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Versions: 00 01 02 03 04 05 draft-ietf-avtext-rtp-stream-pause

Network Working Group                                           A. Akram
Internet-Draft                                                 B. Burman
Updates: 5104 (if approved)                                     Ericsson
Intended status: Standards Track                                 R. Even
Expires: August 18, 2014                             Huawei Technologies
                                                           M. Westerlund
                                                                Ericsson
                                                       February 14, 2014


                   RTP Media Stream Pause and Resume
              draft-westerlund-avtext-rtp-stream-pause-05

Abstract

   With the increased popularity of real-time multimedia applications,
   it is desirable to provide good control of resource usage, and users
   also demand more control over communication sessions.  This document
   describes how a receiver in a multimedia conversation can pause and
   resume incoming data from a sender by sending real-time feedback
   messages when using Real-time Transport Protocol (RTP) for real time
   data transport.  This document extends the Codec Control Messages
   (CCM) RTCP feedback package by explicitly allowing and describing
   specific use of existing CCM messages and adding a group of new real-
   time feedback messages used to pause and resume RTP data streams.
   This document updates RFC 5104.

Status of This Memo

   This Internet-Draft is submitted in full conformance with the
   provisions of BCP 78 and BCP 79.

   Internet-Drafts are working documents of the Internet Engineering
   Task Force (IETF).  Note that other groups may also distribute
   working documents as Internet-Drafts.  The list of current Internet-
   Drafts is at http://datatracker.ietf.org/drafts/current/.

   Internet-Drafts are draft documents valid for a maximum of six months
   and may be updated, replaced, or obsoleted by other documents at any
   time.  It is inappropriate to use Internet-Drafts as reference
   material or to cite them other than as "work in progress."

   This Internet-Draft will expire on August 18, 2014.








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Copyright Notice

   Copyright (c) 2014 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.

Table of Contents

   1.  Introduction  . . . . . . . . . . . . . . . . . . . . . . . .   3
   2.  Definitions . . . . . . . . . . . . . . . . . . . . . . . . .   4
     2.1.  Abbreviations . . . . . . . . . . . . . . . . . . . . . .   4
     2.2.  Terminology . . . . . . . . . . . . . . . . . . . . . . .   6
     2.3.  Requirements Language . . . . . . . . . . . . . . . . . .   7
   3.  Use Cases . . . . . . . . . . . . . . . . . . . . . . . . . .   7
     3.1.  Point to Point  . . . . . . . . . . . . . . . . . . . . .   7
     3.2.  RTP Mixer to Media Sender . . . . . . . . . . . . . . . .   8
     3.3.  RTP Mixer to Media Sender in Point-to-Multipoint  . . . .   9
     3.4.  Media Receiver to RTP Mixer . . . . . . . . . . . . . . .   9
     3.5.  Media Receiver to Media Sender Across RTP Mixer . . . . .  10
   4.  Design Considerations . . . . . . . . . . . . . . . . . . . .  10
     4.1.  Real-time Nature  . . . . . . . . . . . . . . . . . . . .  10
     4.2.  Message Direction . . . . . . . . . . . . . . . . . . . .  11
     4.3.  Apply to Individual Sources . . . . . . . . . . . . . . .  11
     4.4.  Consensus . . . . . . . . . . . . . . . . . . . . . . . .  11
     4.5.  Acknowledgements  . . . . . . . . . . . . . . . . . . . .  11
     4.6.  Retransmitting Requests . . . . . . . . . . . . . . . . .  12
     4.7.  Sequence Numbering  . . . . . . . . . . . . . . . . . . .  12
   5.  Relation to Other Solutions . . . . . . . . . . . . . . . . .  12
     5.1.  Signaling Technology Performance Comparison . . . . . . .  12
     5.2.  CCM TMMBR / TMMBN . . . . . . . . . . . . . . . . . . . .  20
     5.3.  SDP "inactive" Attribute  . . . . . . . . . . . . . . . .  21
     5.4.  Media Source Selection in SDP . . . . . . . . . . . . . .  21
     5.5.  Conclusion  . . . . . . . . . . . . . . . . . . . . . . .  22
   6.  Solution Overview . . . . . . . . . . . . . . . . . . . . . .  22
     6.1.  Expressing Capability . . . . . . . . . . . . . . . . . .  23
     6.2.  Requesting to Pause . . . . . . . . . . . . . . . . . . .  23
     6.3.  Media Sender Pausing  . . . . . . . . . . . . . . . . . .  25
     6.4.  Requesting to Resume  . . . . . . . . . . . . . . . . . .  26
     6.5.  TMMBR/TMMBN Considerations  . . . . . . . . . . . . . . .  27



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   7.  Participant States  . . . . . . . . . . . . . . . . . . . . .  27
     7.1.  Playing State . . . . . . . . . . . . . . . . . . . . . .  28
     7.2.  Pausing State . . . . . . . . . . . . . . . . . . . . . .  28
     7.3.  Paused State  . . . . . . . . . . . . . . . . . . . . . .  29
       7.3.1.  RTCP BYE Message  . . . . . . . . . . . . . . . . . .  29
       7.3.2.  SSRC Time-out . . . . . . . . . . . . . . . . . . . .  29
     7.4.  Local Paused State  . . . . . . . . . . . . . . . . . . .  30
   8.  Message Format  . . . . . . . . . . . . . . . . . . . . . . .  30
   9.  Message Details . . . . . . . . . . . . . . . . . . . . . . .  32
     9.1.  PAUSE . . . . . . . . . . . . . . . . . . . . . . . . . .  33
     9.2.  PAUSED  . . . . . . . . . . . . . . . . . . . . . . . . .  34
     9.3.  RESUME  . . . . . . . . . . . . . . . . . . . . . . . . .  34
     9.4.  REFUSE  . . . . . . . . . . . . . . . . . . . . . . . . .  35
     9.5.  Transmission Rules  . . . . . . . . . . . . . . . . . . .  36
   10. Signalling  . . . . . . . . . . . . . . . . . . . . . . . . .  36
     10.1.  Offer-Answer Use . . . . . . . . . . . . . . . . . . . .  39
     10.2.  Declarative Use  . . . . . . . . . . . . . . . . . . . .  40
   11. Examples  . . . . . . . . . . . . . . . . . . . . . . . . . .  40
     11.1.  Offer-Answer . . . . . . . . . . . . . . . . . . . . . .  41
     11.2.  Point-to-Point Session . . . . . . . . . . . . . . . . .  42
     11.3.  Point-to-multipoint using Mixer  . . . . . . . . . . . .  45
     11.4.  Point-to-multipoint using Translator . . . . . . . . . .  47
   12. IANA Considerations . . . . . . . . . . . . . . . . . . . . .  50
   13. Security Considerations . . . . . . . . . . . . . . . . . . .  51
   14. Contributors  . . . . . . . . . . . . . . . . . . . . . . . .  51
   15. Acknowledgements  . . . . . . . . . . . . . . . . . . . . . .  51
   16. References  . . . . . . . . . . . . . . . . . . . . . . . . .  51
     16.1.  Normative References . . . . . . . . . . . . . . . . . .  51
     16.2.  Informative References . . . . . . . . . . . . . . . . .  52
   Authors' Addresses  . . . . . . . . . . . . . . . . . . . . . . .  54

1.  Introduction

   As real-time communication attracts more people, more applications
   are created; multimedia conversation applications being one example.
   Multimedia conversation further exists in many forms, for example,
   peer-to-peer chat application and multiparty video conferencing
   controlled by central media nodes, such as RTP Mixers.

   Multimedia conferencing may involve many participants; each has its
   own preferences for the communication session, not only at the start
   but also during the session.  This document describes several
   scenarios in multimedia communication where a conferencing node or
   participant chooses to temporarily pause an incoming RTP [RFC3550]
   media stream from a specific source and later resume it when needed.
   The receiver does not need to terminate or inactivate the RTP session
   and start all over again by negotiating the session parameters, for
   example using SIP [RFC3261] with SDP Offer/Answer [RFC3264].



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   Centralized nodes, like RTP Mixers or MCUs, which either uses logic
   based on voice activity, other measurements, or user input could
   reduce the resources consumed in both the media sender and the
   network by temporarily pausing the media streams that aren't required
   by the RTP Mixer.  If the number of conference participants are
   greater than what the conference logic has chosen to present
   simultaneously to receiving participants, some participant media
   streams sent to the RTP Mixer may not need to be forwarded to any
   other participant.  Those media streams could then be temporarily
   paused.  This becomes especially useful when the media sources are
   provided in multiple encoding versions (Simulcast)
   [I-D.westerlund-avtcore-rtp-simulcast] or with Multi-Session
   Transmission (MST) of scalable encoding such as SVC [RFC6190].  There
   may be some of the defined encodings or combination of scalable
   layers that are not used all of the time.

   As the media streams required at any given point in time is highly
   dynamic in such scenarios, using the out-of-band signalling channel
   for pausing, and even more importantly resuming, a media stream is
   difficult due to the performance requirements.  Instead, the pause
   and resume signalling should be in the media plane and go directly
   between the affected nodes.  When using RTP [RFC3550] for media
   transport, using Extended RTP Profile for Real-time Transport Control
   Protocol (RTCP)-Based Feedback (RTP/AVPF) [RFC4585] appears
   appropriate.  No currently existing RTCP feedback message explicitly
   supports pausing and resuming an incoming media stream.  As this
   affects the generation of packets and may even allow the encoding
   process to be paused, the functionality appears to match Codec
   Control Messages in the RTP Audio-Visual Profile with Feedback (AVPF)
   [RFC5104] and it is proposed to define the solution as a Codec
   Control Message (CCM) extension.

   The Temporary Maximum Media Bitrate Request (TMMBR) message of CCM is
   used by video conferencing systems for flow control.  It is desirable
   to be able to use that method with a bitrate value of zero for pause
   and resume, whenever possible.

2.  Definitions

2.1.  Abbreviations

   3GPP:  3rd Generation Partnership Project

   AVPF:  Audio-Visual Profile with Feedback (RFC 4585)

   BGW:  Border Gateway

   CCM:  Codec Control Messages (RFC 5104)



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   CNAME:  Canonical Name (RTCP SDES)

   CSRC:  Contributing Source (RTP)

   FB:  Feedback (AVPF)

   FCI:  Feedback Control Information (AVPF)

   FIR:  Full Intra Refresh (CCM)

   FMT:  Feedback Message Type (AVPF)

   LTE:  Long-Term Evolution (3GPP)

   MCU:  Multipoint Control Unit

   MTU:  Maximum Transfer Unit

   PT:  Payload Type (RTP)

   RTP:  Real-time Transport Protocol (RFC 3550)

   RTCP:  Real-time Transport Control Protocol (RFC 3550)

   RTCP RR:  RTCP Receiver Report

   SDP:  Session Description Protocol (RFC 4566)

   SGW:  Signaling Gateway

   SIP:  Session Initiation Protocol (RFC 3261)

   SSRC:  Synchronization Source (RTP)

   SVC:  Scalable Video Coding

   TCP:  Transmission Control Protocol (RFC 793)

   TMMBR:  Temporary Maximum Media Bitrate Request (CCM)

   TMMBN:  Temporary Maximum Media Bitrate Notification (CCM)

   UA:  User Agent (SIP)

   UDP:  User Datagram Protocol (RFC 768)






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2.2.  Terminology

   In addition to following, the definitions from RTP [RFC3550], AVPF
   [RFC4585], CCM [RFC5104], and RTP Taxonomy
   [I-D.ietf-avtext-rtp-grouping-taxonomy] also apply in this document.

   Feedback Messages:  CCM [RFC5104] categorized different RTCP feedback
      messages into four types, Request, Command, Indication and
      Notification.  This document places the PAUSE and RESUME messages
      into Request category, PAUSED as Indication and REFUSE as
      Notification.

      PAUSE  Request from a media receiver to pause a stream

      RESUME  Request from a media receiver to resume a paused stream

      PAUSED  Indication from a media sender that a stream is paused

      REFUSE  Notification from a media sender that a PAUSE or RESUME
         request will not be honored

   Acknowledgement:  The confirmation from receiver to sender that the
      message has been received.

   Sender:  The RTP entity that sends an RTP Packet Stream.

   Receiver:  The RTP entity that receives an RTP Packet Stream.

   Mixer:  The intermediate RTP node which receives a Packet Stream from
      different nodes, combines them to make one stream and forwards to
      destinations, in the sense described in Topo-Mixer of RTP
      Topologies [I-D.ietf-avtcore-rtp-topologies-update].

   Participant:  A member which is part of an RTP session, acting as
      receiver, sender or both.

   Paused Sender:  An RTP sender that has stopped its transmission, i.e.
      no other participant receives its RTP transmission, either based
      on having received a PAUSE request, defined in this specification,
      or based on a local decision.

   Pausing Receiver:  An RTP receiver which sends a PAUSE request,
      defined in this specification, to other participant(s).

   Stream:  Used as a short term for Source Packet Stream, unless
      otherwise noted.





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2.3.  Requirements Language

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

3.  Use Cases

   This section discusses the main use cases for media stream pause and
   resume.

3.1.  Point to Point

   This is the most basic use case with an RTP session containing two
   end-points.  Each end-point sends one or more streams.

   +---+         +---+
   | A |<------->| B |
   +---+         +---+

                         Figure 1: Point to Point

   The usage of media stream pause in this use case is to temporarily
   halt media delivery of streams that the sender provides but the
   receiver does not currently use.  This can for example be due to
   minimized applications where the video stream is not actually shown
   on any display, and neither is it used in any other way, such as
   being recorded.

   In this case, since there is only a single receiver of the stream,
   pausing or resuming a stream does not impact anyone else than the
   sender and the single receiver of that stream.

   RTCWEB WG's use case and requirements document
   [I-D.ietf-rtcweb-use-cases-and-requirements] defines the following
   API requirements in Appendix A, used also by W3C WebRTC WG:

   A8 The Web API must provide means for the web application to mute/
      unmute a stream or stream component(s).  When a stream is sent to
      a peer mute status must be preserved in the stream received by the
      peer.

   A9 The Web API must provide means for the web application to cease
      the sending of a stream to a peer.

   This memo provides means to optimize transport usage by stop sending
   muted streams and start sending again when unmuting.




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3.2.  RTP Mixer to Media Sender

   One of the most commonly used topologies in centralized conferencing
   is based on the RTP Mixer [I-D.ietf-avtcore-rtp-topologies-update].
   The main reason for this is that it provides a very consistent view
   of the RTP session towards each participant.  That is accomplished
   through the Mixer originating its' own streams, identified by SSRC,
   and any media sent to the participants will be sent using those
   SSRCs.  If the Mixer wants to identify the underlying Media Sources
   for its' conceptual streams, it can identify them using CSRC.  The
   stream the Mixer provides can be an actual media mix of multiple
   Media Sources, but it might also be switching received streams as
   described in Sections 3.6-3.8 of
   [I-D.ietf-avtcore-rtp-topologies-update].

   +---+      +-----------+      +---+
   | A |<---->|           |<---->| B |
   +---+      |           |      +---+
              |   Mixer   |
   +---+      |           |      +---+
   | C |<---->|           |<---->| D |
   +---+      +-----------+      +---+

                    Figure 2: RTP Mixer in Unicast-only

   Which streams that are delivered to a given receiver, A, can depend
   on several things.  It can either be the RTP Mixer's own logic and
   measurements such as voice activity on the incoming audio streams.
   It can be that the number of sent Media Sources exceed what is
   reasonable to present simultaneously at any given receiver.  It can
   also be a human controlling the conference that determines how the
   media should be mixed; this would be more common in lecture or
   similar applications where regular listeners may be prevented from
   breaking into the session unless approved by the moderator.  The
   streams may also be part of a Simulcast
   [I-D.westerlund-avtcore-rtp-simulcast] or scalable encoded (for
   Multi-Stream Transmission) [RFC6190], thus providing multiple
   versions that can be delivered by the media sender.  These examples
   indicate that there are numerous reasons why a particular stream
   would not currently be in use, but must be available for use at very
   short notice if any dynamic event occurs that causes a different
   stream selection to be done in the Mixer.

   Because of this, it would be highly beneficial if the Mixer could
   request to pause a particular stream from being delivered to it.  It
   also needs to be able to resume delivery with minimal delay.





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   Just as for point-to-point (Section 3.1), there is only a single
   receiver of the stream, the RTP Mixer, and pausing or resuming a
   stream does not affect anyone else than the sender and single
   receiver of that stream.

3.3.  RTP Mixer to Media Sender in Point-to-Multipoint

   This use case is similar to the previous section, however the RTP
   Mixer is involved in three domains that need to be separated; the
   Multicast Network (including participants A and C), participant B,
   and participant D. The difference from above is that A and C share a
   multicast domain, which is depicted below.

              +-----+
   +---+     /       \     +-----------+      +---+
   | A |<---/         \    |           |<---->| B |
   +---+   /   Multi-  \   |           |      +---+
          +    Cast     +->|   Mixer   |
   +---+   \  Network  /   |           |      +---+
   | C |<---\         /    |           |<---->| D |
   +---+     \       /     +-----------+      +---+
              +-----+

                Figure 3: RTP Mixer in Point-to-Multipoint

   If the RTP Mixer pauses a stream from A, it will not only pause the
   stream towards itself, but will also stop the stream from arriving to
   C, which C is heavily impacted by, might not approve of, and should
   thus have a say on.

   If the Mixer resumes a paused stream from A, it will be resumed also
   towards C. In this case, if C is not interested it can simply ignore
   the stream and is not impacted as much as above.

   In this use case there are several receivers of a stream and special
   care must be taken as not to pause a stream that is still wanted by
   some receivers.

3.4.  Media Receiver to RTP Mixer

   An end-point in Figure 2 could potentially request to pause the
   delivery of a given media stream.  Possible reasons include the ones
   in the point to point case (Section 3.1) above.

   When the RTP Mixer is only connected to individual unicast paths, the
   use case and any considerations are identical to the point to point
   use case.




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   However, when the end-point requesting media stream pause is
   connected to the RTP Mixer through a multicast network, such as A or
   C in Figure 3, the use case instead becomes identical to the one in
   Section 3.3, only with reverse direction of the streams and pause/
   resume requests.

3.5.  Media Receiver to Media Sender Across RTP Mixer

   An end-point, like A in Figure 2, could potentially request to pause
   the delivery of a given media stream, like one of B's, over any of
   the SSRCs used by the Mixer by sending a pause request for the CSRC
   identifying the media stream.  However, the authors are of the
   opinion that this is not a suitable solution, for several reasons:

   1.  The Mixer might not include CSRC in it's stream indications.

   2.  An end-point cannot rely on the CSRC to correctly identify the
       media stream to be paused when the delivered media is some type
       of mix.  A more elaborate media stream identification solution is
       needed to support this in the general case.

   3.  The end-point cannot determine if a given media stream is still
       needed by the RTP Mixer to deliver to another session
       participant.

   Due to the above reasons, we exclude this use case from further
   consideration.

4.  Design Considerations

   This section describes the requirements that this specification needs
   to meet.

4.1.  Real-time Nature

   The first section (Section 1) of this specification describes some
   possible reasons why a receiver may pause an RTP sender.  Pausing and
   resuming is time-dependent, i.e. a receiver may choose to pause an
   RTP stream for a certain duration, after which the receiver may want
   the sender to resume.  This time dependency means that the messages
   related to pause and resume must be transmitted to the sender in
   real-time in order for them to be purposeful.  The pause operation is
   arguably not very time critical since it mainly provides a reduction
   of resource usage.  Timely handling of the resume operation is
   however likely to directly impact the end-user's perceived quality
   experience, since it affects the availability of media that the user
   expects to receive more or less instantly.




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4.2.  Message Direction

   It is the responsibility of a media receiver, who wants to pause or
   resume a media stream from the sender(s), to transmit PAUSE and
   RESUME messages.  A media sender who likes to pause itself, can
   simply do it.  Any indication that an RTP media stream is paused is
   the responsibility of the RTP media stream sender and may in some
   cases not even be needed by the media stream receiver.

4.3.  Apply to Individual Sources

   The PAUSE and RESUME messages apply to single RTP media streams
   identified by their SSRC, which means the receiver targets the
   sender's SSRC in the PAUSE and RESUME requests.  If a paused sender
   starts sending with a new SSRC, the receivers will need to send a new
   PAUSE request in order to pause it.  PAUSED indications refer to a
   single one of the sender's own, paused SSRC.

4.4.  Consensus

   An RTP media stream sender should not pause an SSRC that some
   receiver still wishes to receive.  The reason is that in RTP
   topologies where the media stream is shared between multiple
   receivers, a single receiver on that shared network, independent of
   it being multicast, a mesh with joint RTP session or a transport
   Translator based, must not single-handedly cause the media stream to
   be paused without letting all other receivers to voice their opinions
   on whether or not the stream should be paused.  A consequence of this
   is that a newly joining receiver, for example indicated by an RTCP
   Receiver Report containing both a new SSRC and a CNAME that does not
   already occur in the session, firstly needs to learn the existence of
   paused streams, and secondly should be able to resume any paused
   stream.  Any single receiver wanting to resume a stream should also
   cause it to be resumed.

4.5.  Acknowledgements

   RTP and RTCP does not guarantee reliable data transmission.  It uses
   whatever assurance the lower layer transport protocol can provide.
   However, this is commonly UDP that provides no reliability
   guarantees.  Thus it is possible that a PAUSE and/or RESUME message
   transmitted from an RTP end-point does not reach its destination,
   i.e. the targeted RTP media stream sender.  When PAUSE or RESUME
   reaches the RTP media stream sender and are effective, i.e., an
   active media sender pauses, or a resuming have media data to
   transmit, it is immediately seen from the arrival or non-arrival of
   RTP packets for that RTP media stream.  Thus, no explicit
   acknowledgements are required in this case.



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   In some cases when a PAUSE or RESUME message reaches the media
   sender, it will not be able to pause or resume the stream due to some
   local consideration, for example lack of data to transmit.  This
   error condition, a negative acknowledgement, may be needed to avoid
   unnecessary retransmission of requests (Section 4.6).

4.6.  Retransmitting Requests

   When the media stream is not affected as expected by a PAUSE or
   RESUME request, the request may have been lost and the sender of the
   request will need to retransmit it.  The retransmission should take
   the round trip time into account, and will also need to take the
   normal RTCP bandwidth and timing rules applicable to the RTP session
   into account, when scheduling retransmission of feedback.

   When it comes to resume requests that are more time critical, the
   best resume performance may be achieved by repeating the request as
   often as possible until a sufficient number have been sent to reach a
   high probability of request delivery, or the media stream gets
   delivered.

4.7.  Sequence Numbering

   A PAUSE request message will need to have a sequence number to
   separate retransmissions from new requests.  A retransmission keeps
   the sequence number unchanged, while it is incremented every time a
   new PAUSE request is transmitted that is not a retransmission of a
   previous request.

   Since RESUME always takes precedence over PAUSE and are even allowed
   to avoid pausing a stream, there is a need to keep strict ordering of
   PAUSE and RESUME.  Thus, RESUME needs to share sequence number space
   with PAUSE and implicitly references which PAUSE it refers to.  For
   the same reasons, the explicit PAUSED indication also needs to share
   sequence number space with PAUSE and RESUME.

5.  Relation to Other Solutions

   This section compares other possible solutions to achieve a similar
   functionality, along with motivations why the current solution is
   chosen.

5.1.  Signaling Technology Performance Comparison

   Editor's note:  This section is related to the motivation for
      selecting RTCP as signaling technology rather than SIP/SDP and
      should be considered to be removed or at least significantly
      reduced if and when this draft is adopted as a working group



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      draft, since there now seems to be consensus that RTCP is the
      preferred technology.

   This section contains what is thought to be a realistic estimate of
   one-way data transmission times for signaling implementing
   functionalities of this specification.

   Two signaling protocols are compared.  SIP is chosen to represent
   signaling in the control plane and RTCP is chosen to represent
   signaling in the media plane.  For the sake of the comparison, each
   of these two protocols are listed with one favorable and one
   unfavorable condition to give the reader a hint of what range of
   delays that can be expected.  The favorable condition is chosen as
   good as possible, while still realistic.  The unfavorable condition
   is also chosen to be realistically occurring, and is not the worst
   possible or imaginable.  Actual delays can in most cases be expected
   to lie somewhere between those two values.

   It would also be possible to include a signaling protocol using a
   some dedicated signaling channel, separate from SIP and RTCP, into
   the comparison.  Such signaling protocol can be expected to show
   performance somewhere in the range covered by the SIP and RTCP
   comparison below.  The protocol can either use UDP as transport, like
   RTCP, or it can use TCP, like SIP, when the messages becomes too
   large for the MTU.  The data sent on such channel can either be text
   based, in which case the amount of data can be similar to SIP, or it
   can be binary, in which case the amount of data can be similar to
   RTCP.  Therefore, the dedicated signaling channel case is not
   described further in this specification.

   Two different access technologies are compared:

   o  Wired, fixed access is chosen as a representative low-delay
      alternative.

   o  Mobile wireless access according to 3GPP LTE [TS36.201], also
      known as "4G", is chosen as a representative high-delay
      alternative.

   NOTE: LTE is at the time of writing the most recent and best
   performing mobile wireless access.  If an earlier mobile wireless
   access was to be used instead, the estimated transmission times would
   be considerably increased.  For example, it is estimated that using
   3GPP HSPA [TS25.308] (evolved 3G, just previous to LTE) would
   increase RTCP signaling times somewhat and significantly increase
   signaling times for SIP, although those estimates are too preliminary
   to provide any values here.




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   The target scenario includes two UA, residing in two different
   provider's (operator's) network.  Those networks are assumed to be
   geographically close, that is no inter-continental transmission
   delays are included in the estimates.

   Three signaling alternatives are compared:

   o  Wireless UA to wireless UA, including two wireless links, uplink
      and downlink.

   o  Wireless UA to media server (MCU), including a single wireless
      uplink.

   o  Media server (MCU) to wireless UA, including a single wireless
      downlink.

   The reason to include separate results for wireless uplink and
   downlink is that delay times can differ significantly.

   The targeted topology is outlined in the following figure.

                  Provider A's network  .  Provider B's network
                                        .
   +-----+ SIP +------+ SIP +-------+  SIP  +-------+ SIP +-----+
   |Proxy|<--->| AS A |<--->| SGW A |<--.-->| SGW B |<--->|Proxy|
   +-----+     +------+     +-------+   .   +-------+     +-----+
       ^          ^                     .                    ^
       |          | SIP/H.248           .                    |
       |          v                     .                    |
   SIP |       +-----+ RTCP +-------+ RTCP  +-------+    SIP |
       |       | MCU |<---->| BGW A |<--.-->| BGW B |        |
       |       +-----+      +-------+   .   +-------+        |
       v         ^                      .       ^            v
   +------+     / RTCP                  .        \  RTCP  +------+
   | UA A |<---+                        .         +------>| UA B |
   +------+                             .                 +------+

                  Figure 4: Comparison Signaling Topology

   In the figure above, UA is a SIP User Agent, Proxy is a SIP Proxy, AS
   is an Application Server, MCU is a Multipoint Conference Unit, SGW a
   Signaling GateWay, and BGW a media Border GateWay.

   It can be noted that when either one or both UAs use call forwarding
   or have roamed into yet another provider's network, several more
   signaling path nodes and a few more media path nodes could be
   included in the end-to-end signaling path.




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   The MCU is assumed to be located in one of the provider's network.
   Signaling delays between the MCU and a UA are presented as the
   average of MCU and UA being located in the same and different
   provider's networks.

   These assumptions are used for SIP signaling:

   o  A SIP UPDATE is used within an established session to dynamically
      impact individual streams to achieve the pause and resume
      functionality.  The offer and answer SDP contains one audio and
      one video media, compliant with what is suggested in 3GPP MTSI
      [TS26.114], with the addition of SDP feedback message indication
      outlined in this specification (Section 10).  A more complex media
      session with more streams would significantly add to the SDP size.

   o  UDP is used as transport, except when risking to exceed MTU, in
      which case TCP is used instead.  This is evaluated on a per-
      message basis.

   o  Only SIP forward direction is included in the delay estimate, that
      is, delays needed to receive a response such as 200 OK are not
      included.

   o  Favorable case:

      *  SIP SigComp [RFC5049] in dynamic mode is used for SIP and SDP
         signaling on the mobile link, reducing the SIP message size to
         approximately 1/3 of the original size.

   o  Unfavorable case:

      *  SIP message is not compressed on the mobile link.

      *  SIP signaling on the mobile link uses a dedicated mobile
         wireless access radio channel that was idle for some time, has
         entered low power state and thus has to be re-established by
         radio layer signaling before any data can be sent.

   These assumptions are used for RTCP signaling:

   o  A minimal compound RTCP feedback packet is used, including one SR
      and one SDES with only the CNAME item present, with the addition
      of the feedback message outlined in Section 8.

   o  RTCP bandwidth is chosen based on a 200 kbit/s session, which is
      considered to be a low bandwidth for media that would be worth
      pausing, and using the default 5% of this for RTCP traffic results




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      in 10 kbit/s. This low bandwidth makes RTCP scheduling delays be a
      significant factor in the unfavorable case.

   o  Since there are random delay factors in RTCP transmission, the
      expected, most probable value is used in the estimates.

   o  The mobile wireless access channel used for RTCP will always be
      active, that is there will be sufficient data to send at any time
      such that the radio channel will never have to be re-established.
      This is considered reasonable since it is assumed that the same
      channel is not only used for the messages defined in this
      specification, but also for other RTP and RTCP data.

   o  Favorable case:

      *  It is assumed that AVPF Early or Immediate mode can always be
         used for the signaling described in this specification, since
         such signaling will be small in size and only occur
         occasionally in RTCP time scale.

      *  Early mode does not use dithering of send times (T_dither_max
         is set to 0), that is, sender and receiver of the message are
         connected point-to-point.  It can be noted that in case of a
         multiparty session where multiple end-points can see each
         others' messages, and unless the number of end-points is very
         large, it is very unlikely that more than a single end-point
         has the desire to send the same message (defined in this
         specification) as another end-point, and at almost exactly the
         same time.  It is therefore arguably not very meaningful for
         messages in this specification to try to do feedback
         suppression by using a non-zero T_dither_max, even in
         multiparty sessions, but AVPF does not allow for any exemption
         from that rule.

      *  Reduced-size RTCP is used, which is considered appropriate for
         the type of messages defined in this specification.

      *  RTP/RTCP header compression [RFC5225] is not used, not even on
         the mobile link.

   o  Unfavorable case:

      *  The expected, regular AVPF RTCP interval is used, including an
         expected value for timer re-consideration.

      *  A full, not reduced-size, minimal compound RTCP feedback packet
         without header compression is always used.  No reduction of
         scheduling delays from the use of reduced-size RTCP is included



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         in the evaluation, since that would also require a reasonable
         estimate of the mix of compound and non-compound RTCP, which
         was considered too difficult for this study.  The given
         unfavorable delays are thus an over-estimate compared to a more
         realistic case.

   Common to both SIP and RTCP signaling estimates is that no UA
   processing delays are included.  The reason for that decision is that
   processing delays are highly implementation and UA dependent.  It is
   expected that wireless UA will be more limited than fixed UA by
   processing, but they are also constantly and quickly improving so any
   estimate will very quickly be outdated.  More realistic estimates
   will however have to add such delays, which can be expected to be in
   the order of a few to a few tens of milliseconds.  It is expected
   that SIP will be more penalized than RTCP by including processing
   delays, since it has larger and more complex messages.  The
   processing may also include SigComp [RFC5049] compression and
   decompression in the favorable cases.

   As a partial result, the message sizes can be compared, based on the
   messages defined in this specification (Section 8) and a SIP UPDATE
   with contents (Section 10) as discussed above.  Favorable and
   unfavorable message sizes are presented as stacked bars in the figure
   below.  Message sizes include IPv4 headers but no lower layer data,
   are rounded to the nearest 25 bytes, and the bars are to scale.

            250       500       750      1000      1250      1500 [byte]
   +---------+---------+---------+---------+---------+---------+--> Size
   |
   +-+--+
   | |50| 125 RTCP
   +-+--+---------------+--------------------------------------------+
   | SIP                | 525                                   1650 |
   +--------------------+--------------------------------------------+
   |


                     Figure 5: Message Size Comparison

   The signaling delay results of the study are summarized in the
   following two figures.  Favorable and unfavorable values are
   presented as stacked bars.  Since there are many factors that impact
   the calculations, including some random processes, there are
   uncertainty in the calculations and delay values are thus rounded to
   nearest 5 ms.  The bars are to scale.






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             50       100       150       200       250       300   [ms]
   +---------+---------+---------+---------+---------+---------+---> t
   |
   | Wireless UA to Wireless UA
   +---------------------+--------------------------------------+
   | SIP                 | 110                                  | 305
   +-----+---------------+-----------------------------+--------+
   |RTCP | 30                                          | 260
   +-----+---------------------------------------------+
   |
   | Wireless UA to MCU
   +-------------+--+
   | SIP         |70| 85
   +----+--------+--+--------------------------+
   |RTCP| 25                                   | 225
   +----+--------------------------------------+
   |
   | MCU to Wireless UA
   +--------------+-----------------------------------+
   | SIP          | 75                                | 255
   +---+----------+------------------------------+----+
   |   | 20 RTCP                                 | 230
   +---+-----------------------------------------+
   |


           Figure 6: Mobile Access Transmission Delay Comparison

   As can be seen, RTCP has a smaller signaling delay than SIP in a
   majority of cases for this mobile access.  Non-favorable RTCP is
   however always worse than favorable SIP.

   The UA to MCU signaling corresponds to the use case in Section 3.4.
   The reason that unfavorable SIP is more beneficial than unfavorable
   RTCP in this case comes from the fact that latency is fairly short to
   re-establish an uplink radio channel (as was assumed needed for
   unfavorable SIP), while unfavorable RTCP does not benefit from this
   since the delay is mainly due to RTCP Scheduling.

   The MCU to UA signaling corresponds to the use case in Section 3.2.
   It has an unfavorable SIP signaling case with much longer delay than
   UA to MCU above, because the mixer cannot re-establish a downlink
   radio channel as quickly as the UA can establish an uplink.  This
   case is applicable when an MCU wants to resume a paused stream, which
   is likely the most delay sensitive functionality, as discussed in
   Section 4.1.





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   Below are the same cases for fixed access depicted.  Although delays
   are generally shorter, scales are kept the same for easy comparison
   with the previous figure.

             50       100       150       200       250       300   [ms]
   +---------+---------+---------+---------+---------+---------+---> t
   |
   | Fixed UA to Fixed UA
   +------------+
   | SIP        | 65
   +----+-------+---------------------------+
   |RTCP| 25                                | 205
   +----+-----------------------------------+
   |
   | Fixed UA to MCU
   +---------+
   | SIP     | 50
   +---+-----+-----------------------+
   |   | 15 RTCP                     | 200
   +---+-----------------------------+
   |
   | MCU to Fixed UA
   +---------+
   | SIP     | 50
   +---+-----+-----------------------+
   |   | 15 RTCP                     | 200
   +---+-----------------------------+
   |


           Figure 7: Fixed Access Transmission Delay Comparison

   For fixed access, favorable RTCP is still significantly better than
   SIP, but unfavorable RTCP is significantly worse than SIP.  There is
   no difference between favorable and unfavorable SIP, since in fixed
   access there is no channel that needs to be re-established.

   Regarding the unfavorable values above, it should be possible with
   reasonable effort to design UA and network nodes that show favorable
   delays in a majority of cases.

   For SIP, the major delays in the unfavorable cases above comes from
   re-establishing a radio bearer that has entered low power state due
   to inactivity, and large size SIP messages.  The inactivity problem
   can be removed by using for example SIP keep-alive [RFC5626], at the
   cost of reduced battery life to keep the signaling radio bearer
   active, and some very minimal amount of extra data transmission.  The
   large SIP messages can to some extent be reduced by SIP SigComp



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   [RFC5049].  It may however prove harder to reduce delays that comes
   from forwarding the SDP many times between different signaling nodes.

   For RTCP, the major delays comes from low RTCP bandwidth and not
   being able to use Immediate or Early mode, including use of timer re-
   consideration.  UAs and network nodes can explicitly allocate an
   appropriate amount of RTCP bandwidth through use of the b=RS and b=RR
   RTCP bandwidth SDP attributes [RFC3556].  For RTP media streams of
   higher bandwidth than the 200 kbit/s used in this comparison, which
   will be even more interesting to pause, RTCP bandwidth will per
   default also be higher, significantly reducing the signaling delays.
   For example, using a 1000 kbit/s media stream instead of a 200 kbit/s
   stream will reduce the unfavorable RTCP delays from 260 ms to 115 ms
   for Wireless-Wireless, from 225 ms to 80 ms for Wireless-MCU, and
   from 230 ms to 80 ms for MCU-Wireless.

5.2.  CCM TMMBR / TMMBN

   The Codec Control Messages specification [RFC5104] contains two
   messages, Temporary Maximum Media Bitrate Request (TMMBR) and
   Temporary Maximum Media Bitrate Notification (TMMBN), which could
   provide some of the necessary functionality.  TMMBR with a bitrate
   value of 0 could effectively constitute a PAUSE request and TMMBN 0
   could effectively be a PAUSED indication, and there are already
   implementations making use of TMMBR 0 in this way.  It is possible to
   signal per SSRC (Section 4.3) and using the media path for signaling
   (AVPF) [RFC4585] will in most cases provide the shortest achievable
   signaling delay (Section 4.1).  However, in some cases the defined
   semantics for TMMBR differ from what is required for PAUSE.

   When there is only a single receiver of a media stream, TMMBR 0 and
   PAUSE are effectively identical.

   When there are several receivers of the same media stream, the stream
   must not be paused until there are no receiver that desires to
   receive it (Section 4.4), for example there is no disapproving RESUME
   for a PAUSE.  In the presence of several simultaneous receivers, the
   TMMBR semantics is the opposite; the first media receiver that sends
   TMMBR 0 will pause the stream for all receivers.

   When there is only a single receiver of a media stream that is
   paused, TMMBR with a bitrate greater than 0 can effectively function
   as a RESUME, resuming the media stream immediately as needed
   (Section 4.4).

   For the case of multiple simultaneous receivers, TMMBR specifies to
   use a guard period when increasing the bandwidth.  In this case,
   TMMBR/TMMBN semantics (Section 4.2.1.2 of [RFC5104]) requires a media



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   sender to wait 2*RTT+T_dither_max after having sent a TMMBN,
   indicating the intention to increase the bandwidth, before it
   actually increases its bandwidth usage.  The RTT is specified to be
   the longest the media sender knows in the RTP session.  So, there is
   both the delay between the media sender receiving the TMMBR until it
   can send a TMMBN, and the above delay for the guard period before the
   media sender are allowed to resume transmission.  This delay before
   resuming transmission is the most time critical operation in this
   solution, making use of TMMBR as RESUME according to the defined
   semantics infeasible in practice when there are multiple simultaneous
   media stream receivers.

5.3.  SDP "inactive" Attribute

   In SDP [RFC4566], an "inactive" attribute is defined on media level
   and session level.  The attribute is intended to be used to put media
   "on hold", either at the beginning of a session or as a result of
   session re-negotiation [RFC3264], for example using SIP re-INVITE
   [RFC3261], possibly in combination with ITU-T H.248 media gateway
   control.

   This attribute is only possible to specify with media level
   resolution, is not possible to signal per individual media stream
   (SSRC) (Section 4.3), and is thus not usable for RTP sessions
   containing more than a single SSRC.

   There is a per-ssrc attribute defined in [RFC5576], but that does
   currently not allow to set an individual stream (SSRC) inactive.

   Using "inactive" does thus not provide sufficient functionality for
   the purpose of this specification.

5.4.  Media Source Selection in SDP

   There is a draft that selects sources based on SDP
   [I-D.lennox-mmusic-sdp-source-selection] information.  It builds on
   the per-ssrc attribute [RFC5576] discussed above (Section 5.3).

   The semantics differ between selecting a Media Source and pause /
   resume for a stream in topologies other than point-to-point.  For
   example, in RTP Receiver to Mixer (Section 3.4), pausing a stream
   (SSRC) from the mixer should stop it being received altogether, while
   excluding a stream (CSRC) from the mix would just avoid that specific
   Media Source being included in the stream from the mixer.  There is a
   similar difference between resuming a stream (SSRC) from the mixer
   and allowing a Media Source (CSRC) to be included in the mix again.
   This suffers from a lack of functionality for consensus (Section 4.4)




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   and would likely also suffer from lower real-time performance
   (Section 4.1).

5.5.  Conclusion

   As can be seen from Section 5.1, using SIP and SDP to carry pause and
   resume information means that it will need to traverse the entire
   signaling path to reach the signaling destination (either the remote
   end-point or the entity controlling the RTP Mixer), across any
   signaling proxies that potentially also has to process the SDP
   content to determine if they are expected to act on it.  The amount
   of bandwidth required for a SIP/SDP-based signaling solution is in
   the order of at least 10 times more than an RTCP-based solution.

   Especially for UA sitting on mobile wireless access, this will risk
   introducing delays that are too long (Section 4.1) to provide a good
   user experience, and the bandwidth cost may also be considered
   infeasible compared to an RTCP-based solution.

   As seen in the same section, the RTCP data is sent through the media
   path, which is likely shorter (contains fewer intermediate nodes)
   than the signaling path but may anyway have to traverse a few
   intermediate nodes.  The amount of processing and buffering required
   in intermediate nodes to forward those RTCP messages is however
   believed to be significantly less than for intermediate nodes in the
   signaling path.

   Based on those reasons, RTCP is proposed as signaling protocol for
   the pause and resume functionality.  Much of the wanted functionality
   can in a point-to-point case be achieved with the existing TMMBR/
   TMMBN CCM messages [RFC5104], but they cannot be used when the media
   stream is sent to multiple simultaneous receivers.

6.  Solution Overview

   The proposed solution implements PAUSE and RESUME functionality based
   on sending AVPF RTCP feedback messages from any RTP session
   participant that wants to pause or resume a media stream targeted at
   the media stream sender, as identified by the sender SSRC.

   It is proposed to re-use CCM TMMBR and TMMBN [RFC5104] to the extent
   possible, and to define a small set of new RTCP feedback messages
   where new semantics is needed.  Considerations that that apply when
   using TMMBR/TMMBN for pause and resume purposes are also described.

   A single Feedback message specification is used to implement the new
   messages.  The message consists of a number of Feedback Control
   Information (FCI) blocks, where each block can be a PAUSE request, a



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   RESUME request, PAUSED indication, a REFUSE response, or an extension
   to this specification.  This structure allows a single feedback
   message to handle pause functionality on a number of media streams.

   The PAUSED functionality is also defined in such a way that it can be
   used standalone by the media sender to indicate a local decision to
   pause, and inform any receiver of the fact that halting media
   delivery is deliberate and which RTP packet was the last transmitted.

   This section is intended to be explanatory and therefore
   intentionally contains no mandatory statements.  Such statements can
   instead be found in other parts of this specification.

6.1.  Expressing Capability

   An end-point can use an extension to CCM SDP signaling to declare
   capability to understand the messages defined in this specification.
   Capability to understand PAUSED indication is defined separately from
   the others to support partial implementation, which is specifically
   believed to be feasible for the RTP Mixer to Media Sender use case
   (Section 3.2).

   For the case when TMMBR/TMMBN are used for pause and resume purposes,
   it is possible to explicitly express joint support for TMMBR and
   TMMBN, but not for TMMBN only.

6.2.  Requesting to Pause

   An RTP media stream receiver can choose to request PAUSE at any time,
   subject to AVPF timing rules.  This also applies when using TMMBR 0
   in the point-to-point case.

   The PAUSE request contains a PauseID, which is incremented by one (in
   modulo arithmetic) with each PAUSE request that is not a re-
   transmission.  The PauseID is scoped by and thus a property of the
   targeted RTP media stream (SSRC).

   When a non-paused RTP media stream sender receives the PAUSE request,
   it continues to send media while waiting for some time to allow other
   RTP media stream receivers in the same RTP session that saw this
   PAUSE request to disapprove by sending a RESUME (Section 6.4) for the
   same stream and with the same PauseID as in the disapproved PAUSE.
   If such disapproving RESUME arrives at the RTP media stream sender
   during the wait period before the stream is paused, the pause is not
   performed.  In point-to-point configurations, the wait period may be
   set to zero.  Using a wait period of zero is also appropriate when
   using TMMBR 0 and in line with the semantics for that message.




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   If the RTP media stream sender receives further PAUSE requests with
   the available PauseID while waiting as described above, those
   additional requests are ignored.

   If the PAUSE request or TMMBR 0 is lost before it reaches the RTP
   media stream sender, it will be discovered by the RTP media stream
   receiver because it continues to receive the RTP media stream.  It
   will also not see any PAUSED indication (Section 6.3) or TMMBN 0 for
   the stream.  The same condition can be caused by the RTP media stream
   sender having received a disapproving RESUME from a media stream
   receiver A for a PAUSE request sent by a media stream sender B, but
   that the PAUSE sender (B) did not receive the RESUME (from A) and may
   instead think that the PAUSE was lost.  In both cases, a PAUSE
   request can be re-transmitted using the same PauseID.  If using TMMBR
   0 the request MAY be re-transmitted when the requestor fails to
   receive a TMMBN 0 confirmation.

   If the pending stream pause is aborted due to a disapproving RESUME,
   the PauseID from the disapproved PAUSE is invalidated by the RESUME
   and any new PAUSE must use an incremented PauseID (in modulo
   arithmetic) to be effective.

   An RTP media stream sender receiving a PAUSE not using the available
   PauseID informs the RTP media stream receiver sending the ineffective
   PAUSE of this condition by sending a REFUSE response that contains
   the next available PauseID value.  This REFUSE also informs the RTP
   media stream receiver that it is probably not feasible to send
   another PAUSE for some time, not even with the available PauseID,
   since there are other RTP media stream receivers that wish to receive
   the stream.

   A similar situation where an ineffective PauseID is chosen can appear
   when a new RTP media stream receiver joins a session and wants to
   PAUSE a stream, but does not yet know the available PauseID to use.
   The REFUSE response will then provide sufficient information to
   create a valid PAUSE.  The required extra signaling round-trip is not
   considered harmful, since it is assumed that pausing a stream is not
   time-critical (Section 4.1).

   There may be local considerations making it impossible or infeasible
   to pause the stream, and the RTP media stream sender can then respond
   with a REFUSE.  In this case, if the used PauseID would otherwise
   have been effective, the REFUSE contains the same PauseID as in the
   PAUSE request, and the PauseID is kept as available.

   If the RTP media stream sender receives several identical PAUSE for
   an RTP media stream that was already at least once responded with
   REFUSE and the condition causing REFUSE remains, those additional



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   REFUSE should be sent with regular RTCP timing.  A single REFUSE can
   respond to several identical PAUSE requests.

6.3.  Media Sender Pausing

   An RTP media stream sender can choose to pause the stream at any
   time.  This can either be as a result of receiving a PAUSE, or be
   based on some local sender consideration.  When it does, it sends a
   PAUSED indication, containing the available PauseID.  If the stream
   was paused by a TMMBR 0, TMMBN 0 is used as PAUSED indication.  What
   is said on PAUSED in the rest of this paragraph apply also to the use
   of TMMBN 0, except for PAUSED message parameters.  Note that PauseID
   is incremented when pausing locally (without having received a
   PAUSE).  It also sends the PAUSED indication in the next two regular
   RTCP reports, given that the pause condition is then still effective.

   The RTP media stream sender may want to apply some local
   consideration to exactly when the stream is paused, for example
   completing some media unit or a forward error correction block,
   before pausing the stream.

   The PAUSED indication also contains information about the RTP
   extended highest sequence number when the pause became effective.
   This provides RTP media stream receivers with first hand information
   allowing them to know whether they lost any packets just before the
   stream paused or when the stream is resumed again.  This allows RTP
   media stream receivers to quickly and safely take into account that
   the stream is paused, in for example retransmission or congestion
   control algorithms.

   If the RTP media stream sender receives PAUSE requests with the
   available PauseID while the stream is already paused, those requests
   are ignored.

   As long as the stream is being paused, the PAUSED indication MAY be
   sent together with any regular RTCP SR or RR.  Including PAUSED in
   this way allows RTP media stream receivers joining while the stream
   is paused to quickly know that there is a paused stream, what the
   last sent extended RTP sequence number was, and what the next
   available PauseID is to be able to construct valid PAUSE and RESUME
   requests at a later stage.

   When the RTP media stream sender learns that a new end-point has
   joined the RTP session, for example by a new SSRC and a CNAME that
   was not previously seen in the RTP session, it should send PAUSED
   indications for all its paused streams at its earliest opportunity.
   It should in addition continue to include PAUSED indications in at
   least two regular RTCP reports.



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6.4.  Requesting to Resume

   An RTP media stream receiver can request to resume a stream with a
   RESUME request at any time, subject to AVPF timing rules.  If the
   stream was paused with TMMBR 0, resuming the stream is made with
   TMMBR containing a bitrate value larger than 0.  The bitrate value
   used when resuming after a PAUSE with TMMBR 0 is either according to
   known limitations, or the configured maximum for the stream or
   session.  What is said on RESUME in the rest of this paragraph apply
   also to the use of TMMBR with a bitrate value larger than 0, except
   for RESUME message parameters.

   The RTP media stream receiver must include the available PauseID in
   the RESUME request for it to be effective.

   A pausing RTP media stream sender that receives a RESUME including
   the correct available PauseID resumes the stream at the earliest
   opportunity.  Receiving RESUME requests for a stream that is not
   paused does not require any action and can be ignored.

   There may be local considerations, for example that the media device
   is not ready, making it temporarily impossible to resume the stream
   at that point in time, and the RTP media stream sender MAY then
   respond with a REFUSE containing the same PauseID as in the RESUME.
   When receiving such REFUSE with a PauseID identical to the one in the
   sent RESUME, RTP media stream receivers SHOULD then avoid sending
   further RESUME requests for some reasonable amount of time, to allow
   the condition to clear.

   If the RTP media stream sender receives several identical RESUME for
   an RTP media stream that was already at least once responded with
   REFUSE and the condition causing REFUSE remains, those additional
   REFUSE should be sent with regular RTCP timing.  A single REFUSE can
   respond to several identical RESUME requests.

   When resuming a paused media stream, especially for media that makes
   use of temporal redundancy between samples such as video, the
   temporal dependency between samples taken before the pause and at the
   time instant the stream is resumed may not be appropriate to use in
   the encoding.  Should such temporal dependency between before and
   after the media was paused be used by the media sender, it requires
   the media receiver to have saved the sample from before the pause for
   successful continued decoding when resuming.  The use of this
   temporal dependency is left up to the media sender.  If temporal
   dependency is not used when media is resumed, the first encoded
   sample after the pause will not contain any temporal dependency to
   samples before the pause (for video it may be a so-called intra
   picture).  If temporal dependency to before the pause is used by the



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   media sender when resuming, and if the media receiver did not save
   any sample from before the pause, the media receiver can use a FIR
   request [RFC5104] to explicitly ask for a sample without temporal
   dependency (for video a so-called intra picture), even at the same
   time as sending the RESUME.

6.5.  TMMBR/TMMBN Considerations

   As stated, TMMBR/TMMBN may be used to provide pause and resume
   functionality for the point-to-point case.  If the topology is not
   point-to-point, TMMBR/TMMBN cannot safely be used for pause or
   resume.

   This is a brief summary of what functionality is provided when using
   TMMBR/TMMBN:

   TMMBR 0:  Corresponds to PAUSE, without the requirement for any hold-
      off period to wait for RESUME before pausing the media stream.

   TMMBR >0:  Corresponds to RESUME when the media stream was previously
      paused with TMMBR 0.  Since there is only a single media receiver,
      there is no need for the media sender to delay resuming the media
      stream until after sending TMMBN >0, or to apply the hold-off
      period specified in [RFC5104] before increasing the bitrate from
      zero.

   TMMBN 0:  Corresponds to PAUSED.  Also corresponds to a REFUSE
      indication when a media stream is requested to be resumed with
      TMMBR >0.

   TMMBN >0:  Corresponds to a REFUSE indication when a media stream is
      requested to be paused with TMMBR 0.

7.  Participant States

   This document introduces three new states for a media stream in an
   RTP sender, according to the figure and sub-sections below.  Any
   references to PAUSE, PAUSED, RESUME and REFUSE in this section SHALL
   be taken to apply to the extent possible also when TMMBR/TMMBN are
   used (Section 6.5) for this functionality.











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        +------------------------------------------------------+
        |                     Received RESUME                  |
        v                                                      |
   +---------+ Received PAUSE  +---------+ Hold-off period +--------+
   | Playing |---------------->| Pausing |---------------->| Paused |
   |         |<----------------|         |                 |        |
   +---------+ Received RESUME +---------+                 +--------+
     ^     |                        | PAUSE decision           |
     |     |                        v                          |
     |     |  PAUSE decision   +---------+    PAUSE decision   |
     |     +------------------>| Local   |<--------------------+
     +-------------------------| Paused  |
             RESUME decision   +---------+


                        Figure 8: RTP Pause States

7.1.  Playing State

   This state is not new, but is the normal media sending state from
   [RFC3550].  When entering the state, the PauseID MUST be incremented
   by one in modulo arithmetic.  The RTP sequence number for the first
   packet sent after a pause SHALL be incremented by one compared to the
   highest RTP sequence number sent before the pause.  The first RTP
   Time Stamp for the first packet sent after a pause SHOULD be set
   according to capture times at the source.

7.2.  Pausing State

   In this state, the media sender has received at least one PAUSE
   message for the stream in question.  The media sender SHALL wait
   during a hold-off period for the possible reception of RESUME
   messages for the RTP media stream being paused before actually
   pausing media transmission.  The period to wait SHALL be long enough
   to allow another media receiver to respond to the PAUSE with a
   RESUME, if it determines that it would not like to see the stream
   paused.  This delay period (denoted by 'Hold-off period' in the
   figure) is determined by the formula:

      2 * RTT + T_dither_max,

   where RTT is the longest round trip known to the media sender and
   T_dither_max is defined in section 3.4 of [RFC4585].  The hold-off
   period MAY be set to 0 by some signaling (Section 10) means when it
   can be determined that there is only a single receiver, for example
   in point-to-point or some unicast situations.





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   If the RTP media stream sender has set the hold-off period to 0 and
   receives information that it was an incorrect decision and that there
   are in fact several receivers of the stream, for example by RTCP RR,
   it MUST change the hold-off to instead be based on the above formula.

7.3.  Paused State

   An RTP media stream is in paused state when the sender pauses its
   transmission after receiving at least one PAUSE message and the hold-
   off period has passed without receiving any RESUME message for that
   stream.

   When entering the state, the media sender SHALL send a PAUSED
   indication to all known media receivers, and SHALL also repeat PAUSED
   in the next two regular RTCP reports.

   Following sub-sections discusses some potential issues when an RTP
   sender goes into paused state.  These conditions are also valid if an
   RTP Translator is used in the communication.  When an RTP Mixer
   implementing this specification is involved between the participants
   (which forwards the stream by marking the RTP data with its own
   SSRC), it SHALL be a responsibility of the Mixer to control sending
   PAUSE and RESUME requests to the sender.  The below conditions also
   apply to the sender and receiver parts of the RTP Mixer,
   respectively.

7.3.1.  RTCP BYE Message

   When a participant leaves the RTP session, it sends an RTCP BYE
   message.  In addition to the semantics described in section 6.3.4 and
   6.3.7 of RTP [RFC3550], following two conditions MUST also be
   considered when an RTP participant sends an RTCP BYE message,

   o  If a paused sender sends an RTCP BYE message, receivers observing
      this SHALL NOT send further PAUSE or RESUME requests to it.

   o  Since a sender pauses its transmission on receiving the PAUSE
      requests from any receiver in a session, the sender MUST keep
      record of which receiver that caused the RTP media stream to
      pause.  If that receiver sends an RTCP BYE message observed by the
      sender, the sender SHALL resume the RTP media stream.

7.3.2.  SSRC Time-out

   Section 6.3.5 in RTP [RFC3550] describes the SSRC time-out of an RTP
   participant.  Every RTP participant maintains a sender and receiver
   list in a session.  If a participant does not get any RTP or RTCP
   packets from some other participant for the last five RTCP reporting



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   intervals it removes that participant from the receiver list.  Any
   streams that were paused by that removed participant SHALL be
   resumed.

7.4.  Local Paused State

   This state can be entered at any time, based on local decision from
   the media sender.  As for Paused State (Section 7.3), the media
   sender SHALL send a PAUSED indication to all known media receivers,
   when entering the state, and repeat it in the next two regular RTCP
   reports.

   When leaving the state, the stream state SHALL become Playing,
   regardless whether or not there were any media receivers that sent
   PAUSE for that stream, effectively clearing the media sender's memory
   for that media stream.

8.  Message Format

   Section 6 of AVPF [RFC4585] defines three types of low-delay RTCP
   feedback messages, i.e. Transport layer, Payload-specific, and
   Application layer feedback messages.  This document defines a new
   Transport layer feedback message, this message is either a PAUSE
   request, a RESUME request, or one of four different types of
   acknowledgements in response to either PAUSE or RESUME requests.

   The Transport layer feedback messages are identified by having the
   RTCP payload type be RTPFB (205) as defined by AVPF [RFC4585].  The
   PAUSE and RESUME messages are identified by Feedback Message Type
   (FMT) value in common packet header for feedback message defined in
   section 6.1 of AVPF [RFC4585].  The PAUSE and RESUME transport
   feedback message is identified by the FMT value = TBA1.

   The Common Packet Format for Feedback Messages is defined by AVPF
   [RFC4585] is:

    0                   1                   2                   3
    0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |V=2|P|   FMT   |       PT      |          Length               |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                  SSRC of packet sender                        |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                  SSRC of media source                         |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   :            Feedback Control Information (FCI)                 :
   :                                                               :




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   For the PAUSE and RESUME messages, the following interpretation of
   the packet fields will be:

   FMT:  The FMT value identifying the PAUSE and RESUME message: TBA1

   PT:  Payload Type = 205 (RTPFB)

   Length:  As defined by AVPF, i.e. the length of this packet in 32-bit
      words minus one, including the header and any padding.

   SSRC of packet sender:  The SSRC of the RTP session participant
      sending the messages in the FCI.  Note, for end-points that have
      multiple SSRCs in an RTP session, any of its SSRCs MAY be used to
      send any of the pause message types.

   SSRC of media source:  Not used, SHALL be set to 0.  The FCI
      identifies the SSRC the message is targeted for.

   The Feedback Control Information (FCI) field consist of one or more
   PAUSE, RESUME, PAUSED, REFUSE, or any future extension.  These
   messages have the following FCI format:

   0                   1                   2                   3
    0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                           Target SSRC                         |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   | Type  |  Res  | Parameter Len |           PauseID             |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   :                         Type Specific                         :
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+


       Figure 9: Syntax of FCI Entry in the PAUSE and RESUME message

   The FCI fields have the following definitions:

   Target SSRC (32 bits):  For a PAUSE and RESUME messages, this value
      is the SSRC that the request is intended for.  For PAUSED, it MUST
      be the SSRC being paused.  If pausing is the result of a PAUSE
      request, the value in PAUSED is effectively the same as Target
      SSRC in a related PAUSE request.  For REFUSE, it MUST be the
      Target SSRC of the PAUSE or RESUME request that cannot change
      state.  A CSRC MUST NOT be used as a target as the interpretation
      of such a request is unclear.

   Type (4 bits):  The pause feedback type.  The values defined in this
      specification are as follows,



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      0: PAUSE request message

      1: RESUME request message

      2: PAUSED indication message

      3: REFUSE indication message

      4-15:  Reserved for future use

   Res: (4 bits):  Type specific reserved.  SHALL be ignored by
      receivers implementing this specification and MUST be set to 0 by
      senders implementing this specification.

   Parameter Len: (8 bits):  Length of the Type Specific field in 32-bit
      words.  MAY be 0.

   PauseID (16 bits):  Message sequence identification.  SHALL be
      incremented by one modulo 2^16 for each new PAUSE message, unless
      the message is re-transmitted.  The initial value SHOULD be 0.
      The PauseID is scoped by the Target SSRC, meaning that PAUSE,
      RESUME, and PAUSED messages therefore share the same PauseID space
      for a specific Target SSRC.

   Type Specific: (variable):  Defined per pause feedback Type.  MAY be
      empty.

9.  Message Details

   This section contains detailed explanations of each message defined
   in this specification.  All transmissions of request and indications
   are governed by the transmission rules as defined by Section 9.5.

   Any references to PAUSE, PAUSED, RESUME and REFUSE in this section
   SHALL be taken to apply to the extent possible also when TMMBR/TMMBN
   are used (Section 6.5) for this functionality.  TMMBR/TMMBN MAY be
   used instead of the messages defined in this specification when the
   effective topology is point-to-point.  If either sender or receiver
   learns that the topology is not point-to-point, TMMBR/TMMBN MUST NOT
   be used for pause/resume functionality.  If the messages defined in
   this specification are supported in addition to TMMBR/TMMBN, pause/
   resume signaling MUST revert to use those instead.  If the topology
   is not point-to-point and the messages defined in this specification
   are not supported, pause/resume functionality with TMMBR/TMMBN MUST
   NOT be used.






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9.1.  PAUSE

   An RTP media stream receiver MAY schedule PAUSE for transmission at
   any time.

   PAUSE has no defined Type Specific parameters and Parameter Len MUST
   be set to 0.

   PauseID SHOULD be the available PauseID, as indicated by PAUSED
   (Section 9.2) or implicitly determined by previously received PAUSE
   or RESUME (Section 9.3) requests.  A randomly chosen PauseID MAY be
   used if it was not possible to retrieve PauseID information, in which
   case the PAUSE will either succeed, or the correct PauseID can be
   learnt from the returned REFUSE (Section 9.4).  A PauseID that is
   matching the available PauseID is henceforth also called a valid
   PauseID.

   PauseID needs to be incremented by one, in modulo arithmetic, for
   each PAUSE request that is not a retransmission, compared to what was
   used in the last PAUSED indication sent by the media sender.  This is
   to ensure that the PauseID matches what is the current available
   PauseID at the media sender.  The media sender increments what it
   considers to be the available PauseID when entering Playing State
   (Section 7.1).

   For the scope of this specification, a PauseID larger than the
   current one is defined as having a value between and including
   (PauseID + 1) MOD 2^16 and (PauseID + 2^14) MOD 2^16, where "MOD" is
   the modulo operator.  Similarly, a PauseID smaller than the current
   one is defined as having a value between and including (PauseID -
   2^15) MOD 2^16 and (PauseID - 1) MOD 2^16.

   If an RTP media stream receiver that sent a PAUSE with a certain
   PauseID receives a RESUME with the same PauseID, it is RECOMMENDED
   that it refrains from sending further PAUSE requests for some
   appropriate time since the RESUME indicates that there are other
   receivers that still wishes to receive the stream.

   If the targeted RTP media stream does not pause, if no PAUSED
   indication with a larger PauseID than the one used in PAUSE, and if
   no REFUSE is received within 2 * RTT + T_dither_max, the PAUSE MAY be
   scheduled for retransmission, using the same PauseID.  RTT is the
   observed round-trip to the RTP media stream sender and T_dither_max
   is defined in section 3.4 of [RFC4585].

   When an RTP media stream sender in Playing State (Section 7.1)
   receives a valid PAUSE, and unless local considerations currently
   makes it impossible to pause the stream, it SHALL enter Pausing State



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   (Section 7.2) when reaching an appropriate place to pause in the
   media stream, and act accordingly.

   If an RTP media stream sender receives a valid PAUSE while in
   Pausing, Paused (Section 7.3) or Local Paused (Section 7.4) States,
   the received PAUSE SHALL be ignored.

9.2.  PAUSED

   The PAUSED indication MAY be sent either as a result of a valid PAUSE
   (Section 9.1) request, when entering Paused State (Section 7.3), or
   based on a RTP media stream sender local decision, when entering
   Local Paused State (Section 7.4).

   PauseID MUST contain the available, valid value to be included in a
   subsequent RESUME (Section 9.3).

   PAUSED SHALL contain a 32 bit parameter with the RTP extended highest
   sequence number valid when the RTP media stream was paused.
   Parameter Len MUST be set to 1.

   After having entered Paused or Local Paused State and thus having
   sent PAUSED once, PAUSED MUST also be included in the next two
   regular RTCP reports, given that the pause condition is then still
   effective.

   While remaining in Paused or Local Paused States, PAUSED MAY be
   included in all regular RTCP reports.

   When in Paused or Local Paused States, It is RECOMMENDED to send
   PAUSED at the earliest opportunity and also to include it in the next
   two regular RTCP reports, whenever the RTP media sender learns that
   there are end-points that did not previously receive the stream, for
   example by RTCP reports with an SSRC and a CNAME that was not
   previously seen in the RTP session.

9.3.  RESUME

   An RTP media stream receiver MAY schedule RESUME for transmission
   whenever it wishes to resume a paused stream, or to disapprove a
   stream from being paused.

   PauseID SHOULD be the valid PauseID, as indicated by PAUSED
   (Section 9.2) or implicitly determined by previously received PAUSE
   (Section 9.1) or RESUME requests.  A randomly chosen PauseID MAY be
   used if it was not possible to retrieve PauseID information, in which
   case the RESUME will either succeed, or the correct PauseID can be
   learnt from a returned REFUSE (Section 9.4).



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   RESUME has no defined Type Specific parameters and Parameter Len MUST
   be set to 0.

   When an RTP media stream sender in Pausing (Section 7.2), Paused
   (Section 7.3) or Local Paused State (Section 7.4) receives a valid
   RESUME, and unless local considerations currently makes it impossible
   to resume the stream, it SHALL enter Playing State (Section 7.1) and
   act accordingly.  If the RTP media stream sender is incapable of
   honoring the RESUME request with a valid PauseID, or receives a
   RESUME request with an invalid PauseID while in Paused or Pausing
   state, the RTP media stream sender sends a REFUSE message as
   specified below.

   If an RTP media stream sender in Playing State receives a RESUME
   containing either a valid PauseID or a PauseID that is less than the
   valid PauseID, the received RESUME SHALL be ignored.

9.4.  REFUSE

   REFUSE has no defined Type Specific parameters and Parameter Len MUST
   be set to 0.

   If an RTP media sender receives a valid PAUSE (Section 9.1) or RESUME
   (Section 9.3) request that cannot be fulfilled by the sender due to
   some local consideration, it SHALL schedule transmission of a REFUSE
   indication containing the valid PauseID from the rejected request.

   If an RTP media stream sender receives PAUSE or RESUME requests with
   a non-valid PauseID it SHALL schedule a REFUSE response containing
   the available, valid PauseID, except if the RTP media stream sender
   is in Playing State and receives a RESUME with a PauseID less than
   the valid one, in which case the RESUME SHALL be ignored.

   If several PAUSE or RESUME that would render identical REFUSE
   responses are received before the scheduled REFUSE is sent, duplicate
   REFUSE MUST NOT be scheduled for transmission.  This effectively lets
   a single REFUSE respond to several invalid PAUSE or RESUME requests.

   If REFUSE containing a certain PauseID was already sent and yet more
   PAUSE or RESUME messages are received that require additional REFUSE
   with that specific PauseID to be scheduled, and unless the PauseID
   number space has wrapped since REFUSE was last sent with that
   PauseID, further REFUSE messages with that PauseID SHOULD be sent in
   regular RTCP reports.

   An RTP media stream receiver that sent a PAUSE or RESUME request and
   receives a REFUSE containing the same PauseID as in the request




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   SHOULD refrain from sending an identical request for some appropriate
   time to allow the condition that caused REFUSE to clear.

   An RTP media stream receiver that sent a PAUSE or RESUME request and
   receives a REFUSE containing a PauseID different from the request MAY
   schedule another request using the PauseID from the REFUSE
   indication.

9.5.  Transmission Rules

   The transmission of any RTCP feedback messages defined in this
   specification MUST follow the normal AVPF defined timing rules and
   depends on the session's mode of operation.

   All messages defined in this specification MAY use either Regular,
   Early or Immediate timings, taking the following into consideration:

   o  PAUSE SHOULD use Early or Immediate timing, except for
      retransmissions that SHOULD use Regular timing.

   o  The first transmission of PAUSED for each (non-wrapped) PauseID
      SHOULD be sent with Immediate or Early timing, while subsequent
      PAUSED for that PauseID SHOULD use Regular timing.

   o  RESUME SHOULD always use Immediate or Early timing.

   o  The first transmission of REFUSE for each (non-wrapped) PauseID
      SHOULD be sent with Immediate or Early timing, while subsequent
      REFUSE for that PauseID SHOULD use Regular timing.

10.  Signalling

   The capability of handling messages defined in this specification MAY
   be exchanged at a higher layer such as SDP.  This document extends
   the rtcp-fb attribute defined in section 4 of AVPF [RFC4585] to
   include the request for pause and resume.  Like AVPF [RFC4585] and
   CCM [RFC5104], it is RECOMMENDED to use the rtcp-fb attribute at
   media level and it MUST NOT be used at session level.  This
   specification follows all the rules defined in AVPF for rtcp-fb
   attribute relating to payload type in a session description.

      Note: When TMMBR 0 / TMMBN 0 are used to implement pause and
      resume functionality (with the restrictions described in this
      memo), signaling rtcp-fb attribute with ccm tmmbr parameter is
      sufficient and no further signaling is necessary.

   This specification defines two new parameters to the "ccm" feedback
   value defined in CCM [RFC5104], "pause" and "paused".



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   o  "pause" represents the capability to understand the RTCP feedback
      message and all of the defined FCIs of PAUSE, RESUME, PAUSED and
      REFUSE.  A direction sub-parameter is used to determine if a given
      node desires to issue PAUSE or RESUME requests, can respond to
      PAUSE or RESUME requests, or both.

   o  "paused" represents the functionality of supporting the playing
      and local paused states and generate PAUSED FCI when a media
      stream delivery is paused.  A direction sub-parameter is used to
      determine if a given node desires to receive these indications,
      intends to send them, or both.

   The reason for this separation is to make it possible for partial
   implementation of this specification, according to the different
   roles in the use cases section (Section 3).

   A sub-parameter named "nowait", indicating that the hold-off time
   defined in Section 7.2 can be set to 0, reducing the latency before
   the media stream can paused after receiving a PAUSE request.  This
   condition occurs when there will be only a single receiver per
   direction in the RTP session, for example in point-to-point sessions.
   It is also possible to use in scenarios using unidirectional media.
   The conditions that allow "nowait" to be set also indicate that it
   would be possible to use CCM TMMBR/TMMBN as pause/resume signaling.

   A sub-parameter named "dir" is used to indicate in which directions a
   given node will use the pause or paused functionality.  The node
   being configured or issuing an offer or an answer uses the
   directionality in the following way.  Note that pause and paused have
   separate and different definitions.

   Direction ("dir") values for "pause" is defined as follows:

   sendonly:  The node intends to send PAUSE and RESUME requests for
      other nodes' media streams and is thus also capable of receiving
      PAUSED and REFUSE.  It will not support receiving PAUSE and RESUME
      requests.

   recvonly:  The node supports receiving PAUSE and RESUME requests
      targeted for media streams sent by the node.  It will send PAUSED
      and REFUSE as needed.  The node will not send any PAUSE and RESUME
      requests.

   sendrecv:  The node supports receiving PAUSE and RESUME requests
      targeted for media streams sent by the node.  The node intends to
      send PAUSE and RESUME requests for other nodes' media streams.
      Thus the node is capable of sending and receiving all types of
      pause messages.  This is the default value.  If the "dir"



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      parameter is omitted, it MUST be interpreted to represent this
      value.

   Direction values for "paused" is defined as follows:

   sendonly:  The node intends to send PAUSED indications whenever it
      pauses media delivery in any of its media streams.  It has no need
      to receive PAUSED indications itself.

   recvonly:  The node desires to receive PAUSED indications whenever
      any media stream sent by another node is paused.  It does not
      intend to send any PAUSED indications.

   sendrecv:  The nodes desires to receive PAUSED indications and
      intends to send PAUSED indications whenever any media stream is
      paused.  This is the default value.  If the "dir" parameter is
      omitted, it MUST be interpreted to represent this value.

   This is the resulting ABNF [RFC5234], extending existing ABNF in
   section 7.1 of CCM [RFC5104]:

   rtcp-fb-ccm-param =/ SP "pause" *(SP pause-attr)
                      / SP "paused" *(SP paused-attr)
   pause-attr        = direction
                     / "nowait"
                     / token ; for future extensions
   paused-attr       = direction
                     / token ; for future extensions
   direction         = "dir=" direction-alts
   direction-alts    = "sendonly" / "recvonly" / "sendrecv"


                              Figure 10: ABNF

   An endpoint implementing this specification and using SDP to signal
   capability SHOULD indicate both of the new "pause" and "paused"
   parameters with ccm signaling.  When negotiating usage, it is
   possible select either of them, noting that "pause" contain the full
   "paused" functionality.  A sender or receiver SHOULD NOT use the
   messages from this specification towards receivers that did not
   declare capability for it.

   There MUST NOT be more than one "a=rtcp-fb" line with "pause" and one
   with "paused" applicable to a single payload type in the SDP, unless
   the additional line uses "*" as payload type, in which case "*" SHALL
   be interpreted as applicable to all listed payload types that does
   not have an explicit "pause" or "paused" specification.




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   There MUST NOT be more than a single direction sub-parameter per
   "pause" and "paused" parameter.  There MUST NOT be more than a single
   "nowait" sub-parameter per "pause" parameter.

10.1.  Offer-Answer Use

   An offerer implementing this specification needs to include "pause"
   and/or "paused" CCM parameters with suitable directionality parameter
   ("dir") in the SDP, according to what messages it intends to send and
   desires or is capable to receive in the session.  It is RECOMMENDED
   to include both "pause" and "paused" if "pause" is supported, to
   enable at least the "paused" functionality if the answer only
   supports "paused" or different directionality for the two
   functionalities.  The "pause" and "paused" functionalities are
   negotiated independently, although the "paused" functionality is part
   of the "pause" functionality.  As a result, an answerer MAY remove
   "pause" or "paused" lines from the SDP depending on the agreed mode
   of functionality.

   In offer/answer, the "dir" parameter is interpreted based on the
   agent providing the SDP.  The node described in the offer is the
   offerer, and the answerer is described in an answer.  In other words,
   an offer for "paused dir=sendonly" means that the offerer intends to
   send PAUSED indications whenever it pauses media delivery in any of
   its media streams.

   An answerer receiving an offer with a "pause" parameter with
   dir=sendrecv MAY remove the pause line in its answer, respond with
   pause keeping sendrecv for full bi-directionality, or it may change
   dir value to either sendonly or recvonly based on its capabilities
   and desired functionality.  An offer with a "pause" parameter with
   dir=sendonly or dir=recvonly is either completely removed or accepted
   with reverse directionality, i.e. sendonly becomes recvonly or
   recvonly becomes sendonly.

   An answer receiving an offer with "paused" has the same choices as
   for "pause" above.  It should be noted that the directionality of
   pause is the inverse of media direction, while the directionality of
   paused is the same as the media direction.

   If the offerer believes that itself and the intended answerer are
   likely the only end-points in the RTP session, it MAY include the
   "nowait" sub-parameter on the "pause" line in the offer.  If an
   answerer receives the "nowait" sub-parameter on the "pause" line in
   the SDP, and if it has information that the offerer and itself are
   not the only end-points in the RTP session, it MUST NOT include any
   "nowait" sub-parameter on its "pause" line in the SDP answer.  The
   answerer MUST NOT add "nowait" on the "pause" line in the answer



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   unless it is present on the "paused" line in the offer.  If both
   offer and answer contained a "nowait" parameter, then the hold-off
   time is configured to 0 at both offerer and answerer.

10.2.  Declarative Use

   In declarative use, the SDP is used to configure the node receiving
   the SDP.  This has implications on the interpretation of the SDP
   signalling extensions defined in this draft.  First, it is normally
   only necessary to include either "pause" or "paused" parameter to
   indicate the level of functionality the node should use in this RTP
   session.  Including both is only necessary if some implementations
   only understands "paused" and some other can understand both.  Thus
   indicating both means use pause if you understand it, and if you only
   understand paused, use that.

   The "dir" directionality parameter indicates how the configured node
   should behave.  For example "pause" with sendonly:

   sendonly:  The node intends to send PAUSE and RESUME requests for
      other nodes' media streams and is thus also capable of receiving
      PAUSED and REFUSE.  It will not support receiving PAUSE and RESUME
      requests.

   In this example, the configured node should send PAUSE and RESUME
   requests if has reason for it.  It does not need to respond to any
   PAUSE or RESUME requests as that is not supported.

   The "nowait" parameter, if included, is followed as specified.  It is
   the responsibility of the declarative SDP sender to determine if a
   configured node will participate in a session that will be point to
   point, based on the usage.  For example, a conference client being
   configured for an any source multicast session using SAP [RFC2974]
   will not be in a point to point session, thus "nowait" cannot be
   included.  An RTSP [RFC2326] client receiving a declarative SDP may
   very well be in a point to point session, although it is highly
   doubtful that an RTSP client would need to support this
   specification, considering the inherent PAUSE support in RTSP.

11.  Examples

   The following examples shows use of PAUSE and RESUME messages,
   including use of offer-answer:

   1.  Offer-Answer

   2.  Point-to-Point session




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   3.  Point-to-multipoint using Mixer

   4.  Point-to-multipoint using Translator

11.1.  Offer-Answer

   The below figures contains an example how to show support for pausing
   and resuming the streams, as well as indicating whether or not the
   hold-off period can be set to 0.

   v=0
   o=alice 3203093520 3203093520 IN IP4 alice.example.com
   s=Pausing Media
   t=0 0
   c=IN IP4 alice.example.com
   m=audio 49170 RTP/AVPF 98 99
   a=rtpmap:98 G719/48000
   a=rtpmap:99 PCMA/8000
   a=rtcp-fb:* ccm pause nowait
   a=rtcp-fb:* ccm paused


           Figure 11: SDP Offer With Pause and Resume Capability

   The offerer supports all of the messages defined in this
   specification and offers a sendrecv stream.  The offerer also
   believes that it will be the sole receiver of the answerer's stream
   as well as that the answerer will be the sole receiver of the
   offerer's stream and thus includes the "nowait" sub-parameter for
   both "pause" and "paused" parameters.

   This is the SDP answer:

   v=0
   o=bob 293847192 293847192 IN IP4 bob.example.com
   s=-
   t=0 0
   c=IN IP4 bob.example.com
   m=audio 49202 RTP/AVPF 98
   a=rtpmap:98 G719/48000
   a=rtcp-fb:98 ccm pause dir=sendonly
   a=rtcp-fb:98 ccm paused


          Figure 12: SDP Answer With Pause and Resume Capability

   The answerer will not allow its sent streams to be paused or resumed
   and thus support pause only in sendonly mode.  It does support paused



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   and intends to send it, and also desires to receive PAUSED
   indications.  Thus paused in sendrecv mode is included in the answer.
   The answerer somehow knows that it will not be a point-to-point RTP
   session and has therefore removed "nowait" from the "pause" line,
   meaning that the offerer must use a non-zero hold-off time when being
   requested to pause the stream.

   When using TMMBR 0 / TMMBN 0 to achieve pause and resume
   functionality, there are no differences in SDP compared to CCM
   [RFC5104] and therefore no such examples are included here.

11.2.  Point-to-Point Session

   This is the most basic scenario, which involves two participants,
   each acting as a sender and/or receiver.  Any RTP data receiver sends
   PAUSE or RESUME messages to the sender, which pauses or resumes
   transmission accordingly.  The hold-off time before pausing a stream
   is 0.

   +---------------+                   +---------------+
   |  RTP Sender   |                   | RTP Receiver  |
   +---------------+                   +---------------+
          :           t1: RTP data           :
          | -------------------------------> |
          |           t2: PAUSE(3)           |
          | <------------------------------- |
          |       < RTP data paused >        |
          |           t3: PAUSED(3)          |
          | -------------------------------> |
          :       < Some time passes >       :
          |           t4: RESUME(3)          |
          | <------------------------------- |
          |           t5: RTP data           |
          | -------------------------------> |
          :       < Some time passes >       :
          |           t6: PAUSE(4)           |
          | <------------------------------- |
          |       < RTP data paused >        |
          :                                  :


          Figure 13: Pause and Resume Operation in Point-to-Point

   Figure 13 shows the basic pause and resume operation in Point-to-
   Point scenario.  At time t1, an RTP sender sends data to a receiver.
   At time t2, the RTP receiver requests the sender to pause the stream,
   using PauseID 3 (which it knew since before in this example).  The
   sender pauses the data and replies with a PAUSED containing the same



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   PauseID.  Some time later (at time t4) the receiver requests the
   sender to resume, which resumes its transmission.  The next PAUSE,
   sent at time t6, contains an updated PauseID (4).

   +---------------+                   +---------------+
   |  RTP Sender   |                   | RTP Receiver  |
   +---------------+                   +---------------+
          :           t1: RTP data           :
          | -------------------------------> |
          |           t2: TMMBR 0            |
          | <------------------------------- |
          |       < RTP data paused >        |
          |           t3: TMMBN 0            |
          | -------------------------------> |
          :       < Some time passes >       :
          |           t4: TMMBR 150000       |
          | <------------------------------- |
          |           t5: RTP data           |
          | -------------------------------> |
          :       < Some time passes >       :
          |           t6: TMMBR 0            |
          | <------------------------------- |
          |       < RTP data paused >        |
          :                                  :


            Figure 14: TMMBR Pause and Resume in Point-to-Point

   Figure 14 describes the same point-to-point scenario as above, but
   using TMMBR/TMMBN signaling.





















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   +---------------+                       +---------------+
   |  RTP Sender   |                       | RTP Receiver  |
   +---------------+                       +---------------+
          :           t1: RTP data                :
          | ------------------------------------> |
          |                   t2: PAUSE(7), lost  |
          |                   <---X-------------- |
          |                                       |
          |           t3: RTP data                |
          | ------------------------------------> |
          :                                       :
          |    <Timeout, still receiving data>    |
          |           t4: PAUSE(7)                |
          | <------------------------------------ |
          |          < RTP data paused >          |
          |           t5: PAUSED(7)               |
          | ------------------------------------> |
          :          < Some time passes >         :
          |                   t6: RESUME(7), lost |
          |                   <---X-------------- |
          |           t7: RESUME(7)               |
          | <------------------------------------ |
          |           t8: RTP data                |
          | ------------------------------------> |
          |           t9: RESUME(7)               |
          | <------------------------------------ |
          :                                       :


         Figure 15: Pause and Resume Operation With Messages Lost

   Figure 15 describes what happens if a PAUSE message from an RTP media
   stream receiver does not reach the RTP media stream sender.  After
   sending a PAUSE message, the RTP media stream receiver waits for a
   time-out to detect if the RTP media stream sender has paused the data
   transmission or has sent PAUSED indication according to the rules
   discussed in Section 7.3.  As the PAUSE message is lost on the way
   (at time t2), RTP data continues to reach to the RTP media stream
   receiver.  When the timer expires, the RTP media stream receiver
   schedules a retransmission of the PAUSE message, which is sent at
   time t4.  If the PAUSE message now reaches the RTP media stream
   sender, it pauses the RTP media stream and replies with PAUSED.

   At time t6, the RTP media stream receiver wishes to resume the stream
   again and sends a RESUME, which is lost.  This does not cause any
   severe effect, since there is no requirement to wait until further
   RESUME are sent and another RESUME are sent already at time t7, which
   now reaches the RTP media stream sender that consequently resumes the



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   stream at time t8.  The time interval between t6 and t7 can vary, but
   may for example be one RTCP feedback transmission interval as
   determined by the AVPF rules.

   The RTP media stream receiver did not realize that the RTP stream was
   resumed in time to stop yet another scheduled RESUME from being sent
   at time t9.  This is however harmless since the RESUME PauseID is
   less than the valid one and will be ignored by the RTP media stream
   sender.  It will also not cause any unwanted resume even if the
   stream was paused based on a PAUSE from some other receiver before
   receiving the RESUME, since the valid PauseID is now larger than the
   one in the stray RESUME and will only cause a REFUSE containing the
   new valid PauseID from the RTP media stream sender.

   +---------------+                 +---------------+
   |  RTP Sender   |                 | RTP Receiver  |
   +---------------+                 +---------------+
          :           t1: RTP data          :
          | ------------------------------> |
          |           t2: PAUSE(11)         |
          | <------------------------------ |
          |                                 |
          |  < Can not pause RTP data >     |
          |           t3: REFUSE(11)        |
          | ------------------------------> |
          |                                 |
          |           t4: RTP data          |
          | ------------------------------> |
          :                                 :


           Figure 16: Pause Request is Refused in Point-to-Point

   In Figure 16, the receiver requests to pause the sender, which
   refuses to pause due to some consideration local to the sender and
   responds with a REFUSE message.

11.3.  Point-to-multipoint using Mixer

   An RTP Mixer is an intermediate node connecting different transport-
   level clouds.  The Mixer receives streams from different RTP sources,
   selects or combines them based on the application's needs and
   forwards the generated stream(s) to the destination.  The Mixer
   typically puts its' own SSRC(s) in RTP data packets instead of the
   original source(s).

   The Mixer keeps track of all the media streams delivered to the Mixer
   and how they are currently used.  In this example, it selects the



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   video stream to deliver to the receiver R based on the voice activity
   of the media senders.  The video stream will be delivered to R using
   M's SSRC and with an CSRC indicating the original source.

   Note that PauseID is not of any significance for the example and is
   therefore omitted in the description.

     +-----+            +-----+            +-----+            +-----+
     |  R  |            |  M  |            | S1  |            | S2  |
     +-----+            +-----|            +-----+            +-----+
        :                  :   t1:RTP(S1)     :                  :
        | t2:RTP(M:S1)     |<-----------------|                  |
        |<-----------------|                  |                  |
        |                  | t3:RTP(S2)       |                  |
        |                  |<------------------------------------|
        |                  |  t4: PAUSE(S2)   |                  |
        |                  |------------------------------------>|
        |                  |                  |   t5: PAUSED(S2) |
        |                  |<------------------------------------|
        |                  |                  | <S2:No RTP to M> |
        |                  | t6: RESUME(S2)   |                  |
        |                  |------------------------------------>|
        |                  |                  | t7: RTP to M     |
        |                  |<------------------------------------|
        |   t8:RTP(M:S2)   |                  |                  |
        |<-----------------|                  |                  |
        |                  | t9:PAUSE(S1)     |                  |
        |                  |----------------->|                  |
        |                  | t10:PAUSED(S1)   |                  |
        |                  |<-----------------|                  |
        |                  | <S1:No RTP to M> |                  |
        :                  :                  :                  :


     Figure 17: Pause and Resume Operation for a Voice Activated Mixer

   The session starts at t1 with S1 being the most active speaker and
   thus being selected as the single video stream to be delivered to R
   (t2) using the Mixer SSRC but with S1 as CSRC (indicated after the
   colon in the figure).  Then S2 joins the session at t3 and starts
   delivering media to the Mixer.  As S2 has less voice activity then
   S1, the Mixer decides to pause S2 at t4 by sending S2 a PAUSE
   request.  At t5, S2 acknowledges with a PAUSED and at the same
   instant stops delivering RTP to the Mixer.  At t6, the user at S2
   starts speaking and becomes the most active speaker and the Mixer
   decides to switch the video stream to S2, and therefore quickly sends
   a RESUME request to S2.  At t7, S2 has received the RESUME request
   and acts on it by resuming RTP media delivery to M. When the media



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   from t7 arrives at the Mixer, it switches this media into its SSRC
   (M) at t8 and changes the CSRC to S2.  As S1 now becomes unused, the
   Mixer issues a PAUSE request to S1 at t9, which is acknowledged at
   t10 with a PAUSED and the RTP media stream from S1 stops being
   delivered.

11.4.  Point-to-multipoint using Translator

   A transport Translator in an RTP session forwards the message from
   one peer to all the others.  Unlike Mixer, the Translator does not
   mix the streams or change the SSRC of the messages or RTP media.
   These examples are to show that the messages defined in this
   specification can be safely used also in a transport Translator case.
   The parentheses in the figures contains (Target SSRC, PauseID)
   information for the messages defined in this specification.

   +-------------+     +-------------+     +--------------+
   |  Sender(S)  |     | Translator  |     | Receiver(R)  |
   +-------------+     +-------------|     +--------------+
          : t1: RTP(S)        :                   :
          |------------------>|                   |
          |                   | t2: RTP (S)       |
          |                   |------------------>|
          |                   | t3: PAUSE(S,3)    |
          |                   |<------------------|
          | t4:PAUSE(S,3)     |                   |
          |<------------------|                   |
          : < Sender waiting for possible RESUME> :
          |          < RTP data paused >          |
          | t5: PAUSED(S,3)   |                   |
          |------------------>|                   |
          |                   | t6: PAUSED(S,3)   |
          |                   |------------------>|
          :                   :                   :
          |                   | t7: RESUME(S,3)   |
          |                   |<------------------|
          | t8: RESUME(S,3)   |                   |
          |<------------------|                   |
          | t9: RTP (S)       |                   |
          |------------------>|                   |
          |                   | t10: RTP (S)      |
          |                   |------------------>|
          :                   :                   :


   Figure 18: Pause and Resume Operation Between Two Participants Using
                               a Translator




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   Figure 18 describes how a Translator can help the receiver in pausing
   and resuming the sender.  The sender S sends RTP data to the receiver
   R through Translator, which just forwards the data without modifying
   the SSRCs.  The receiver sends a PAUSE request to the sender, which
   in this example knows that there may be more receivers of the stream
   and waits a non-zero hold-off time to see if there is any other
   receiver that wants to receive the data, does not receive any
   disapproving RESUME, hence pauses itself and replies with PAUSED.
   Similarly the receiver resumes the sender by sending RESUME request
   through Translator.  Since this describes only a single pause
   operation for a single media sender, all messages uses a single
   PauseID, in this example 3.







































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   +-----+            +-----+            +-----+            +-----+
   |  S  |            |  T  |            | R1  |            | R2  |
   +-----+            +-----|            +-----+            +-----+
      : t1:RTP(S)        :                  :                  :
      |----------------->|                  |                  |
      |                  | t2:RTP(S)        |                  |
      |                  |----------------->------------------>|
      |                  | t3:PAUSE(S,7)    |                  |
      |                  |<-----------------|                  |
      | t4:PAUSE(S,7)    |                  |                  |
      |<-----------------|------------------------------------>|
      |                  |                  |   t5:RESUME(S,7) |
      |                  |<------------------------------------|
      | t6:RESUME(S,7)   |                  |                  |
      |<-----------------|                  |                  |
      |                  |<RTP stream continues to R1 and R2>  |
      |                  |                  |   t7: PAUSE(S,8) |
      |                  |<------------------------------------|
      | t8:PAUSE(S,8)    |                  |                  |
      |<-----------------|                  |                  |
      :                  :                  :                  :
      | < Pauses RTP Packet Stream >        |                  |
      | t9:PAUSED(S,8)   |                  |                  |
      |----------------->|                  |                  |
      |                  | t10:PAUSED(S,8)  |                  |
      |                  |----------------->------------------>|
      :                  :                  :                  :
      |                  | t11:RESUME(S,8)  |                  |
      |                  |<-----------------|                  |
      | t12:RESUME(S,8)  |                  |                  |
      |<-----------------|                  |                  |
      | t13:RTP(S)       |                  |                  |
      |----------------->|                  |                  |
      |                  | t14:RTP(S)       |                  |
      |                  |----------------->------------------>|
      :                  :                  :                  :


     Figure 19: Pause and Resume Operation Between One Sender and Two
                       Receivers Through Translator

   Figure 19 explains the pause and resume operations when a transport
   Translator is involved between a sender and two receivers in an RTP
   session.  Each message exchange is represented by the time it
   happens.  At time t1, Sender (S) starts sending media to the
   Translator, which is forwarded to R1 and R2 through the Translator,
   T. R1 and R2 receives RTP data from Translator at t2.  At this point,




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   both R1 and R2 will send RTCP Receiver Reports to S informing that
   they receive S's media stream.

   After some time (at t3), R1 chooses to pause the stream.  On
   receiving the PAUSE request from R1 at t4, S knows that there are at
   least one receiver that may still want to receive the data and uses a
   non-zero hold-off period to wait for possible RESUME messages.  R2
   did also receive the PAUSE request at time t4 and since it still
   wants to receive the stream, it sends a RESUME for it at time t5,
   which is forwarded to the sender S by the translator T. The sender S
   sees the RESUME at time t6 and continues to send data to T which
   forwards to both R1 and R2.  At t7, the receiver R2 chooses to pause
   the stream by sending a PAUSE request with an updated PauseID.  The
   sender S still knows that there are more than one receiver (R1 and
   R2) that may want the stream and again waits a non-zero hold-off
   time, after which and not having received any disapproving RESUME, it
   concludes that the stream must be paused.  S now stops sending the
   stream and replies with PAUSED to R1 and R2.  When any of the
   receivers (R1 or R2) chooses to resume the stream from S, in this
   example R1, it sends a RESUME request to the sender.  The RTP sender
   immediately resumes the stream.

   Consider also an RTP session which includes one or more receivers,
   paused sender(s), and a Translator.  Further assume that a new
   participant joins the session, which is not aware of the paused
   sender(s).  On receiving knowledge about the newly joined
   participant, e.g. any RTP traffic or RTCP report (i.e. either SR or
   RR) from the newly joined participant, the paused sender(s)
   immediately sends PAUSED indications for the paused streams since
   there is now a receiver in the session that did not pause the
   sender(s) and may want to receive the streams.  Having this
   information, the newly joined participant has the same possibility as
   any other participant to resume the paused streams.

12.  IANA Considerations

   As outlined in Section 8, this specification requests IANA to
   allocate

   1.  The FMT number TBA1 to be allocated to the PAUSE and RESUME
       functionality from this specification.

   2.  The 'pause' and 'paused' tags to be used with ccm under rtcp-fb
       AVPF attribute in SDP.

   3.  The 'nowait' parameter to be used with the 'pause' and 'paused'
       tags in SDP.




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   4.  A registry listing registered values for 'pause' Types.

   5.  PAUSE, RESUME, PAUSED, and REFUSE with the listed numbers in the
       pause Type registry.

13.  Security Considerations

   This document extends the CCM [RFC5104] and defines new messages,
   i.e. PAUSE and RESUME.  The exchange of these new messages MAY have
   some security implications, which need to be addressed by the user.
   Following are some important implications,

   1.  Identity spoofing - An attacker can spoof him/herself as an
       authenticated user and can falsely pause or resume any source
       transmission.  In order to prevent this type of attack, a strong
       authentication and integrity protection mechanism is needed.

   2.  Denial of Service (DoS) - An attacker can falsely pause all
       source streams which MAY result in Denial of Service (DoS).  An
       Authentication protocol may prevent this attack.

   3.  Man-in-Middle Attack (MiMT) - The pausing and resuming of an RTP
       source is prone to a Man-in-Middle attack.  Public key
       authentication may be used to prevent MiMT.

14.  Contributors

   Daniel Groendal contributed in the creation and writing of earlier
   versions of this specification.

15.  Acknowledgements

   Daniel Grondal made valuable contributions during the initial
   versions of this draft.

16.  References

16.1.  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.






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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
              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.

   [RFC5234]  Crocker, D. and P. Overell, "Augmented BNF for Syntax
              Specifications: ABNF", STD 68, RFC 5234, January 2008.

16.2.  Informative References

   [I-D.ietf-avtcore-rtp-topologies-update]
              Westerlund, M. and S. Wenger, "RTP Topologies", draft-
              ietf-avtcore-rtp-topologies-update-01 (work in progress),
              October 2013.

   [I-D.ietf-avtext-rtp-grouping-taxonomy]
              Lennox, J., Gross, K., Nandakumar, S., and G. Salgueiro,
              "A Taxonomy of Grouping Semantics and Mechanisms for Real-
              Time Transport Protocol (RTP) Sources", draft-ietf-avtext-
              rtp-grouping-taxonomy-00 (work in progress), November
              2013.

   [I-D.ietf-rtcweb-use-cases-and-requirements]
              Holmberg, C., Hakansson, S., and G. Eriksson, "Web Real-
              Time Communication Use-cases and Requirements", draft-
              ietf-rtcweb-use-cases-and-requirements-14 (work in
              progress), February 2014.

   [I-D.lennox-mmusic-sdp-source-selection]
              Lennox, J. and H. Schulzrinne, "Mechanisms for Media
              Source Selection in the Session Description Protocol
              (SDP)", draft-lennox-mmusic-sdp-source-selection-05 (work
              in progress), October 2012.

   [I-D.westerlund-avtcore-rtp-simulcast]
              Westerlund, M. and S. Nandakumar, "Using Simulcast in RTP
              Sessions", draft-westerlund-avtcore-rtp-simulcast-03 (work
              in progress), October 2013.

   [RFC2326]  Schulzrinne, H., Rao, A., and R. Lanphier, "Real Time
              Streaming Protocol (RTSP)", RFC 2326, April 1998.

   [RFC2974]  Handley, M., Perkins, C., and E. Whelan, "Session
              Announcement Protocol", RFC 2974, October 2000.



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   [RFC3261]  Rosenberg, J., Schulzrinne, H., Camarillo, G., Johnston,
              A., Peterson, J., Sparks, R., Handley, M., and E.
              Schooler, "SIP: Session Initiation Protocol", RFC 3261,
              June 2002.

   [RFC3264]  Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model
              with Session Description Protocol (SDP)", RFC 3264, June
              2002.

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

   [RFC4566]  Handley, M., Jacobson, V., and C. Perkins, "SDP: Session
              Description Protocol", RFC 4566, July 2006.

   [RFC5049]  Bormann, C., Liu, Z., Price, R., and G. Camarillo,
              "Applying Signaling Compression (SigComp) to the Session
              Initiation Protocol (SIP)", RFC 5049, December 2007.

   [RFC5225]  Pelletier, G. and K. Sandlund, "RObust Header Compression
              Version 2 (ROHCv2): Profiles for RTP, UDP, IP, ESP and
              UDP-Lite", RFC 5225, April 2008.

   [RFC5576]  Lennox, J., Ott, J., and T. Schierl, "Source-Specific
              Media Attributes in the Session Description Protocol
              (SDP)", RFC 5576, June 2009.

   [RFC5626]  Jennings, C., Mahy, R., and F. Audet, "Managing Client-
              Initiated Connections in the Session Initiation Protocol
              (SIP)", RFC 5626, October 2009.

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

   [TS25.308]
              3GPP, "High Speed Downlink Packet Access (HSDPA); Overall
              description; Stage 2", 3GPP TS 25.308 10.6.0, December
              2011, <http://www.3gpp.org/ftp/Specs/html-info/25308.htm>.

   [TS26.114]
              3GPP, "IP Multimedia Subsystem (IMS); Multimedia
              telephony; Media handling and interaction", 3GPP TS 26.114
              10.7.0, June 2013,
              <http://www.3gpp.org/ftp/Specs/html-info/26114.htm>.





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   [TS36.201]
              3GPP, "Evolved Universal Terrestrial Radio Access
              (E-UTRA); LTE physical layer; General description", 3GPP
              TS 36.201 10.0.0, December 2010,
              <http://www.3gpp.org/ftp/Specs/html-info/36201.htm>.

Authors' Addresses

   Azam Akram
   Ericsson
   Farogatan 6
   SE - 164 80 Kista
   Sweden

   Phone: +46107142658
   Fax:   +46107175550
   Email: muhammad.azam.akram@ericsson.com
   URI:   www.ericsson.com


   Bo Burman
   Ericsson
   Farogatan 6
   SE - 164 80 Kista
   Sweden

   Phone: +46107141311
   Fax:   +46107175550
   Email: bo.burman@ericsson.com
   URI:   www.ericsson.com


   Roni Even
   Huawei Technologies
   Tel Aviv
   Israel

   Email: roni.even@mail01.huawei.com


   Magnus Westerlund
   Ericsson
   Farogatan 6
   SE- Kista 164 80
   Sweden

   Phone: +46107148287
   Email: magnus.westerlund@ericsson.com



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