet.

       D.  Now update tp for all the sent SSRCs to tt_avg.

       E.  If AVPF or SAVPF profile is being used update T_rr_last to

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

   When using AVPF or SAVPF profile, when following the scheduling
   algorithm for regular transmission in Section 3.5.3 then the case of
   T_rr_interval == 0, as well as option 1, 2a and 2b for T_rr_interval
   != 0, results in transmission of a regular RTCP packet that follows
   the above and updates the necessary variables.  However, when the
   transmission is suppressed per 2c, then tp is updated to tc, as no
   aggregation has taken place.

   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 this algorithm the actual transmission interval for
   any SSRC triggering an RTCP compound packet transmission is following
   the regular transmission rules.  The value tp is set to somewhere in
   the interval [0,1.5/1.21828*Td] ahead of tc.  The actual value is
   average of one instance of tc and the randomized transmission times
   of the additional SSRCs, thus the lower range of the interval is more
   probable.  This setting is performed to compensate for the bias that
   is otherwise introduced by picking the shortest tn value out of the N
   SSRCs included in aggregate.

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   The algorithm 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 as follows:

   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.  When reconsideration is done when the
      timer for 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 were in
      the future instead of set to the time of the actual last
      transmission at the time of updating tp.

   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.  The setting of the tp value forward in time related to the
      tc could be believed to have negative effect.  However, the reason
      for this setting is to compensate for bias caused by picking the
      shortest tn out of the N aggregated.  This bias remains over a
      reduction in the number of SSRCs.  The reverse reconsideration
      compensates the reduction independently of aggregation being used
      or not.  The negative effect that can occur on removing an SSRC is
      that the most favourable tn belonged to the removed SSRC.  The
      impact of this is limited to delaying the transmission, in the
      worst case, one reporting interval.

   In conclusion the investigations performed has found no significant
   negative impact on the scheduling algorithm.

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:

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   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 [RFC4585][RFC5104].

   o  If separate SSRCs are used to send and receive media, then the
      corresponding SSRC SHOULD be used for feedback, since they have
      differing RTCP bandwidth fractions.  This can also affect the
      consideration if the SSRC can be used in immediate mode or not.

   o  Some RTCP feedback packet types require consistency in the SSRC
      used.  For example, if a TMMBR limitation [RFC5104] is set by an
      SSRC, the same SSRC needs to be used to remove the limitation.

   o  If several SSRCs are suitable for sending feedback, if might be
      desirable to use an SSRC that allows the sending of feedback as an
      early RTCP packet.

   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  Whether a session is considered to be point-to-point or multiparty
      is not based on the number of SSRCs, but the number of endpoints
      one directly interacts with in the RTP session.  This is
      determined by counting the number of CNAMEs used by the SSRCs
      received.  A RTP session MUST be considered multiparty if more
      than one CNAME is received, unless signalling explicitly indicates
      that the session is to be handled as point to point, or RTCP
      reporting groups [I-D.ietf-avtcore-rtp-multi-stream-optimisation]
      are used.  If RTCP reporting groups are used, the classification

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      is solely based on whether the endpoint receives a single
      reporting group, indicating point to point, or if multiple
      reporting groups are received (or a mixture of sources using and
      sources not using reporting groups), which is classified as
      multiparty.  Note that contributing sources (CSRCs) can be bound
      to any number of different CNAMEs and do not affect the
      determination of whether the session is multiparty.  Similarly,
      SSRC/CSRC values that are only seen in the source field of an SDES
      packet do not affect this determination.

   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.

   o  If the SSRC is not allowed to send an early RTCP packet, then the
      feedback message MAY be queued for transmission as part of any
      early or regular scheduled transmission that can occur within the
      maximum useful lifetime of the feedback message (T_max_fb_delay).
      This modifies the behaviour in bullet 4a) in Section 3.5.2 of

   The above rule for determining if a RTP session is to be considered
   point-to-point or multiparty is simple and straightforward and works
   in most cases.  The goal with the above classification is to
   determine if the resources associated with RTP and RTCP are shared
   with only one peer or multiple other endpoints.  This is significant
   as it affects the impact and the necessary processing and resource
   consumption.  Relying on only CNAME will result in classifying some
   few situations where one might actually have only one peer as a
   multiparty situation.  The known situations are the following ones:

   Endpoint with multiple synchronization contexts:  An endpoint that is
      part of a point-to-point session can have multiple synchronization
      contexts, for example due to forwarding an external media source
      into a interactive real-time conversation.  In this case the
      classification will consider the peer as two endpoints, while the
      actual RTP/RTCP transmission will be under the control of one

   Selective Forwarding Middlebox:  The SFM as defined in Section 3.7 of
      [I-D.ietf-avtcore-rtp-topologies-update] has control over the
      transmission and configurations between itself and each peer
      endpoint individually.  It also fully controls the RTCP packets
      being forwarded between the individual legs.  Thus, this type of
      middlebox can be compared to the RTP mixer, which uses its own
      SSRCs to mix or select the media it forwards, that will be
      classified as a point-to-point RTP session by the above rule.

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   In the above cases it is very reasonable to use RTCP reporting groups
   [I-D.ietf-avtcore-rtp-multi-stream-optimisation].  If that extension
   is used, an endpoint can indicate that the multitude of CNAMEs are in
   fact under a single endpoint or middlebox control by using only a
   single reporting group.

   The above rules will also classify some sessions where the endpoint
   is connected to an RTP mixer as being point to point.  For example
   the mixer could act as gateway to an Any Source Multicast based RTP
   session for the discussed endpoint.  However, this will in most cases
   be okay, as the RTP mixer provides separation between the two parts
   of the session.  The responsibility falls on the mixer to act
   accordingly in each domain.

   Note: The above usage of point-to-point or multiparty as classifiers
   is actually misleading, but we maintain these labels to match what is
   used in [RFC4585] as this ensures that the right algorithms are

   To conclude we note that in some cases signalling can be used to
   override the rule when it would result in the wrong classification.

6.  RTCP Considerations for Streams with Disparate Rates

   An RTP session has a single set of parameters that configure the
   session bandwidth.  These are the RTCP sender and receiver fractions
   (e.g., 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 base RTCP reporting interval, before randomisation, will be the
   same for every sending SSRC in an RTP session.  Similarly, every
   receiving SSRC in an RTP session will have the same base reporting
   interval, although this can differ from the reporting interval chosen
   by sending SSRCs.  This uniform RTCP reporting interval for all SSRCs
   can result in RTCP reports being sent more often, or too seldom, than
   is considered desirable for a RTP stream.

   For example, consider a scenario when an audio flow sending at tens
   of kilobits per second is multiplexed into an RTP session with a
   multi-megabit high quality video flow.  If the session bandwidth is
   configured based on the video sending rate, and the default RTCP
   bandwidth fraction of 5% of the session bandwidth is used, it is
   likely that the RTCP bandwidth will exceed the audio sending rate.
   If the reduced minimum RTCP interval described in Section 6.2 of
   [RFC3550] is then used in the session, as appropriate for video where
   rapid feedback on damaged I-frames is wanted, the uniform reporting
   interval for all senders could mean that audio sources are expected
   to send RTCP packets more often than they send audio data packets.

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   This bandwidth mismatch can be reduced by careful tuning of the RTCP
   parameters, especially trr_int when the RTP/AVPF profile is used,
   cannot be avoided entirely, as it is inherent in the design of the
   RTCP timing rules, and affects all RTP sessions that contain flows
   with greatly mismatched bandwidth.

   Different media rates or desired RTCP behaviours can also occur
   between SSRCs carrying the same media type.  A common case in
   multiparty conferencing is when only one or two video source are
   shown in higher resolution, while the others are shown as small
   thumbnails, with the choice of which is shown in high resolution
   being voice activity controlled.  Here the differences are both in
   actual media rate and in choices for what feedback messages might be
   needed.  Other examples of differences that can exist are due to the
   intended usage of a media source.  A media source carrying the video
   of the speaker in a conference is different from a document camera.
   Basic parameters that can differ in this case are frame-rate,
   acceptable end-to-end delay, and the SNR fidelity of the image.
   These differences affect not only the needed bit-rates, but also
   possible transmission behaviours, usable repair mechanisms, what
   feedback messages the control and repair requires, the transmission
   requirements on those feedback messages, and monitoring of the RTP
   stream delivery.

   Sending multiple media types in a single RTP session causes that
   session to contain more SSRCs than if each media type was sent in a
   separate RTP session.  For example, if two participants each send an
   audio and a video flow in a single RTP session, that session will
   comprise four SSRCs, but if separate RTP sessions had been used for
   audio and video, each of those two RTP sessions would comprise only
   two SSRCs.  Sending multiple media streams in an RTP session hence
   increases the amount of cross reporting between the SSRCs, as each
   SSRC reports on all other SSRCs in the session.  This increases the
   size of the RTCP reports, causing them to be sent less often than
   would be the case if separate RTP sessions where used for a given
   RTCP bandwidth.

   Finally, when an RTP session contains multiple media types, it is
   important to note that the RTCP reception quality reports, feedback
   messages, and extended report blocks used might not be applicable to
   all media types.  Endpoints will need to consider the media type of
   each SSRC only send or process reports and feedback 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.

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   From an RTCP perspective, therefore, it can be seen that there are
   advantages to using separate RTP sessions for each media stream,
   rather than sending multiple media streams in a single RTP session.
   However, these are frequently offset by the need to reduce port use,
   to ease NAT/firewall traversal, achieved by combining media streams
   into a single RTP session.  The following sections consider some of
   the issues with using RTCP in sessions with multiple media streams in
   more detail.

6.1.  Timing out SSRCs

   Various issues have been identified with timing out SSRC values when
   sending multiple media streams in an RTP session.

6.1.1.  Problems with RTP/AVPF the T_rr_interval Parameter

   The RTP/AVPF profile includes a method to prevent RTCP reports from
   being sent too often.  This mechanism is described in Section 3.5.3
   of [RFC4585], and is controlled by the T_rr_interval parameter.  It
   works as follows.  When a regular RTCP report is sent, a new random
   value, T_rr_current_interval, is generated, drawn evenly in the range
   0.5 to 1.5 times T_rr_interval.  If a regular RTCP packet is to be
   sent earlier then T_rr_current_interval seconds after the previous
   regular RTCP packet, and there are no feedback messages to be sent,
   then that regular RTCP packet is suppressed, and the next regular
   RTCP packet is scheduled.  The T_rr_current_interval is recalculated
   each time a regular RTCP packet is sent.  The benefit of suppression
   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 skews the distribution of
   the RTCP sending intervals compared to the regular RTCP reporting
   intervals.  The standard RTCP timing rules, including reconsideration
   and the compensation factor, result in the intervals between sending
   RTCP packets having a distribution that is skewed towards the upper
   end of the range [0.5/1.21828, 1.5/1.21828]*Td, where Td is the
   deterministic calculated RTCP reporting interval.  With Td = 5s this
   distribution covers the range [2.052s, 6.156s].  In comparison, the
   RTP/AVPF suppression rules act in an interval that is 0.5 to 1.5
   times T_rr_interval; for T_rr_interval = 5s this is [2.5s, 7.5s].

   The effect of this is that the time between consecutive RTCP packets
   when using T_rr_interval suppression can become large.  The maximum
   time interval between sending one regular RTCP packet and the next,
   when T_rr_interval is being used, occurs when T_rr_current_interval
   takes its maximum value and a regular RTCP packet is suppressed at
   the end of the suppression period, then the next regular RTCP packet

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   is scheduled after its largest possible reporting interval.  Taking
   the worst case of the two intervals gives a maximum time between two
   RTCP reports of 1.5*T_rr_interval + 1.5/1.21828*Td.

   This behaviour can be surprising when Td and T_rr_interval have the
   same value.  That is, when T_rr_interval is configured to match the
   regular RTCP reporting interval.  In this case, one might expect that
   regular RTCP packets are sent according to their usual schedule, but
   feedback packets can be sent early.  However, the above-mentioned
   issue results in the RTCP packets actually being sent in the range
   [0.5*Td, 2.731*Td] with a highly non-uniform distribution, rather
   than the range [0.41*Td, 1.23*Td].  This is perhaps unexpected, but
   is not a problem in itself.  However, when coupled with packet loss,
   it raises the issue of premature timeout.

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 configured RTCP bandwidth allows for an average RTCP
   reporting interval shorter than 5 seconds, the timeout is 25 seconds
   of no activity from the SSRC (RTP or RTCP), otherwise the timeout is
   5 average reporting intervals.

   RTP/AVPF [RFC4585] introduces different timeout behaviours depending
   on the value of T_rr_interval.  When T_rr_interval is 0, it uses the
   same timeout calculation as RTP/AVP.  However, when T_rr_interval is
   non-zero, it replaces Tmin in the timeout calculation, most likely to
   speed up detection of timed out SSRCs.  However, using a non-zero
   T_rr_interval has two consequences for RTP behaviour.

   First, due to suppression, the number of RTP and RTCP packets sent by
   an SSRC that is not an active RTP sender can become very low, because
   of the issue discussed in Section 6.1.1.  As the RTCP packet interval
   can be as long as 2.73*Td, then during a 5*Td time period an endpoint
   might in fact transmit only a single RTCP packet.  The long intervals
   result in fewer RTCP packets, to a point where a single RTCP packet
   loss can sometimes result in timing out an SSRC.

   Second, the RTP/AVPF changes to the timeout rules reduce robustness
   to misconfiguration.  It is common to use RTP/AVPF configured such
   that RTCP packets can be sent frequently, to allow rapid feedback,
   however this makes timeouts very sensitive to T_rr_interval.  For
   example, if two SSRCs are configured one with T_rr_interval = 0.1s
   and the other with T_rr_interval = 0.6s, then this small difference
   will result in the SSRC with the shorter T_rr_interval timing out the
   other if it stops sending RTP packets, since the other RTCP reporting
   interval is more than five times its own.  When RTP/AVP is used, or

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   RTP/AVPF with T_rr_interval = 0, this is a non-issue, as the timeout
   period will be 25s, and differences between configured RTCP bandwidth
   can only cause premature timeouts when the reporting intervals are
   greater than 5s and differ by a factor of five.  To limit the scope
   for such problematic misconfiguration, we propose an update to the
   RTP/AVPF timeout rules in Section 6.1.4.

6.1.3.  Interoperability Between RTP/AVP and RTP/AVPF

   If endpoints implementing the RTP/AVP and RTP/AVPF profiles (or their
   secure variants) are combined within a single RTP session, and the
   RTP/AVPF endpoints use a non-zero T_rr_interval that is significantly
   below 5 seconds, there is a risk that the RTP/AVPF endpoints will
   prematurely timeout the SSRCs of the RTP/AVP endpoints, due to their
   different RTCP timeout rules.  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 SSRCs of the
   RTP/AVPF endpoints.

   Mixing endpoints using two different RTP profiles within a single RTP
   session is NOT RECOMMENDED.  However, if mixed RTP profiles are used,
   and the RTP/AVPF endpoints are not updated to follow Section 6.1.4 of
   this memo, then the RTP/AVPF session SHOULD be configured to use
   T_rr_interval = 4 seconds to avoid premature timeouts.

   The choice of T_rr_interval = 4 seconds for interoperability might
   appear strange.  Intuitively, this value ought to be 5 seconds, to
   make both the RTP/AVP and RTP/AVPF use the same timeout period.
   However, the behaviour outlined in Section 6.1.1 shows that actual
   RTP/AVPF reporting intervals can be longer than expected.  Setting
   T_rr_interval = 4 seconds gives actual RTCP intervals near to those
   expected by RTP/AVP, ensuring interoperability.

6.1.4.  Updated SSRC Timeout Rules

   To ensure interoperability and avoid premature timeouts, all SSRCs in
   an RTP session MUST use the same timeout behaviour.  However,
   previous specification are inconsistent in this regard.  To avoid
   interoperability issues, this memo updates the timeout rules as

   o  For the RTP/AVP, RTP/SAVP, RTP/AVPF, and RTP/SAVPF profiles, the
      timeout interval SHALL be calculated using a multiplier of five
      times the deterministic RTCP reporting interval.  That is, the
      timeout interval SHALL be 5*Td.

   o  For the RTP/AVP, RTP/SAVP, RTP/AVPF, and RTP/SAVPF profiles,
      calculation of Td, for the purpose of calculating the participant

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      timeout only, SHALL be done using a Tmin value of 5 seconds and
      not the reduced minimal interval, even if the reduced minimum
      interval is 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

6.2.1.  RTP/AVP and RTP/SAVP

   When using the RTP/AVP or RTP/SAVP profiles, the options for tuning
   the RTCP reporting intervals are limited to the RTCP sender and
   receiver bandwidth, and whether the minimum RTCP interval is scaled
   according to the bandwidth.  As the scheduling algorithm includes
   both randomisation and reconsideration, one cannot simply calculate
   the expected average transmission interval using the formula for Td
   given in Section 6.3.1 of [RFC3550].  However, by considering the
   inputs to that expression, and the randomisation and reconsideration
   rules, we can begin to understand the behaviour of the RTCP
   transmission interval.

   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 Tmin.
       The default value of Tmin is 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 per second.
       In SDP the RTP session bandwidth is signalled using a "b=AS"
       line.  An RTP Session bandwidth of 72kbps results in Tmin being 5
       seconds.  An RTP session bandwidth of 360kbps of course gives a
       Tmin of 1 second, and to achieve a Tmin equal to once every frame
       for a 25 frame-per-second video stream requires an RTP session
       bandwidth of 9Mbps.  Use of the RTP/AVPF or RTP/SAVPF profile
       allows more frequent RTCP reports for the same bandwidth, as
       discussed below.

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   c.  The value of Td scales with the number of SSRCs and the average
       size of the RTCP reports, to keep the overall RTCP bandwidth

   d.  The actual transmission interval for a Td value is in the range
       [0.5*Td/1.21828,1.5*Td/1.21828], and the distribution is skewed,
       due to reconsideration, with the majority of the probability mass
       being above Td.  This means, for example, that for Td = 5s, the
       actual transmission interval will be distributed in the range
       [2.052s, 6.156s], and tending towards the upper half of the
       interval.  Note that Tmin parameter limits the value of Td before
       randomisation and reconsideration are applied, so the actual
       transmission interval will cover a range extending below Tmin.

   Given the above, we can calculate the number of SSRCs, n, that an RTP
   session with 5% of the session bandwidth assigned to RTCP can support
   while maintaining Td equal to Tmin.  This will tell us how many media
   streams we can report on, keeping the RTCP overhead within acceptable
   bounds.  We make two assumptions that simplify the calculation: that
   all SSRCs are senders, and that they all send compound RTCP packets
   comprising an SR packet with n-1 report blocks, followed by an SDES
   packet containing a 16 octet CNAME value [RFC7022] (such RTCP packets
   will vary in size between 54 and 798 octets depending on n, up to the
   maximum of 31 report blocks that can be included in an SR packet).
   If we put this packet size, and a 5% RTCP bandwidth fraction into the
   RTCP interval calculation in Section 6.3.1 of [RFC3550], and
   calculate the value of n needed to give Td = Tmin for the scaled
   minimum interval, we find n=9 SSRCs can be supported (irrespective of
   the interval, due to the way the reporting interval scales with the
   session bandwidth).  We see that to support more SSRCs, we need to
   increase the RTCP bandwidth fraction from 5%; changing the session
   bandwidth does not help due to the limit of Tmin.

   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 have a powerful additional tool
   for tuning RTCP transmissions: the T_rr_interval parameter.  Use of
   this parameter allows short RTCP reporting intervals; alternatively
   it gives the ability to sent frequent RTCP feedback without sending
   frequent regular RTCP reports.

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   The use of the RTP/AVPF or RTP/SAVPF profile with T_rr_interval set
   to a value greater than zero but smaller than Tmin allows more
   frequent RTCP feedback than the RTP/AVP or RTP/SAVP profiles, for a
   given RTCP bandwidth.  This happens because Tmin is set to zero after
   the transmission of the initial RTCP report, causing the reporting
   interval for later packet to be determined by the usual RTCP
   bandwidth-based calculation, with Tmin=0, and the T_rr_interval.
   This has the effect that we are no longer restricted by the minimal
   interval (whether the default 5 second minimum, or the reduced
   minimum interval).  Rather, the RTCP bandwidth and the T_rr_interval
   are the governing factors, allowing faster feedback.  Applications
   that care about rapid regular RTCP feedback ought to consider using
   the RTP/AVPF or RTP/SAVPF profile, even if they don't use the
   feedback features of that profile.

   The use of the RTP/AVPF or RTP/SAVPF profile allows RTCP feedback
   packets to be sent frequently, without also requiring regular RTCP
   reports to be sent frequently, since T_rr_interval limits the rate at
   which regular RTCP packets can be sent, while still permitting RTCP
   feedback packets to be sent.  Applications that can use feedback
   packets for some media streams, e.g., video streams, but don't want
   frequent regular reporting for other media streams, can 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,
   for the video reporting.  This allows the available RTCP bandwidth to
   be devoted on the feedback that provides the most utility for the

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

   When T_rr_interval is non-zero, there are configurations that need to
   be avoided.  If the RTCP bandwidth chosen is such that the Td value
   is smaller than, but close to, T_rr_interval, then the actual regular
   RTCP packet transmission interval can become very large, as discussed
   in Section 6.1.1.  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 the RTP/AVPF or RTP/SAVPF profiles, using T_rr_interval = 0 has
   utility, and results in a behaviour where the RTCP transmission is
   only limited by the bandwidth, i.e., no Tmin limitations at all.

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   This allows more frequent regular RTCP reporting than can be achieved
   using the RTP/AVP profile.  Many configurations of RTCP will not
   consume all the bandwidth that they have been configured to use, but
   this configuration will consume what it has been given.  Note that
   the same behaviour will be achieved as long as T_rr_interval is
   smaller than 1/3 of Td as that prevents T_rr_interval from affecting
   the transmission.

   There exists no method for using different regular RTCP reporting
   intervals depending on the media type or individual media stream,
   other than using a separate RTP session for each type or stream.

7.  Security Considerations

   When using the secure RTP protocol (RTP/SAVP) [RFC3711], or the
   secure variant of the feedback profile (RTP/SAVPF) [RFC5124], the
   cryptographic context of a compound secure RTCP 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.

   Otherwise, the standard security considerations of RTP apply; sending
   multiple media streams from a single endpoint in a single RTP session
   does not appear to have different security consequences than sending
   the same number of media streams spread across different RTP

8.  IANA Considerations

   No IANA actions are needed.

9.  References

9.1.  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.

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   [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

   [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.

9.2.  Informative References

              Westerlund, M., Perkins, C., and J. Lennox, "Sending
              Multiple Types of Media in a Single RTP Session", draft-
              ietf-avtcore-multi-media-rtp-session-07 (work in
              progress), March 2015.

              Lennox, J., Westerlund, M., Wu, W., 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-05 (work
              in progress), February 2015.

              Westerlund, M. and S. Wenger, "RTP Topologies", draft-
              ietf-avtcore-rtp-topologies-update-06 (work in progress),
              March 2015.

              Duckworth, M., Pepperell, A., and S. Wenger, "Framework
              for Telepresence Multi-Streams", draft-ietf-clue-
              framework-21 (work in progress), March 2015.

              Holmberg, C., Alvestrand, H., and C. Jennings,
              "Negotiating Media Multiplexing Using the Session
              Description Protocol (SDP)", draft-ietf-mmusic-sdp-bundle-
              negotiation-18 (work in progress), March 2015.

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

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Internet-Draft  Multiple Media Streams in an RTP Session      March 2015

   [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.

   [RFC5104]  Wenger, S., Chandra, U., Westerlund, M., and B. Burman,
              "Codec Control Messages in the RTP Audio-Visual Profile
              with Feedback (AVPF)", RFC 5104, February 2008.

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

   [RFC7022]  Begen, A., Perkins, C., Wing, D., and E. Rescorla,
              "Guidelines for Choosing RTP Control Protocol (RTCP)
              Canonical Names (CNAMEs)", RFC 7022, September 2013.

Authors' Addresses

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

   Email: jonathan@vidyo.com

   Magnus Westerlund
   Farogatan 6
   SE-164 80 Kista

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

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Internet-Draft  Multiple Media Streams in an RTP Session      March 2015

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

   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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