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rfc:rfc8108

Internet Engineering Task Force (IETF) J. Lennox Request for Comments: 8108 Vidyo Updates: 3550, 4585 M. Westerlund Category: Standards Track Ericsson ISSN: 2070-1721 Q. Wu

                                                                Huawei
                                                            C. Perkins
                                                 University of Glasgow
                                                            March 2017
        Sending Multiple RTP Streams in a Single RTP Session

Abstract

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

Status of This Memo

 This is an Internet Standards Track document.
 This document is a product of the Internet Engineering Task Force
 (IETF).  It represents the consensus of the IETF community.  It has
 received public review and has been approved for publication by the
 Internet Engineering Steering Group (IESG).  Further information on
 Internet Standards is available in Section 2 of RFC 7841.
 Information about the current status of this document, any errata,
 and how to provide feedback on it may be obtained at
 http://www.rfc-editor.org/info/rfc8108.

Lennox, et al. Standards Track [Page 1] RFC 8108 Multiple Media Streams in an RTP Session March 2017

Copyright Notice

 Copyright (c) 2017 IETF Trust and the persons identified as the
 document authors.  All rights reserved.
 This document is subject to BCP 78 and the IETF Trust's Legal
 Provisions Relating to IETF Documents
 (http://trustee.ietf.org/license-info) in effect on the date of
 publication of this document.  Please review these documents
 carefully, as they describe your rights and restrictions with respect
 to this document.  Code Components extracted from this document must
 include Simplified BSD License text as described in Section 4.e of
 the Trust Legal Provisions and are provided without warranty as
 described in the Simplified BSD License.

Lennox, et al. Standards Track [Page 2] RFC 8108 Multiple Media Streams in an RTP Session March 2017

Table of Contents

 1. Introduction ....................................................4
 2. Terminology .....................................................4
 3. Use Cases for Multi-Stream Endpoints ............................4
    3.1. Endpoints with Multiple Capture Devices ....................4
    3.2. Multiple Media Types in a Single RTP Session ...............5
    3.3. Multiple Stream Mixers .....................................5
    3.4. Multiple SSRCs for a Single Media Source ...................5
 4. Use of RTP by Endpoints That Send Multiple Media Streams ........6
 5. Use of RTCP by Endpoints That Send Multiple Media Streams .......6
    5.1. RTCP Reporting Requirement .................................7
    5.2. Initial Reporting Interval .................................7
    5.3. Aggregation of Reports into Compound RTCP Packets ..........8
         5.3.1. Maintaining AVG_RTCP_SIZE ...........................9
         5.3.2. Scheduling RTCP when Aggregating Multiple SSRCs ....10
    5.4. Use of RTP/AVPF or RTP/SAVPF Feedback .....................13
         5.4.1. Choice of SSRC for Feedback Packets ................13
         5.4.2. Scheduling an RTCP Feedback Packet .................14
 6. Adding and Removing SSRCs ......................................15
    6.1. Adding RTP Streams ........................................16
    6.2. Removing RTP Streams ......................................16
 7. RTCP Considerations for Streams with Disparate Rates ...........17
    7.1. Timing Out SSRCs ..........................................19
         7.1.1. Problems with the RTP/AVPF T_rr_interval
                Parameter ..........................................19
         7.1.2. Avoiding Premature Timeout .........................20
         7.1.3. Interoperability between RTP/AVP and RTP/AVPF ......21
         7.1.4. Updated SSRC Timeout Rules .........................22
    7.2. Tuning RTCP Transmissions .................................22
         7.2.1. RTP/AVP and RTP/SAVP ...............................22
         7.2.2. RTP/AVPF and RTP/SAVPF .............................24
 8. Security Considerations ........................................25
 9. References .....................................................26
    9.1. Normative References ......................................26
    9.2. Informative References ....................................26
 Acknowledgments ...................................................29
 Authors' Addresses ................................................29

Lennox, et al. Standards Track [Page 3] RFC 8108 Multiple Media Streams in an RTP Session March 2017

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 source and, thus,
 used a single RTP stream and 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 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; thus, the specification is somewhat unclear in
 places.
 This memo updates [RFC3550] to clarify behavior in use cases where
 endpoints use multiple SSRCs.  It also updates [RFC4585] to resolve
 problems with regard to timeout of inactive SSRCs and to clarify
 behavior around 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 single RTP session is when an endpoint
 has multiple capture devices and, hence, can generate multiple media
 sources, of the same media type and characteristics.  For example,
 telepresence systems of the type described by the CLUE Telepresence
 Framework [CLUE-FRAME] often have multiple cameras or microphones
 covering various areas of a room and, hence, send several RTP streams
 of each type within a single RTP session.

Lennox, et al. Standards Track [Page 4] RFC 8108 Multiple Media Streams in an RTP Session March 2017

3.2. Multiple Media Types in a Single RTP Session

 Recent work has updated RTP [MULTI-RTP] and Session Description
 Protocol (SDP) [SDP-BUNDLE] 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 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 that can involve a central device
 that itself generates multiple RTP 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 [RFC7667].  This is a middlebox that
 receives RTP 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: it terminates
 RTCP reporting about the mixed streams; it can rewrite SSRCs,
 timestamps, and sequence numbers, as well as the contents of the RTP
 payloads; and it can turn sources on and off at will without
 appearing to generate 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 multiple SSRCs can be used to send
 data from a single media source within a single RTP session.  These
 include, but are not limited to, transport robustness tools, such as
 the RTP retransmission payload format [RFC4588], that require one
 SSRC to be used for the media data and another SSRC for the repair
 data.  Similarly, some layered media encoding schemes, for example,
 H.264 Scalable Video Coding (SVC) [RFC6190], can be used in a
 configuration where each layer is sent using a different SSRC within
 a single RTP session.

Lennox, et al. Standards Track [Page 5] RFC 8108 Multiple Media Streams in an RTP Session March 2017

4. Use of RTP by Endpoints That Send Multiple Media Streams

 RTP is inherently a group communication protocol.  Each endpoint in
 an RTP session will use one or more SSRCs, as will some types of RTP-
 level middlebox.  Accordingly, unless restrictions on the number of
 SSRCs have been signaled, RTP endpoints can expect to receive RTP
 data packets sent using a number of different SSRCs, within a single
 RTP session.  This can occur irrespective of whether the RTP session
 is running over a point-to-point connection or a multicast group,
 since middleboxes can be used to connect multiple transport
 connections together into a single RTP session (the RTP session is
 defined by the shared SSRC space, not by the transport connections).
 Furthermore, if RTP mixers are used, some SSRCs might only be visible
 in the contributing source (CSRC) list of an RTP packet and in RTCP,
 and might not appear directly as the SSRC of an RTP data packet.
 Every RTP endpoint will have an allocated share of the available
 session bandwidth, as determined by signaling and congestion control.
 The endpoint needs to keep its total media sending rate within this
 share.  However, endpoints that send multiple RTP streams do not
 necessarily need to subdivide their share of the available bandwidth
 independently or uniformly to each RTP stream and its SSRCs.  In
 particular, an endpoint can vary the bandwidth allocation to
 different streams depending on their needs, and it 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
 RTP streams, or their redundancy streams, as more or less bandwidth
 becomes available.

5. Use of RTCP by Endpoints That Send Multiple Media Streams

 RTCP is defined in Section 6 of [RFC3550].  The description of the
 protocol is phrased in terms of the behavior 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 have multiple SSRC values, implementations MUST treat each
 SSRC as a separate participant in the RTP session, so that an
 endpoint that has multiple SSRCs counts as multiple participants.

Lennox, et al. Standards Track [Page 6] RFC 8108 Multiple Media Streams in an RTP Session March 2017

5.1. RTCP Reporting Requirement

 An RTP endpoint that has multiple SSRCs MUST treat each SSRC as a
 separate participant in the RTP session.  Each SSRC will maintain its
 own RTCP-related state information and, hence, will have its own RTCP
 reporting interval that determines when it sends RTCP reports.  If
 the mechanism in [MULTI-STREAM-OPT] 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
 randomized in the usual way, but reports can be aggregated as
 described in Section 5.3.

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 an RTP session, independent of the
 number of SSRCs used by the endpoint.  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_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.

Lennox, et al. Standards Track [Page 7] RFC 8108 Multiple Media Streams in an RTP Session March 2017

    Note: The above is chosen to match the TCP maximum initial window
    of four packets [RFC3390], not the larger TCP initial windows for
    which there is an ongoing experiment [RFC6928].  The reason for
    this is a desire to be conservative, since an RTP endpoint will
    also in many cases start sending RTP data packets 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
 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.
 This allows RTP translators and mixers to generate compound RTCP
 packets that contain multiple Sender Report (SR) or Receiver Report
 (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 can 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.

Lennox, et al. Standards Track [Page 8] RFC 8108 Multiple Media Streams in an RTP Session March 2017

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 SR or 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.
 A participant 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 behavior specified in Section 7.1.2 of this
 memo.  This mitigation is in place in those cases where the RTCP
 bandwidth is sufficiently high that an endpoint, using 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-updated endpoint's RTCP reporting is
 still negatively impacted even if the premature timeouts of its SSRCs

Lennox, et al. Standards Track [Page 9] RFC 8108 Multiple Media Streams in an RTP Session March 2017

 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 be 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 when Aggregating Multiple SSRCs

 This section revises and extends the behavior defined in Section 6.3
 of [RFC3550], and in Section 3.5.3 of [RFC4585] if the RTP/AVPF
 profile or the RTP/SAVPF profile is used, regarding actions to take
 when scheduling and sending RTCP packets where multiple reporting
 SSRCs are aggregating their RTCP packets into the same compound RTCP
 packet.  These changes to the RTCP scheduling rules are needed to
 maintain important RTCP timing properties, including the inter-packet
 distribution, and the behavior during flash joins and other changes
 in session membership.
 The variables tn, tp, tc, T, and Td used in the following are defined
 in Section 6.3 of [RFC3550].  The variables T_rr_interval and
 T_rr_last are defined in [RFC4585].
 Each endpoint MUST schedule RTCP transmission independently for each
 of its SSRCs using the regular calculation of tn for the RTP profile
 being used.  Each time the timer tn expires for an SSRC, the endpoint
 MUST perform RTCP timer reconsideration and, if applicable,
 suppression based on T_rr_interval.  If the result indicates that a
 compound RTCP packet is to be sent by that SSRC, and the transmission
 is not an early RTCP packet [RFC4585], then the endpoint SHOULD try
 to aggregate RTCP packets of additional SSRCs that are scheduled in
 the future into the compound RTCP packet before it is sent.  The
 reason to limit or not aggregate due to backwards compatibility
 reasons is discussed in Section 5.3.1.
 Aggregation proceeds as follows.  The endpoint selects the SSRC that
 has the smallest tn value after the current time, tc, and prepares
 the RTCP packets that SSRC would send if its timer tn expired at tc.
 If those RTCP packets will fit into the compound RTCP packet that is
 being generated, taking into account the path MTU and the previously
 added RTCP packets, then they are added to the compound RTCP packet;
 otherwise, they are discarded.  This process is repeated for each
 SSRC, in order of increasing tn, until the compound RTCP packet is
 full or all SSRCs have been aggregated.  At that point, the compound
 RTCP packet is sent.

Lennox, et al. Standards Track [Page 10] RFC 8108 Multiple Media Streams in an RTP Session March 2017

 When the compound RTCP packet is sent, the endpoint MUST update tp,
 tn, and T_rr_last (if applicable) for each SSRC that was included.
 These variables are updated as follows:
 a.  For the first SSRC that reported in the compound RTCP packet, set
     the effective transmission time, tt, of that SSRC to tc.
 b.  For each additional SSRC that reported in the compound RTCP
     packet, calculate the transmission time that SSRC would have had
     if it had not been aggregated into the compound RTCP packet.
     This is derived by taking tn for that SSRC, then performing
     reconsideration and updating tn until tp + T <= tn.  Once this is
     done, set the effective transmission time, tt, for that SSRC to
     the calculated value of tn.  If the RTP/AVPF profile or the RTP/
     SAVPF profile is being used, then suppression based on
     T_rr_interval MUST NOT be used in this calculation.
 c.  Calculate average effective transmission time, tt_avg, for the
     compound RTCP packet based on the tt values for all SSRCs sent in
     the compound RTCP packet.  Set tp for each of the SSRCs sent in
     the compound RTCP packet to tt_avg.  If the RTP/AVPF profile or
     the RTP/SAVPF profile is being used, set T_tt_last for each SSRC
     sent in the compound RTCP packet to tt_avg.
 d.  For each of the SSRCs sent in the compound RTCP packet, calculate
     new tn values based on the updated parameters and the usual RTCP
     timing rules and reschedule the timers.
 When using the RTP/AVPF profile or the RTP/SAVPF profile, the above
 mechanism only attempts to aggregate RTCP packets when the compound
 RTCP packet to be sent is not an early RTCP packet, and hence the
 algorithm in Section 3.5.3 of [RFC4585] will control RTCP scheduling.
 If T_rr_interval == 0, or if T_rr_interval != 0 and option 1, 2a, or
 2b of the algorithm are chosen, then the above mechanism updates the
 necessary variables.  However, if the transmission is suppressed per
 option 2c of the algorithm, then tp is updated to tc as aggregation
 has not taken place.
 Reverse reconsideration MUST be performed following Section 6.3.4 of
 [RFC3550].  In some cases, this can lead to the value of tp after
 reverse reconsideration being larger than tc.  This is not a problem,
 and has the desired effect of proportionally pulling the tp value
 towards tc (as well as tn) as the reporting interval shrinks in
 direct proportion the reduced group size.
 The above algorithm has been shown in simulations [Sim88] [Sim92] to
 maintain the inter-RTCP packet transmission time distribution for
 each SSRC and to consume the same amount of bandwidth as

Lennox, et al. Standards Track [Page 11] RFC 8108 Multiple Media Streams in an RTP Session March 2017

 non-aggregated RTCP packets.  With this algorithm, the actual
 transmission interval for an 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 the 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 compensates for
 the bias that is otherwise introduced by picking the shortest tn
 value out of the N SSRCs included in aggregate.
 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 behavior under SSRC group size
 changes is as follows:
 RTP sessions where the number of SSRCs is growing:  When the group
    size is growing, Td grows in proportion to the number of new SSRCs
    in the group.  When reconsideration is performed due to expiry of
    the tn timer, 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
    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 SSRCs is 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 whether or not
    aggregation is being used.  The negative effect that can occur on
    removing an SSRC is that the most favorable 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 have found no significant
 negative impact on the scheduling algorithm.

Lennox, et al. Standards Track [Page 12] RFC 8108 Multiple Media Streams in an RTP Session March 2017

5.4. Use of RTP/AVPF or RTP/SAVPF Feedback

 This section discusses the transmission of RTP/AVPF feedback packets
 when the transmitting endpoint has multiple SSRCs.  The guidelines in
 this section also apply to endpoints using the RTP/SAVPF profile.

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 for 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 of whether or not the SSRC can be used in immediate
    mode.
 o  Some RTCP feedback packet types require consistency in the SSRC
    used.  For example, if a Temporary Maximum Media Stream Bit Rate
    Request (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, it 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 contain 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].

Lennox, et al. Standards Track [Page 13] RFC 8108 Multiple Media Streams in an RTP Session March 2017

5.4.2. Scheduling an RTCP Feedback Packet

 When an SSRC has a need to transmit a feedback packet in early mode,
 it MUST schedule that packet following the algorithm in Section 3.5
 of [RFC4585] modified as follows:
 o  To determine whether an RTP session is considered to be a point-
    to-point session or a multiparty session, an endpoint MUST count
    the number of distinct RTCP SDES CNAME values used by the SSRCs
    listed in the SSRC field of RTP data packets it receives and in
    the "SSRC of sender" field of RTCP SR, RR, RTPFB, or PSFB packets
    it receives.  An RTP session is considered to be a multiparty
    session if more than one CNAME is used by those SSRCs, unless
    signaling indicates that the session is to be handled as point to
    point or RTCP reporting groups [MULTI-STREAM-OPT] are used.  If
    RTCP reporting groups are used, an RTP session is considered to be
    a point-to-point session if the endpoint receives only a single
    reporting group and is considered to be a multiparty session if
    multiple reporting groups are received or a combination of
    reporting groups and SSRCs that are not part of a reporting group
    are received.  Endpoints MUST NOT determine whether an RTP session
    is multiparty or point to point based on the type of connection
    (unicast or multicast) used, or on the number of SSRCs received.
 o  When checking if there is already a scheduled compound RTCP packet
    containing feedback messages (Step 2 in Section 3.5.2 of
    [RFC4585]), that check MUST be done considering all local SSRCs.
 o  If an 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 behavior in item 4a in Section 3.5.2 of
    [RFC4585].
 The first bullet point above specifies a rule to determine if an RTP
 session is to be considered a point-to-point session or a multiparty
 session.  This rule is straightforward to implement, but is known to
 incorrectly classify some sessions as multiparty sessions.  The known
 problems are as follows:
 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 an 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
    endpoint.

Lennox, et al. Standards Track [Page 14] RFC 8108 Multiple Media Streams in an RTP Session March 2017

 Selective Forwarding Middlebox:  The Selective Forwarding Middlebox
    (SFM) as defined in Section 3.7 of [RFC7667] 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.
 In the above cases, it is very reasonable to use RTCP reporting
 groups [MULTI-STREAM-OPT].  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 RTP session based on Any Source
 Multicast 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.
 Finally, we note that signaling mechanisms could be defined to
 override the rules when they would result in the wrong
 classification.

6. Adding and Removing SSRCs

 The set of SSRCs present in a single RTP session can vary over time
 due to changes in the number of endpoints in the session or due to
 changes in the number or type of RTP streams being sent.
 Every endpoint in an RTP session will have at least one SSRC that it
 uses for RTCP reporting, and for sending media if desired.  It can
 also have additional SSRCs, for sending extra media sources or for
 additional RTCP reporting.  If the set of media sources being sent
 changes, then the set of SSRCs being sent will change.  Changes in
 the media format or clock rate might also require changes in the set
 of SSRCs used.  An endpoint can also have more SSRCs than it has
 active RTP streams, and send RTCP relating to SSRCs that are not
 currently sending RTP data packets so that its peers are aware of the
 SSRCs, and have the associated context (e.g., clock synchronization
 and an SDES CNAME) in place to be able to play out media as soon as
 they becomes active.
 In the following, we describe some considerations around adding and
 removing RTP streams and their associated SSRCs.

Lennox, et al. Standards Track [Page 15] RFC 8108 Multiple Media Streams in an RTP Session March 2017

6.1. Adding RTP Streams

 When an endpoint joins an RTP session, it can have zero, one, or more
 RTP streams it will send, or that it is prepared to send.  If it has
 no RTP stream it plans to send, it still needs an SSRC that will be
 used to send RTCP feedback.  If it will send one or more RTP streams,
 it will need the corresponding number of SSRC values.  The SSRCs used
 by an endpoint are made known to other endpoints in the RTP session
 by sending RTP and RTCP packets.  SSRCs can also be signaled using
 non-RTP means (e.g., [RFC5576]).  Unless restricted by signaling, an
 endpoint can, at any time, send an additional RTP stream, identified
 by a new SSRC (this might be associated with a signaling event, but
 that is outside the scope of this memo).  This makes the new SSRC
 visible to the other endpoints in the session, since they share the
 single SSRC space inherent in the definition of an RTP session.
 An endpoint that has never sent an RTP stream will have an SSRC that
 it uses for RTCP reporting.  If that endpoint wants to start sending
 an RTP stream, it is RECOMMENDED that it use its existing SSRC for
 that stream, since otherwise the participant count in the RTP session
 will be unnecessarily increased, leading to a longer RTCP reporting
 interval and larger RTCP reports due to cross reporting.  If the
 endpoint wants to start sending more than one RTP stream, it will
 need to generate a new SSRC for the second and any subsequent RTP
 streams.
 An endpoint that has previously stopped sending an RTP stream, and
 that wants to start sending a new RTP stream, cannot generally reuse
 the existing SSRC, and often needs to generate a new SSRC, because an
 SSRC cannot change media type (e.g., audio to video) or RTP timestamp
 clock rate [RFC7160] and because the SSRC might be associated with a
 particular semantic by the application (note: an RTP stream can pause
 and restart using the same SSRC, provided RTCP is sent for that SSRC
 during the pause; these rules only apply to new RTP streams reusing
 an existing SSRC).

6.2. Removing RTP Streams

 An SSRC is removed from an RTP session in one of two ways.  When an
 endpoint stops sending RTP and RTCP packets using an SSRC, then that
 SSRC will eventually time out as described in Section 6.3.5 of
 [RFC3550].  Alternatively, an SSRC can be explicitly removed from use
 by sending an RTCP BYE packet as described in Section 6.3.7 of
 [RFC3550].  It is RECOMMENDED that SSRCs be removed from use by
 sending an RTCP BYE packet.  Note that [RFC3550] requires that the
 RTCP BYE SHOULD be the last RTP/RTCP packet sent in the RTP session

Lennox, et al. Standards Track [Page 16] RFC 8108 Multiple Media Streams in an RTP Session March 2017

 for an SSRC.  If an endpoint needs to restart an RTP stream after
 sending an RTCP BYE for its SSRC, it needs to generate a new SSRC
 value for that stream.
 The finality of sending RTCP BYE means that endpoints need to
 consider if the ceasing of transmission of an RTP stream is temporary
 or permanent.  Temporary suspension of media transmission using a
 particular RTP stream (SSRC) needs to maintain that SSRC as an active
 participant, by continuing RTCP transmission for it.  That way the
 media sending can be resumed immediately, knowing that the context is
 in place.  When permanently halting transmission, a participant needs
 to send an RTCP BYE to allow the other participants to use the RTCP
 bandwidth resources and clean up their state databases.
 An endpoint that ceases transmission of all its RTP streams but
 remains in the RTP session MUST maintain at least one SSRC that is to
 be used for RTCP reporting and feedback (i.e., it cannot send a BYE
 for all SSRCs, but needs to retain at least one active SSRC).  As
 some Feedback packets can be bound to media type, there might be a
 need to maintain one SSRC per media type within an RTP session.  An
 alternative can be to create a new SSRC to use for RTCP reporting and
 feedback.  However, to avoid the perception that an endpoint drops
 completely out of an RTP session, such a new SSRC ought to be
 established first -- before terminating all the existing SSRCs.

7. 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 [RFC3556]) 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 randomization,
 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 an RTP stream.
 For example, consider a scenario in which 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

Lennox, et al. Standards Track [Page 17] RFC 8108 Multiple Media Streams in an RTP Session March 2017

 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.
 This bandwidth mismatch can be reduced by careful tuning of the RTCP
 parameters, especially trr_int when the RTP/AVPF profile is used, but
 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 behaviors can also occur with
 SSRCs carrying the same media type.  A common case in multiparty
 conferencing is when a small number of video streams are shown in
 high resolution, while the others are shown as low-resolution
 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 Signal-to-Noise Ratio (SNR)
 fidelity of the image.  These differences affect not only the needed
 bitrates, but also possible transmission behaviors, 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.  Other similar scenarios can
 also exist.
 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 RTP stream 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.  Hence, sending multiple RTP streams in an RTP
 session 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, and only send or process reports and feedback that apply
 to that particular SSRC and its media type.  Signaling solutions

Lennox, et al. Standards Track [Page 18] RFC 8108 Multiple Media Streams in an RTP Session March 2017

 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.
 From an RTCP perspective, therefore, it can be seen that there are
 advantages to using separate RTP sessions for each media source,
 rather than sending multiple media sources 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 sources
 into a single RTP session.  The following sections consider some of
 the issues with using RTCP in sessions with multiple media sources in
 more detail.

7.1. Timing Out SSRCs

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

7.1.1. Problems with the RTP/AVPF T_rr_interval Parameter

 The RTP/AVPF profile includes a method to prevent regular RTCP
 reports from being sent too often.  This mechanism is described in
 Section 3.5.3 of [RFC4585]; it 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 than 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 = 5 s,
 this distribution covers the range [2.052 s, 6.156 s].  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
 [2.5 s, 7.5 s].

Lennox, et al. Standards Track [Page 19] RFC 8108 Multiple Media Streams in an RTP Session March 2017

 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
 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 behavior 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.

7.1.2. Avoiding Premature Timeout

 In RTP/AVP [RFC3550] the timeout behavior is simple; it 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 behaviors 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 behavior.
 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 7.1.1.  As the RTCP packet interval
 can be as long as 2.73*Td, 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;

Lennox, et al. Standards Track [Page 20] RFC 8108 Multiple Media Streams in an RTP Session March 2017

 however, this makes timeouts very sensitive to T_rr_interval.  For
 example, if two SSRCs are configured, one with T_rr_interval = 0.1 s
 and the other with T_rr_interval = 0.6 s, 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
 RTP/AVPF with T_rr_interval = 0, this is a non-issue, as the timeout
 period will be 25 s, and differences between configured RTCP
 bandwidth can only cause premature timeouts when the reporting
 intervals are greater than 5 s and differ by a factor of five.  To
 limit the scope for such problematic misconfiguration, we define an
 update to the RTP/AVPF timeout rules in Section 7.1.4.

7.1.3. 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 time out 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 significantly larger than 5 seconds,
 there is a risk that the RTP/AVP endpoints will time out 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 7.1.4 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 behavior outlined in Section 7.1.1 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.

Lennox, et al. Standards Track [Page 21] RFC 8108 Multiple Media Streams in an RTP Session March 2017

7.1.4. Updated SSRC Timeout Rules

 To ensure interoperability and avoid premature timeouts, all SSRCs in
 an RTP session MUST use the same timeout behavior.  However, previous
 specifications are inconsistent in this regard.  To avoid
 interoperability issues, this memo updates the timeout rules as
 follows:
 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
    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 behavior for the RTP/AVPF or RTP/SAVPF profiles when
 T_rr_interval != 0.  Specifically, the first paragraph of
 Section 3.5.4 of [RFC4585] is updated to use Tmin instead of
 T_rr_interval in the timeout calculation for RTP/AVPF entities.

7.2. Tuning RTCP Transmissions

 This subsection discusses what tuning can be done to reduce the
 downsides of the shared RTCP packet intervals.  First, what
 possibilities exist for the RTP/AVP [RFC3551] profile are listed
 followed by what additional tools are provided by RTP/AVPF [RFC4585].

7.2.1. 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 randomization 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 randomization and reconsideration
 rules, we can begin to understand the behavior of the RTCP
 transmission interval.

Lennox, et al. Standards Track [Page 22] RFC 8108 Multiple Media Streams in an RTP Session March 2017

 Let's start with some basic observations:
 a.  Unless the scaled minimum RTCP interval is used, 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 signaled using a "b=AS"
     line.  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 frame-per-second video stream requires an RTP
     session bandwidth of 9 Mbps.  Use of the RTP/AVPF or RTP/SAVPF
     profile allows more frequent RTCP reports for the same bandwidth,
     as discussed below.
 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
     constant.
 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 = 5 s, the
     actual transmission interval will be distributed in the range
     [2.052 s, 6.156 s], and tending towards the upper half of the
     interval.  Note that Tmin parameter limits the value of Td before
     randomization 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 RTP
 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 without
 changing the scaled minimum interval, we need to increase the RTCP

Lennox, et al. Standards Track [Page 23] RFC 8108 Multiple Media Streams in an RTP Session March 2017

 bandwidth fraction from 5%; changing the session bandwidth to a
 higher value would reduce the Tmin.  However, if using the default 5%
 allocation of RTCP bandwidth, an increase will result in more SSRCs
 being supported given a fixed Td target.
 Based on the above, when using the RTP/AVP profile or the RTP/SAVP
 profile, the key limitation for rapid RTCP reporting in small unicast
 sessions is going to be the Tmin value.  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.

7.2.2. 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.
 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 RTP streams, e.g., video streams, but don't want
 frequent regular reporting for other RTP 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,

Lennox, et al. Standards Track [Page 24] RFC 8108 Multiple Media Streams in an RTP Session March 2017

 for the video reporting.  This allows the available RTCP bandwidth to
 be devoted on the feedback that provides the most utility for the
 application.
 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 behavior
 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 7.1.1.  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
 the range becoming [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 behavior where the RTCP transmission is only
 limited by the bandwidth, i.e., no Tmin limitations at all.  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 behavior 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 RTP stream, other
 than using a separate RTP session for each type or stream.

8. 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] that
 allow use of per-SSRC keying.
 Otherwise, the standard security considerations of RTP apply; sending
 multiple RTP streams from a single endpoint in a single RTP session
 does not appear to have different security consequences than sending
 the same number of RTP streams spread across different RTP sessions.

Lennox, et al. Standards Track [Page 25] RFC 8108 Multiple Media Streams in an RTP Session March 2017

9. References

9.1. Normative References

 [RFC2119]  Bradner, S., "Key words for use in RFCs to Indicate
            Requirement Levels", BCP 14, RFC 2119,
            DOI 10.17487/RFC2119, March 1997,
            <http://www.rfc-editor.org/info/rfc2119>.
 [RFC3550]  Schulzrinne, H., Casner, S., Frederick, R., and V.
            Jacobson, "RTP: A Transport Protocol for Real-Time
            Applications", STD 64, RFC 3550, DOI 10.17487/RFC3550,
            July 2003, <http://www.rfc-editor.org/info/rfc3550>.
 [RFC3711]  Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K.
            Norrman, "The Secure Real-time Transport Protocol (SRTP)",
            RFC 3711, DOI 10.17487/RFC3711, March 2004,
            <http://www.rfc-editor.org/info/rfc3711>.
 [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,
            DOI 10.17487/RFC4585, July 2006,
            <http://www.rfc-editor.org/info/rfc4585>.
 [RFC5124]  Ott, J. and E. Carrara, "Extended Secure RTP Profile for
            Real-time Transport Control Protocol (RTCP)-Based Feedback
            (RTP/SAVPF)", RFC 5124, DOI 10.17487/RFC5124, February
            2008, <http://www.rfc-editor.org/info/rfc5124>.
 [RFC5506]  Johansson, I. and M. Westerlund, "Support for Reduced-Size
            Real-Time Transport Control Protocol (RTCP): Opportunities
            and Consequences", RFC 5506, DOI 10.17487/RFC5506, April
            2009, <http://www.rfc-editor.org/info/rfc5506>.

9.2. Informative References

 [CLUE-FRAME]
            Duckworth, M., Ed., Pepperell, A., and S. Wenger,
            "Framework for Telepresence Multi-Streams", Work in
            Progress, draft-ietf-clue-framework-25, January 2016.
 [MULTI-RTP]
            Westerlund, M., Perkins, C., and J. Lennox, "Sending
            Multiple Types of Media in a Single RTP Session", Work in
            Progress, draft-ietf-avtcore-multi-media-rtp-session-13,
            December 2015.

Lennox, et al. Standards Track [Page 26] RFC 8108 Multiple Media Streams in an RTP Session March 2017

 [MULTI-STREAM-OPT]
            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",
            Work in Progress, draft-ietf-avtcore-rtp-multi-
            stream-optimisation-12, March 2016.
 [RFC3390]  Allman, M., Floyd, S., and C. Partridge, "Increasing TCP's
            Initial Window", RFC 3390, DOI 10.17487/RFC3390, October
            2002, <http://www.rfc-editor.org/info/rfc3390>.
 [RFC3551]  Schulzrinne, H. and S. Casner, "RTP Profile for Audio and
            Video Conferences with Minimal Control", STD 65, RFC 3551,
            DOI 10.17487/RFC3551, July 2003,
            <http://www.rfc-editor.org/info/rfc3551>.
 [RFC3556]  Casner, S., "Session Description Protocol (SDP) Bandwidth
            Modifiers for RTP Control Protocol (RTCP) Bandwidth",
            RFC 3556, DOI 10.17487/RFC3556, July 2003,
            <http://www.rfc-editor.org/info/rfc3556>.
 [RFC3830]  Arkko, J., Carrara, E., Lindholm, F., Naslund, M., and K.
            Norrman, "MIKEY: Multimedia Internet KEYing", RFC 3830,
            DOI 10.17487/RFC3830, August 2004,
            <http://www.rfc-editor.org/info/rfc3830>.
 [RFC4588]  Rey, J., Leon, D., Miyazaki, A., Varsa, V., and R.
            Hakenberg, "RTP Retransmission Payload Format", RFC 4588,
            DOI 10.17487/RFC4588, July 2006,
            <http://www.rfc-editor.org/info/rfc4588>.
 [RFC5104]  Wenger, S., Chandra, U., Westerlund, M., and B. Burman,
            "Codec Control Messages in the RTP Audio-Visual Profile
            with Feedback (AVPF)", RFC 5104, DOI 10.17487/RFC5104,
            February 2008, <http://www.rfc-editor.org/info/rfc5104>.
 [RFC5576]  Lennox, J., Ott, J., and T. Schierl, "Source-Specific
            Media Attributes in the Session Description Protocol
            (SDP)", RFC 5576, DOI 10.17487/RFC5576, June 2009,
            <http://www.rfc-editor.org/info/rfc5576>.
 [RFC6190]  Wenger, S., Wang, Y., Schierl, T., and A. Eleftheriadis,
            "RTP Payload Format for Scalable Video Coding", RFC 6190,
            DOI 10.17487/RFC6190, May 2011,
            <http://www.rfc-editor.org/info/rfc6190>.

Lennox, et al. Standards Track [Page 27] RFC 8108 Multiple Media Streams in an RTP Session March 2017

 [RFC6928]  Chu, J., Dukkipati, N., Cheng, Y., and M. Mathis,
            "Increasing TCP's Initial Window", RFC 6928,
            DOI 10.17487/RFC6928, April 2013,
            <http://www.rfc-editor.org/info/rfc6928>.
 [RFC7022]  Begen, A., Perkins, C., Wing, D., and E. Rescorla,
            "Guidelines for Choosing RTP Control Protocol (RTCP)
            Canonical Names (CNAMEs)", RFC 7022, DOI 10.17487/RFC7022,
            September 2013, <http://www.rfc-editor.org/info/rfc7022>.
 [RFC7160]  Petit-Huguenin, M. and G. Zorn, Ed., "Support for Multiple
            Clock Rates in an RTP Session", RFC 7160,
            DOI 10.17487/RFC7160, April 2014,
            <http://www.rfc-editor.org/info/rfc7160>.
 [RFC7667]  Westerlund, M. and S. Wenger, "RTP Topologies", RFC 7667,
            DOI 10.17487/RFC7667, November 2015,
            <http://www.rfc-editor.org/info/rfc7667>.
 [SDP-BUNDLE]
            Holmberg, C., Alvestrand, H., and C. Jennings,
            "Negotiating Media Multiplexing Using the Session
            Description Protocol (SDP)", Work in Progress,
            draft-ietf-mmusic-sdp-bundle-negotiation-36, October 2016.
 [Sim88]    Westerlund, M., "SIMULATION RESULTS FOR MULTI-STREAM",
            IETF 88 Proceedings, November 2013,
            <https://www.ietf.org/proceedings/88/slides/
            slides-88-avtcore-0.pdf>.
 [Sim92]    Westerlund, M., Lennox, J., Perkins, C., and Q. Wu,
            "Changes in RTP Multi-stream", IETF 92 Proceedings, March
            2015, <https://www.ietf.org/proceedings/92/slides/
            slides-92-avtcore-0.pdf>.

Lennox, et al. Standards Track [Page 28] RFC 8108 Multiple Media Streams in an RTP Session March 2017

Acknowledgments

 The authors like to thank Harald Alvestrand and everyone else who has
 been involved in the development of this document.

Authors' Addresses

 Jonathan Lennox
 Vidyo, Inc.
 433 Hackensack Avenue
 Seventh Floor
 Hackensack, NJ  07601
 United States of America
 Email: jonathan@vidyo.com
 Magnus Westerlund
 Ericsson
 Farogatan 2
 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: bill.wu@huawei.com
 Colin Perkins
 University of Glasgow
 School of Computing Science
 Glasgow  G12 8QQ
 United Kingdom
 Email: csp@csperkins.org

Lennox, et al. Standards Track [Page 29]

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