<pre>Internet Engineering Task Force (IETF)                      B. Ver Steeg
Request for Comments: 6285                                      A. Begen
Category: Standards Track                                          Cisco
ISSN: 2070-1721                                          T. Van Caenegem
                                                          Alcatel-Lucent
                                                                  Z. Vax
                                                    Magnum Semiconductor
                                                               June 2011


       <span class="h1">Unicast-Based Rapid Acquisition of Multicast RTP Sessions</span>

Abstract

   When an RTP receiver joins a multicast session, it may need to
   acquire and parse certain Reference Information before it can process
   any data sent in the multicast session.  Depending on the join time,
   length of the Reference Information repetition (or appearance)
   interval, size of the Reference Information, and the application and
   transport properties, the time lag before an RTP receiver can
   usefully consume the multicast data, which we refer to as the
   Acquisition Delay, varies and can be large.  This is an undesirable
   phenomenon for receivers that frequently switch among different
   multicast sessions, such as video broadcasts.

   In this document, we describe a method using the existing RTP and RTP
   Control Protocol (RTCP) machinery that reduces the acquisition delay.
   In this method, an auxiliary unicast RTP session carrying the
   Reference Information to the receiver precedes or accompanies the
   multicast stream.  This unicast RTP flow can be transmitted at a
   faster than natural bitrate to further accelerate the acquisition.
   The motivating use case for this capability is multicast applications
   that carry real-time compressed audio and video.  However, this
   method can also be used in other types of multicast applications
   where the acquisition delay is long enough to be a problem.

Status of This Memo

   This is an Internet Standards Track document.

   This document is a product of the Internet Engineering Task Force
   (IETF).  It represents the consensus of the IETF community.  It has
   received public review and has been approved for publication by the
   Internet Engineering Steering Group (IESG).  Further information on
   Internet Standards is available in <a href="./rfc5741#section-2">Section&nbsp;2 of RFC 5741</a>.






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<span class="grey"><a href="./rfc6285">RFC 6285</a>                          RAMS                         June 2011</span>


   Information about the current status of this document, any errata,
   and how to provide feedback on it may be obtained at
   <a href="http://www.rfc-editor.org/info/rfc6285">http://www.rfc-editor.org/info/rfc6285</a>.

Copyright Notice

   Copyright (c) 2011 IETF Trust and the persons identified as the
   document authors.  All rights reserved.

   This document is subject to <a href="https://www.rfc-editor.org/bcp/bcp78">BCP 78</a> and the IETF Trust's Legal
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   carefully, as they describe your rights and restrictions with respect
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   the Trust Legal Provisions and are provided without warranty as
   described in the Simplified BSD License.

   This document may contain material from IETF Documents or IETF
   Contributions published or made publicly available before November
   10, 2008.  The person(s) controlling the copyright in some of this
   material may not have granted the IETF Trust the right to allow
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   Without obtaining an adequate license from the person(s) controlling
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   outside the IETF Standards Process, and derivative works of it may
   not be created outside the IETF Standards Process, except to format
   it for publication as an RFC or to translate it into languages other
   than English.

Table of Contents

   <a href="#section-1">1</a>.  Introduction . . . . . . . . . . . . . . . . . . . . . . . . .  <a href="#page-3">3</a>
     <a href="#section-1.1">1.1</a>.  Acquisition of RTP Streams vs. RTP Sessions  . . . . . . .  <a href="#page-6">6</a>
     <a href="#section-1.2">1.2</a>.  Outline  . . . . . . . . . . . . . . . . . . . . . . . . .  <a href="#page-6">6</a>
   <a href="#section-2">2</a>.  Requirements Notation  . . . . . . . . . . . . . . . . . . . .  <a href="#page-7">7</a>
   <a href="#section-3">3</a>.  Definitions  . . . . . . . . . . . . . . . . . . . . . . . . .  <a href="#page-7">7</a>
   <a href="#section-4">4</a>.  Elements of Delay in Multicast Applications  . . . . . . . . .  <a href="#page-8">8</a>
   5.  Protocol Design Considerations and Their Effect on
       Resource Management for Rapid Acquisition  . . . . . . . . . . <a href="#page-10">10</a>
   <a href="#section-6">6</a>.  Rapid Acquisition of Multicast RTP Sessions (RAMS) . . . . . . <a href="#page-12">12</a>
     <a href="#section-6.1">6.1</a>.  Overview . . . . . . . . . . . . . . . . . . . . . . . . . <a href="#page-12">12</a>
     <a href="#section-6.2">6.2</a>.  Message Flows  . . . . . . . . . . . . . . . . . . . . . . <a href="#page-13">13</a>
     <a href="#section-6.3">6.3</a>.  Synchronization of Primary Multicast Streams . . . . . . . <a href="#page-24">24</a>
     <a href="#section-6.4">6.4</a>.  Burst Shaping and Congestion Control in RAMS . . . . . . . <a href="#page-25">25</a>
     <a href="#section-6.5">6.5</a>.  Failure Cases  . . . . . . . . . . . . . . . . . . . . . . <a href="#page-27">27</a>
   <a href="#section-7">7</a>.  Encoding of the Signaling Protocol in RTCP . . . . . . . . . . <a href="#page-28">28</a>



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     <a href="#section-7.1">7.1</a>.  Extensions . . . . . . . . . . . . . . . . . . . . . . . . <a href="#page-29">29</a>
       <a href="#section-7.1.1">7.1.1</a>.  Vendor-Neutral Extensions  . . . . . . . . . . . . . . <a href="#page-30">30</a>
       <a href="#section-7.1.2">7.1.2</a>.  Private Extensions . . . . . . . . . . . . . . . . . . <a href="#page-30">30</a>
     <a href="#section-7.2">7.2</a>.  RAMS Request . . . . . . . . . . . . . . . . . . . . . . . <a href="#page-31">31</a>
     <a href="#section-7.3">7.3</a>.  RAMS Information . . . . . . . . . . . . . . . . . . . . . <a href="#page-34">34</a>
       <a href="#section-7.3.1">7.3.1</a>.  Response Code Definitions  . . . . . . . . . . . . . . <a href="#page-37">37</a>
     <a href="#section-7.4">7.4</a>.  RAMS Termination . . . . . . . . . . . . . . . . . . . . . <a href="#page-39">39</a>
   <a href="#section-8">8</a>.  SDP Signaling  . . . . . . . . . . . . . . . . . . . . . . . . <a href="#page-40">40</a>
     <a href="#section-8.1">8.1</a>.  Definitions  . . . . . . . . . . . . . . . . . . . . . . . <a href="#page-40">40</a>
     <a href="#section-8.2">8.2</a>.  Requirements . . . . . . . . . . . . . . . . . . . . . . . <a href="#page-41">41</a>
     <a href="#section-8.3">8.3</a>.  Example and Discussion . . . . . . . . . . . . . . . . . . <a href="#page-41">41</a>
   <a href="#section-9">9</a>.  NAT Considerations . . . . . . . . . . . . . . . . . . . . . . <a href="#page-44">44</a>
   <a href="#section-10">10</a>. Security Considerations  . . . . . . . . . . . . . . . . . . . <a href="#page-45">45</a>
   <a href="#section-11">11</a>. IANA Considerations  . . . . . . . . . . . . . . . . . . . . . <a href="#page-47">47</a>
     <a href="#section-11.1">11.1</a>. Registration of SDP Attributes . . . . . . . . . . . . . . <a href="#page-48">48</a>
     <a href="#section-11.2">11.2</a>. Registration of SDP Attribute Values . . . . . . . . . . . <a href="#page-48">48</a>
     <a href="#section-11.3">11.3</a>. Registration of FMT Values . . . . . . . . . . . . . . . . <a href="#page-48">48</a>
     <a href="#section-11.4">11.4</a>. SFMT Values for RAMS Messages Registry . . . . . . . . . . <a href="#page-48">48</a>
     <a href="#section-11.5">11.5</a>. RAMS TLV Space Registry  . . . . . . . . . . . . . . . . . <a href="#page-49">49</a>
     <a href="#section-11.6">11.6</a>. RAMS Response Code Space Registry  . . . . . . . . . . . . <a href="#page-50">50</a>
   <a href="#section-12">12</a>. Contributors . . . . . . . . . . . . . . . . . . . . . . . . . <a href="#page-52">52</a>
   <a href="#section-13">13</a>. Acknowledgments  . . . . . . . . . . . . . . . . . . . . . . . <a href="#page-52">52</a>
   <a href="#section-14">14</a>. References . . . . . . . . . . . . . . . . . . . . . . . . . . <a href="#page-52">52</a>
     <a href="#section-14.1">14.1</a>. Normative References . . . . . . . . . . . . . . . . . . . <a href="#page-52">52</a>
     <a href="#section-14.2">14.2</a>. Informative References . . . . . . . . . . . . . . . . . . <a href="#page-54">54</a>

<span class="h2"><a class="selflink" id="section-1" href="#section-1">1</a>.  Introduction</span>

   Most multicast flows carry a stream of inter-related data.  Receivers
   need to acquire certain information to start processing any data sent
   in the multicast session.  This document refers to this information
   as Reference Information.  The Reference Information is
   conventionally sent periodically in the multicast session (although
   its content can change over time) and usually consists of items such
   as a description of the schema for the rest of the data, references
   to which data to process, encryption information including keys, and
   any other information required to process the data in the multicast
   stream [<a href="#ref-IC2009" title="&quot;Reducing Channel Change Times in IPTV with Real-Time Transport Protocol (IEEE Internet Computing)&quot;">IC2009</a>].

   Real-time multicast applications require receivers to buffer data.
   Receivers may have to buffer data to smooth out the network jitter,
   to allow loss-repair methods such as Forward Error Correction and
   retransmission to recover the missing packets, and to satisfy the
   data-processing requirements of the application layer.

   When a receiver joins a multicast session, it has no control over
   what point in the flow is currently being transmitted.  Sometimes the
   receiver might join the session right before the Reference



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   Information is sent in the session.  In this case, the required
   waiting time is usually minimal.  Other times, the receiver might
   join the session right after the Reference Information has been
   transmitted.  In this case, the receiver has to wait for the
   Reference Information to appear again in the flow before it can start
   processing any multicast data.  In some other cases, the Reference
   Information is not contiguous in the flow but dispersed over a large
   period, which forces the receiver to wait for the whole Reference
   Information to arrive before starting to process the rest of the
   data.

   The net effect of waiting for the Reference Information and waiting
   for various buffers to fill up is that receivers can experience
   significantly large delays in data processing.  In this document, we
   refer to the difference between the time an RTP receiver wants to
   join the multicast session and the time the RTP receiver acquires all
   the necessary Reference Information as the Acquisition Delay.  The
   acquisition delay might not be the same for different receivers; it
   usually varies depending on the join time, length of the Reference
   Information repetition (or appearance) interval, and size of the
   Reference Information, as well as the application and transport
   properties.

   The varying nature of the acquisition delay adversely affects the
   receivers that frequently switch among multicast sessions.  While
   this problem equally applies to both any-source multicast (ASM) and
   source-specific multicast (SSM) applications, in this specification
   we address it for the SSM-based applications by describing a method
   that uses the fundamental tools offered by the existing RTP and RTCP
   protocols [<a href="./rfc3550" title="&quot;RTP: A Transport Protocol for Real-Time Applications&quot;">RFC3550</a>].  In this method, either the multicast source (or
   the distribution source in an SSM session) retains the Reference
   Information for a period after its transmission, or an intermediary
   network element (that we refer to as Retransmission Server) joins the
   multicast session and continuously caches the Reference Information
   as it is sent in the session and acts as a feedback target (see
   [<a href="./rfc5760" title="&quot;RTP Control Protocol (RTCP) Extensions for Single-Source Multicast Sessions with Unicast Feedback&quot;">RFC5760</a>]) for the session.  When an RTP receiver wishes to join the
   same multicast session, instead of simply issuing a Source Filtering
   Group Management Protocol (SFGMP) Join message, it sends a request to
   the feedback target for the session and asks for the Reference
   Information.  The retransmission server starts a new unicast RTP
   (retransmission) session and sends the Reference Information to the
   RTP receiver over that session.  If there is residual bandwidth, the
   retransmission server might burst the Reference Information faster
   than its natural rate.  As soon as the receiver acquires the
   Reference Information, it can join the multicast session and start
   processing the multicast data.  A simplified network diagram showing
   this method through an intermediary network element is depicted in
   Figure 1.



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<span class="grey"><a href="./rfc6285">RFC 6285</a>                          RAMS                         June 2011</span>


   This method potentially reduces the acquisition delay.  We refer to
   this method as Unicast-Based Rapid Acquisition of Multicast RTP
   Sessions.  A primary use case for this method is to reduce the
   channel-change times in IPTV networks where compressed video streams
   are multicast in different SSM sessions and viewers randomly join
   these sessions.

                                        -----------------------
                                  +---&gt;|     Intermediary      |
                                  |    |    Network Element    |
                                  | ...|(Retransmission Server)|
                                  | :   -----------------------
                                  | :
                                  | v
           -----------          ----------             ----------
          | Multicast |        |          |----------&gt;| Joining  |
          |  Source   |-------&gt;|  Router  |..........&gt;|   RTP    |
          |           |        |          |           | Receiver |
           -----------          ----------             ----------
                                    |
                                    |                  ----------
                                    +----------------&gt;| Existing |
                                                      |    RTP   |
                                                      | Receiver |
                                                       ----------


          -------&gt; Multicast RTP Flow
          .......&gt; Unicast RTP Flow

    Figure 1: Rapid Acquisition through an Intermediary Network Element

   A principle design goal in this solution is to use the existing tools
   in the RTP/RTCP protocol family.  This improves the versatility of
   the existing implementations and promotes faster deployment and
   better interoperability.  To this effect, we use the unicast
   retransmission support of RTP [<a href="./rfc4588" title="&quot;RTP Retransmission Payload Format&quot;">RFC4588</a>] and the capabilities of RTCP
   to handle the signaling needed to accomplish the acquisition.

   A reasonable effort has been made in this specification to design a
   solution that reliably works in both engineered and best-effort
   networks.  However, a proper congestion control combined with the
   desired behavior of this solution is difficult to achieve.  Rather,
   this solution has been designed based on the assumption that the
   retransmission server and the RTP receivers have some knowledge about
   where the bottleneck between them is.  This assumption does not
   generally hold unless both the retransmission server and the RTP
   receivers are in the same edge network.  Thus, this solution should



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   not be used across any best-effort path of the Internet.
   Furthermore, this solution should only be used in networks that are
   already carrying non-congestion-responsive multicast traffic and have
   throttling mechanisms in the retransmission servers to ensure the
   (unicast) burst traffic is a known constant upper-bound multiplier on
   the multicast load.

<span class="h3"><a class="selflink" id="section-1.1" href="#section-1.1">1.1</a>.  Acquisition of RTP Streams vs. RTP Sessions</span>

   In this memo, we describe a protocol that handles the rapid
   acquisition of a single multicast RTP session (called a primary
   multicast RTP session) carrying one or more RTP streams (called
   primary multicast streams).  If desired, multiple instances of this
   protocol may be run in parallel to acquire multiple RTP sessions
   simultaneously.

   When an RTP receiver requests the Reference Information from the
   retransmission server, it can opt to rapidly acquire a specific
   subset of the available RTP streams in the primary multicast RTP
   session.  Alternatively, the RTP receiver can request the rapid
   acquisition of all of the RTP streams in that RTP session.
   Regardless of how many RTP streams are requested by the RTP receiver
   or how many will be actually sent by the retransmission server, only
   one unicast RTP session will be established by the retransmission
   server.  This unicast RTP session is separate from the associated
   primary multicast RTP session.  As a result, there are always two
   different RTP sessions in a single instance of the rapid acquisition
   protocol:  (i) the primary multicast RTP session with its associated
   unicast feedback and (ii) the unicast RTP session.

   If the RTP receiver wants to rapidly acquire multiple RTP sessions
   simultaneously, separate unicast RTP sessions will be established for
   each of them.

<span class="h3"><a class="selflink" id="section-1.2" href="#section-1.2">1.2</a>.  Outline</span>

   The rest of this specification is as follows.  <a href="#section-3">Section 3</a> provides a
   list of the definitions frequently used in this document.  In
   <a href="#section-4">Section 4</a>, we describe the delay components in generic multicast
   applications.  <a href="#section-5">Section 5</a> presents an overview of the protocol design
   considerations for rapid acquisition.  We provide the protocol
   details of the rapid acquisition method in Sections <a href="#section-6">6</a> and <a href="#section-7">7</a>.
   Sections <a href="#section-8">8</a> and <a href="#section-9">9</a> discuss the Session Description Protocol (SDP)
   signaling issues with examples and NAT-related issues, respectively.
   Finally, <a href="#section-10">Section 10</a> discusses the security considerations, and
   <a href="#section-11">Section 11</a> details the IANA considerations.





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<span class="grey"><a href="./rfc6285">RFC 6285</a>                          RAMS                         June 2011</span>


<span class="h2"><a class="selflink" id="section-2" href="#section-2">2</a>.  Requirements Notation</span>

   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 [<a href="./rfc2119" title="&quot;Key words for use in RFCs to Indicate Requirement Levels&quot;">RFC2119</a>].

<span class="h2"><a class="selflink" id="section-3" href="#section-3">3</a>.  Definitions</span>

   This document uses the following acronyms and definitions frequently:

   (Primary) SSM Session:  The multicast session to which RTP receivers
   can join at a random point in time.  A primary SSM session can carry
   multiple RTP streams.

   Primary Multicast RTP Session:  The multicast RTP session an RTP
   receiver is interested in acquiring rapidly.  From the RTP receiver's
   viewpoint, the primary multicast RTP session has one associated
   unicast RTCP feedback stream to a Feedback Target, in addition to the
   primary multicast RTP stream(s).

   Primary Multicast (RTP) Streams:  The RTP stream(s) carried in the
   primary multicast RTP session.

   Source Filtering Group Management Protocol (SFGMP):  Following the
   definition in [<a href="./rfc4604" title="&quot;Using Internet Group Management Protocol Version 3 (IGMPv3) and Multicast Listener Discovery Protocol Version 2 (MLDv2) for Source- Specific Multicast&quot;">RFC4604</a>], SFGMP refers to the Internet Group
   Management Protocol (IGMP) version 3 [<a href="./rfc3376" title="&quot;Internet Group Management Protocol, Version 3&quot;">RFC3376</a>] and the Multicast
   Listener Discovery Protocol (MLD) version 2 [<a href="./rfc3810" title="&quot;Multicast Listener Discovery Version 2 (MLDv2) for IPv6&quot;">RFC3810</a>] in the IPv4 and
   IPv6 networks, respectively.  However, the rapid acquisition method
   introduced in this document does not depend on a specific version of
   either of these group management protocols.  In the remainder of this
   document, SFGMP will refer to any group management protocol that has
   Join and Leave functionalities.

   Feedback Target (FT):  Unicast RTCP feedback target as defined in
   [<a href="./rfc5760" title="&quot;RTP Control Protocol (RTCP) Extensions for Single-Source Multicast Sessions with Unicast Feedback&quot;">RFC5760</a>].  FT_Ap denotes a specific feedback target running on a
   particular address and port.

   Retransmission (or Burst) Packet:  An RTP packet that is formatted as
   defined in <a href="./rfc4588#section-4">Section&nbsp;4 of [RFC4588]</a>.  The payload of a retransmission
   or burst packet comprises the retransmission payload header followed
   by the payload of the original RTP packet.

   Reference Information:  The set of certain media content and metadata
   information that is sufficient for an RTP receiver to start usefully
   consuming a media stream.  The meaning, format, and size of this
   information are specific to the application and are out of the scope
   of this document.




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   Preamble Information:  A more compact form of the whole or a subset
   of the Reference Information transmitted out-of-band.

   (Unicast) Burst (or Retransmission) RTP Session:  The unicast RTP
   session used to send one or more unicast burst RTP streams and their
   associated RTCP messages.  The terms "burst RTP session" and
   "retransmission RTP session" can be used interchangeably.

   (Unicast) Burst (Stream):  A unicast stream of RTP retransmission
   packets that enable an RTP receiver to rapidly acquire the Reference
   Information associated with a primary multicast stream.  Each burst
   stream is identified by its Synchronization Source (SSRC) identifier
   that is unique in the primary multicast RTP session.  Following the
   session-multiplexing guidelines in [<a href="./rfc4588" title="&quot;RTP Retransmission Payload Format&quot;">RFC4588</a>], each unicast burst
   stream will use the same SSRC and Canonical Name (CNAME) as its
   primary multicast RTP stream.

   Retransmission Server (RS):  The RTP/RTCP endpoint that can generate
   the retransmission packets and the burst streams.  The RS may also
   generate other non-retransmission packets to aid rapid acquisition.

<span class="h2"><a class="selflink" id="section-4" href="#section-4">4</a>.  Elements of Delay in Multicast Applications</span>

   In a source-specific multicast (SSM) delivery system, there are three
   major elements that contribute to the acquisition delay when an RTP
   receiver switches from one multicast session to another one.  These
   are:

   o  Multicast-switching delay

   o  Reference Information latency

   o  Buffering delays

   Multicast-switching delay is the delay that is experienced when
   leaving the current multicast session (if any) and joining the new
   multicast session.  In typical systems, the multicast join and leave
   operations are handled by a group management protocol.  For example,
   the receivers and routers participating in a multicast session can
   use the Internet Group Management Protocol (IGMP) version 3 [<a href="./rfc3376" title="&quot;Internet Group Management Protocol, Version 3&quot;">RFC3376</a>]
   or the Multicast Listener Discovery Protocol (MLD) version 2
   [<a href="./rfc3810" title="&quot;Multicast Listener Discovery Version 2 (MLDv2) for IPv6&quot;">RFC3810</a>].  In either of these protocols, when a receiver wants to
   join a multicast session, it sends a message to its upstream router
   and the routing infrastructure sets up the multicast forwarding state
   to deliver the packets of the multicast session to the new receiver.
   The join times vary depending on the proximity of the upstream
   router, the current state of the multicast tree, the load on the
   system, and the protocol implementation.  Current systems provide



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<span class="grey"><a href="./rfc6285">RFC 6285</a>                          RAMS                         June 2011</span>


   join latencies, usually less than 200 milliseconds (ms).  If the
   receiver had been participating in another multicast session before
   joining the new session, it needs to send a Leave message to its
   upstream router to leave the session.  In common multicast routing
   protocols, the leave times are usually smaller than the join times;
   however, it is possible that the Leave and Join messages might get
   lost, in which case the multicast-switching delay inevitably
   increases.

   Reference Information latency is the time it takes the receiver to
   acquire the Reference Information.  It is highly dependent on the
   proximity of the actual time the receiver joined the session to the
   next time the Reference Information will be sent to the receivers in
   the session, whether or not the Reference Information is sent
   contiguously, and the size of the Reference Information.  For some
   multicast flows, there is a little or no interdependency in the data,
   in which case the Reference Information latency will be nil or
   negligible.  For other multicast flows, there is a high degree of
   interdependency.  One example of interest is the multicast flows that
   carry compressed audio/video.  For these flows, the Reference
   Information latency can become quite large and be a major contributor
   to the overall delay.

   The buffering component of the overall acquisition delay is driven by
   the way the application layer processes the payload.  In many
   multicast applications, an unreliable transport protocol such as UDP
   [<a href="./rfc0768" title="&quot;User Datagram Protocol&quot;">RFC0768</a>] is often used to transmit the data packets, and the
   reliability, if needed, is usually addressed through other means such
   as Forward Error Correction (e.g., [<a href="./rfc6015" title="&quot;RTP Payload Format for 1-D Interleaved Parity Forward Error Correction (FEC)&quot;">RFC6015</a>]) and retransmission.
   These loss-repair methods require buffering at the receiver side to
   function properly.  In many applications, it is also often necessary
   to de-jitter the incoming data packets before feeding them to the
   application.  The de-jittering process also increases the buffering
   delays.  Besides these network-related buffering delays, there are
   also specific buffering needs that are required by the individual
   applications.  For example, standard video decoders typically require
   a certain amount, sometimes up to a few seconds, of coded video data
   to be available in the pre-decoding buffers prior to starting to
   decode the video bitstream.












<span class="grey">Ver Steeg, et al.            Standards Track                    [Page 9]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-10" ></span>
<span class="grey"><a href="./rfc6285">RFC 6285</a>                          RAMS                         June 2011</span>


<span class="h2"><a class="selflink" id="section-5" href="#section-5">5</a>.  Protocol Design Considerations and Their Effect on Resource</span>
<span class="h2">    Management for Rapid Acquisition</span>

   This section is for informational purposes and does not contain
   requirements for implementations.

   Rapid acquisition is an optimization of a system that is expected to
   continue to work correctly and properly whether or not the
   optimization is effective or even fails due to lost control and
   feedback messages, congestion, or other problems.  This is
   fundamental to the overall design requirements surrounding the
   protocol definition and to the resource management schemes to be
   employed together with the protocol (e.g., Quality of Service (QoS)
   machinery, server load management, etc).  In particular, the system
   needs to operate within a number of constraints:

   o  First, a rapid acquisition operation must fail gracefully.  The
      user experience must not be significantly worse for trying and
      failing to complete rapid acquisition compared to simply joining
      the multicast session.

   o  Second, providing the rapid acquisition optimizations must not
      cause collateral damage to either the multicast session being
      joined or other multicast sessions sharing resources with the
      rapid acquisition operation.  In particular, the rapid acquisition
      operation must avoid interference with the multicast session that
      might be simultaneously being received by other hosts.  In
      addition, it must also avoid interference with other multicast and
      non-multicast sessions sharing the same network resources.  These
      properties are possible but are usually difficult to achieve.

   One challenge is the existence of multiple bandwidth bottlenecks
   between the receiver and the server(s) in the network providing the
   rapid acquisition service.  In commercial IPTV deployments, for
   example, bottlenecks are often present in the aggregation network
   connecting the IPTV servers to the network edge, the access links
   (e.g., DSL, Data Over Cable Service Interface Specification
   (DOCSIS)), and the home network of the subscribers.  Some of these
   links might serve only a single subscriber, limiting congestion
   impact to the traffic of only that subscriber, but others can be
   shared links carrying multicast sessions of many subscribers.  Also
   note that the state of these links can vary over time.  The receiver
   might have knowledge of a portion of this network or might have
   partial knowledge of the entire network.  The methods employed by the
   devices to acquire this network state information is out of the scope
   of this document.  The receiver should be able to signal the server
   with the bandwidth that it believes it can handle.  The server also
   needs to be able to rate limit the flow in order to stay within the



<span class="grey">Ver Steeg, et al.            Standards Track                   [Page 10]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-11" ></span>
<span class="grey"><a href="./rfc6285">RFC 6285</a>                          RAMS                         June 2011</span>


   performance envelope that it knows about.  Both the server and
   receiver need to be able to inform the other of changes in the
   requested and delivered rates.  However, the protocol must be robust
   in the presence of packet loss, so this signaling must include the
   appropriate default behaviors.

   A second challenge is that for some uses (e.g., high-bitrate video)
   the unicast burst bitrate is high while the flow duration of the
   unicast burst is short.  This is because the purpose of the unicast
   burst is to allow the RTP receiver to join the multicast quickly and
   thereby limit the overall resources consumed by the burst.  Such
   high-bitrate, short-duration flows are not amenable to conventional
   admission-control techniques.  For example, end-to-end per-flow
   signaled admission-control techniques such as Resource Reservation
   Protocol (RSVP) have too much latency and control channel overhead to
   be a good fit for rapid acquisition.  Similarly, using a TCP (or TCP-
   like) approach with a 3-way handshake and slow-start to avoid
   inducing congestion would defeat the purpose of attempting rapid
   acquisition in the first place by introducing many round-trip times
   (RTTs) of delay.

   These observations lead to certain unavoidable requirements and goals
   for a rapid acquisition protocol.  These are:

   o  The protocol must be designed to allow a deterministic upper bound
      on the extra bandwidth used (compared to just joining the
      multicast session).  A reasonable size bound is e*B, where B is
      the nominal bandwidth of the primary multicast streams and e is an
      excess-bandwidth coefficient.  The total duration of the unicast
      burst must have a reasonable bound; long unicast bursts devolve to
      the bandwidth profile of multi-unicast for the whole system.

   o  The scheme should minimize (or better eliminate) the overlap of
      the unicast burst and the primary multicast stream.  This
      minimizes the window during which congestion could be induced on a
      bottleneck link compared to just carrying the multicast or unicast
      packets alone.

   o  The scheme must minimize (or better eliminate) any gap between the
      unicast burst and the primary multicast stream, which has to be
      repaired later or, in the absence of repair, will result in loss
      being experienced by the application.

   In addition to the above, there are some other protocol design issues
   to be considered.  First, there is at least one RTT of "slop" in the
   control loop.  In starting a rapid acquisition burst, this manifests
   as the time between the client requesting the unicast burst and the
   burst description and/or the first unicast burst packets arriving at



<span class="grey">Ver Steeg, et al.            Standards Track                   [Page 11]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-12" ></span>
<span class="grey"><a href="./rfc6285">RFC 6285</a>                          RAMS                         June 2011</span>


   the receiver.  For managing and terminating the unicast burst, there
   are two possible approaches for the control loop.  First, the
   receiver can adapt to the unicast burst as received, converge based
   on observation, and explicitly terminate the unicast burst with a
   second control loop exchange (which takes a minimum of one RTT, just
   as starting the unicast burst does).  Alternatively, the server
   generating the unicast burst can precompute the burst parameters
   based on the information in the initial request and tell the receiver
   the burst duration.

   The protocol described in the next section allows either method of
   controlling the rapid acquisition unicast burst.

<span class="h2"><a class="selflink" id="section-6" href="#section-6">6</a>.  Rapid Acquisition of Multicast RTP Sessions (RAMS)</span>

   We start this section with an overview of the Rapid Acquisition of
   Multicast RTP Sessions (RAMS) method.

<span class="h3"><a class="selflink" id="section-6.1" href="#section-6.1">6.1</a>.  Overview</span>

   [<a id="ref-RFC5760">RFC5760</a>] specifies an extension to the RTP Control Protocol (RTCP)
   to use unicast feedback in an SSM session.  It defines an
   architecture that introduces the concept of Distribution Source,
   which, in an SSM context, distributes the RTP data and redistributes
   RTCP information to all RTP receivers.  This RTCP information is
   retrieved from the Feedback Target, to which RTCP unicast feedback
   traffic is sent.  One or more entities different from the
   Distribution Source MAY implement the feedback target functionality,
   and different RTP receivers MAY use different feedback targets.

   This document builds further on these concepts to reduce the
   acquisition delay when an RTP receiver joins a multicast session at a
   random point in time by introducing the concept of the Burst Source
   and new RTCP feedback messages.  The Burst Source has a cache where
   the most recent packets from the primary multicast RTP session are
   continuously stored.  When an RTP receiver wants to receive a primary
   multicast stream, it can first request a unicast burst from the Burst
   Source before it joins the SSM session.  In this burst, the packets
   are formatted as RTP retransmission packets [<a href="./rfc4588" title="&quot;RTP Retransmission Payload Format&quot;">RFC4588</a>] and carry
   Reference Information.  This information allows the RTP receiver to
   start usefully consuming the RTP packets sent in the primary
   multicast RTP session.

   Using an accelerated bitrate (as compared to the nominal bitrate of
   the primary multicast stream) for the unicast burst implies that at a
   certain point in time, the payload transmitted in the unicast burst
   is going to be the same as the payload in the associated multicast
   stream, i.e., the unicast burst will catch up with the primary



<span class="grey">Ver Steeg, et al.            Standards Track                   [Page 12]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-13" ></span>
<span class="grey"><a href="./rfc6285">RFC 6285</a>                          RAMS                         June 2011</span>


   multicast stream.  At this point, the RTP receiver no longer needs to
   receive the unicast burst and can join the SSM session.  This method
   is referred to as the Rapid Acquisition of Multicast RTP Sessions
   (RAMS).

   This document defines extensions to [<a href="./rfc4585" title="&quot;Extended RTP Profile for Real-time Transport Control Protocol (RTCP)-Based Feedback (RTP/AVPF)&quot;">RFC4585</a>] for an RTP receiver to
   request a unicast burst as well as for additional control messaging
   that can be leveraged during the acquisition process.

<span class="h3"><a class="selflink" id="section-6.2" href="#section-6.2">6.2</a>.  Message Flows</span>

   As shown in Figure 2, the main entities involved in rapid acquisition
   and the message flows are:

   o  Multicast Source

   o  Feedback Target (FT)

   o  Burst/Retransmission Source (BRS)

   o  RTP Receiver (RTP_Rx)






























<span class="grey">Ver Steeg, et al.            Standards Track                   [Page 13]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-14" ></span>
<span class="grey"><a href="./rfc6285">RFC 6285</a>                          RAMS                         June 2011</span>


    -----------                                       --------------
   |           |------------------------------------&gt;|              |
   |           |.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-&gt;|              |
   |           |                                     |              |
   | Multicast |          ----------------           |              |
   |  Source   |         | Retransmission |          |              |
   |           |--------&gt;|  Server  (RS)  |          |              |
   |           |.-.-.-.-&gt;|                |          |              |
   |           |         |  ------------  |          |              |
    -----------          | |  Feedback  | |&lt;.=.=.=.=.|              |
                         | | Target (FT)| |&lt;~~~~~~~~~| RTP Receiver |
   PRIMARY MULTICAST     |  ------------  |          |   (RTP_Rx)   |
   RTP SESSION with      |                |          |              |
   UNICAST FEEDBACK      |                |          |              |
                         |                |          |              |
   - - - - - - - - - - - |- - - - - - - - |- - - - - |- - - - - - - |- -
                         |                |          |              |
   UNICAST BURST         |  ------------  |          |              |
   (or RETRANSMISSION)   | |   Burst/   | |&lt;~~~~~~~~&gt;|              |
   RTP SESSION           | |  Retrans.  | |.........&gt;|              |
                         | |Source (BRS)| |&lt;.=.=.=.=&gt;|              |
                         |  ------------  |          |              |
                         |                |          |              |
                          ----------------            --------------

   -------&gt; Multicast RTP Flow
   .-.-.-.&gt; Multicast RTCP Flow
   .=.=.=.&gt; Unicast RTCP Reports
   ~~~~~~~&gt; Unicast RTCP Feedback Messages
   .......&gt; Unicast RTP Flow

        Figure 2: Flow Diagram for Unicast-Based Rapid Acquisition

   As defined in [<a href="./rfc5760" title="&quot;RTP Control Protocol (RTCP) Extensions for Single-Source Multicast Sessions with Unicast Feedback&quot;">RFC5760</a>], the feedback target (FT) is the entity to
   which the RTP_Rx sends its RTCP feedback messages indicating packet
   loss by means of an RTCP NACK message or indicating RTP_Rx's desire
   to rapidly acquire the primary multicast RTP session by means of an
   RTCP feedback message defined in this document.  While the Burst/
   Retransmission Source (BRS) is responsible for responding to these
   messages and for further RTCP interaction with the RTP_Rx in the case
   of a rapid acquisition process, it is assumed in the remainder of
   this document that these two logical entities (FT and BRS) are
   combined in a single physical entity and they share state.  In the
   remainder of the text, the term Retransmission Server (RS) is used
   whenever appropriate, to refer to this single physical entity.






<span class="grey">Ver Steeg, et al.            Standards Track                   [Page 14]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-15" ></span>
<span class="grey"><a href="./rfc6285">RFC 6285</a>                          RAMS                         June 2011</span>


   The FT is involved in the primary multicast RTP session and receives
   unicast feedback for that session as in [<a href="./rfc5760" title="&quot;RTP Control Protocol (RTCP) Extensions for Single-Source Multicast Sessions with Unicast Feedback&quot;">RFC5760</a>].  The BRS is
   involved in the unicast burst (or retransmission) RTP session and
   transmits the unicast burst and retransmission packets formatted as
   RTP retransmission packets [<a href="./rfc4588" title="&quot;RTP Retransmission Payload Format&quot;">RFC4588</a>] in a single separate unicast RTP
   session to each RTP_Rx.  In the unicast burst RTP session, as in any
   other RTP session, the BRS and RTP_Rx regularly send the periodic
   sender and receiver reports, respectively.

   The unicast burst is started by an RTCP feedback message that is
   defined in this document based on the common packet format provided
   in [<a href="./rfc4585" title="&quot;Extended RTP Profile for Real-time Transport Control Protocol (RTCP)-Based Feedback (RTP/AVPF)&quot;">RFC4585</a>].  An RTP retransmission is triggered by an RTCP NACK
   message defined in [<a href="./rfc4585" title="&quot;Extended RTP Profile for Real-time Transport Control Protocol (RTCP)-Based Feedback (RTP/AVPF)&quot;">RFC4585</a>].  Both of these messages are sent to the
   FT of the primary multicast RTP session and can start the unicast
   burst/retransmission RTP session.

   In the extended RTP profile for RTCP-based feedback (RTP/Audio-Visual
   Profile with Feedback (AVPF)), there are certain rules that apply to
   scheduling of both of these messages sent to the FT in the primary
   multicast RTP session; these are detailed in <a href="./rfc4585#section-3.5">Section&nbsp;3.5 of
   [RFC4585]</a>.  One of the rules states that in a multi-party session
   such as the SSM sessions we are considering in this specification, an
   RTP_Rx cannot send an RTCP feedback message for a minimum of one
   second after joining the session (i.e., Tmin=1.0 second).  While this
   rule has the goal of avoiding problems associated with flash crowds
   in typical multi-party sessions, it defeats the purpose of rapid
   acquisition.  Furthermore, when RTP receivers delay their messages
   requesting a burst by a deterministic or even a random amount, it
   still does not make a difference since such messages are not related
   and their timings are independent from each other.  Thus, in this
   specification, we initialize Tmin to zero and allow the RTP receivers
   to send a burst request message right at the beginning.  For the
   subsequent messages (e.g., updated requests) during rapid
   acquisition, the timing rules of [<a href="./rfc4585" title="&quot;Extended RTP Profile for Real-time Transport Control Protocol (RTCP)-Based Feedback (RTP/AVPF)&quot;">RFC4585</a>] still apply.

   Figure 3 depicts an example of messaging flow for rapid acquisition.
   The RTCP feedback messages are explained below.  The optional
   messages are indicated in parentheses, and they might or might not be
   present during rapid acquisition.  In a scenario where rapid
   acquisition is performed by a feedback target co-located with the
   media sender, the same method (with the identical message flows)
   still applies.









<span class="grey">Ver Steeg, et al.            Standards Track                   [Page 15]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-16" ></span>
<span class="grey"><a href="./rfc6285">RFC 6285</a>                          RAMS                         June 2011</span>


                  -------------------------
                 | Retransmission  Server  |
    -----------  |  ------   ------------  |   --------    ------------
   | Multicast | | |  FT  | | Burst/Ret. | |  |        |  |    RTP     |
   |  Source   | | |      | |   Source   | |  | Router |  |  Receiver  |
   |           | |  ------   ------------  |  |        |  |  (RTP_Rx)  |
    -----------  |      |         |        |   --------    ------------
     |            -------------------------       |                |
     |                  |         |               |                |
     |-- RTP Multicast ----------&gt;---------------&gt;|                |
     |-. RTCP Multicast -.-.-.-.-&gt;-.-.-.-.-.-.-.-&gt;|                |
     |                  |         |               |                |
     |                  |         |********************************|
     |                  |         |*      PORT MAPPING SETUP      *|
     |                  |         |********************************|
     |                  |         |               |                |
     |                  |&lt;~~~~~~~~~~~~~~~~~~~~~~~~~ RTCP RAMS-R ~~~|
     |                  |         |               |                |
     |                  |         |********************************|
     |                  |         |* UNICAST SESSION ESTABLISHED  *|
     |                  |         |********************************|
     |                  |         |               |                |
     |                  |         |~~~ RTCP RAMS-I ~~~~~~~~~~~~~~~&gt;|
     |                  |         |               |                |
     |                  |         |... Unicast RTP Burst .........&gt;|
     |                  |         |               |                |
     |                  |&lt;~~~~~~~~~~~~~~~~~~~~~~~~ (RTCP RAMS-R) ~~|
     |                  |         |               |                |
     |                  |         |~~ (RTCP RAMS-I) ~~~~~~~~~~~~~~&gt;|
     |                  |         |               |                |
     |                  |         |               |                |
     |                  |         |               |&lt;= SFGMP Join ==|
     |                  |         |               |                |
     |-- RTP Multicast -------------------------------------------&gt;|
     |-. RTCP Multicast -.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.&gt;|
     |                  |         |               |                |
     |                  |         |               |                |
     |                  |         |&lt;~~~~~~~~~~~~~~~ RTCP RAMS-T ~~~|
     |                  |         |               |                |
     :                  :         :               :                :
     |                  |         |&lt;.=.= Unicast RTCP Reports .=.=&gt;|
     :                  :         :     for the unicast session    :
     |                  |         |               |                |








<span class="grey">Ver Steeg, et al.            Standards Track                   [Page 16]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-17" ></span>
<span class="grey"><a href="./rfc6285">RFC 6285</a>                          RAMS                         June 2011</span>


   -------&gt; Multicast RTP Flow
   .-.-.-.&gt; Multicast RTCP Flow
   .=.=.=.&gt; Unicast RTCP Reports
   ~~~~~~~&gt; Unicast RTCP Feedback Messages
   =======&gt; SFGMP Messages
   .......&gt; Unicast RTP Flow

        Figure 3: Message Flows for Unicast-Based Rapid Acquisition

   This document defines the expected behaviors of the RS and RTP_Rx.
   It is instructive to consider independently operating implementations
   on the RS and RTP_Rx that request the burst, describe the burst,
   start the burst, join the multicast session, and stop the burst.
   These implementations send messages to each other, but provisions are
   needed for the cases where the control messages get lost, or
   reordered, or are not being delivered to their destinations.

   The following steps describe rapid acquisition in detail:

   1.   Port Mapping Setup:  For the primary multicast RTP session, the
        RTP and RTCP destination ports are declaratively specified
        (refer to <a href="#section-8">Section 8</a> for examples in SDP).  However, the RTP_Rx
        needs to choose its RTP and RTCP receive ports for the unicast
        burst RTP session.  Since this unicast session is established
        after the RTP_Rx has sent a RAMS Request (RAMS-R) message as
        unicast feedback in the primary multicast RTP session, the
        RTP_Rx MUST first set up the port mappings between the unicast
        and multicast sessions and send this mapping information to the
        FT along with the RAMS-R message so that the BRS knows how to
        communicate with the RTP_Rx.

        The details of this setup procedure are explained in [<a href="./rfc6284" title="&quot;Port Mapping Between Unicast and Multicast RTP Sessions&quot;">RFC6284</a>].
        Other NAT-related issues are left to <a href="#section-9">Section 9</a> to keep the
        present discussion focused on the RAMS message flows.

   2.   Request:  The RTP_Rx sends a rapid acquisition request (RAMS-R)
        for the primary multicast RTP session to the unicast feedback
        target of that session.  The request contains the SSRC
        identifier of the RTP_Rx (aka SSRC of packet sender) and can
        contain the media sender SSRC identifier(s) of the primary
        multicast stream(s) of interest (aka SSRC of media source).  The
        RAMS-R message can contain parameters that constrain the burst,
        such as the buffer and bandwidth limits.

        Before joining the SSM session, the RTP_Rx learns the addresses
        for the multicast source, group, and RS by out-of-band means.
        If the RTP_Rx desires to rapidly acquire only a subset of the
        primary multicast streams available in the primary multicast RTP



<span class="grey">Ver Steeg, et al.            Standards Track                   [Page 17]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-18" ></span>
<span class="grey"><a href="./rfc6285">RFC 6285</a>                          RAMS                         June 2011</span>


        session, the RTP_Rx MUST also acquire the SSRC identifiers for
        the desired RTP streams out-of-band.  Based on this information,
        the RTP_Rx populates the desired SSRC(s) in the RAMS-R message.

        When the FT successfully receives the RAMS-R message, the BRS
        responds to it by accepting or rejecting the request.
        Immediately before the BRS sends any RTP or RTCP packet(s)
        described below, it establishes the unicast burst RTP session.

   3.   Response:  The BRS sends RAMS Information (RAMS-I) message(s) to
        the RTP_Rx to convey the status for the burst(s) requested by
        the RTP_Rx.

        If the primary multicast RTP session associated with the FT_Ap
        (a specific feedback target running on a particular address and
        port) on which the RAMS-R message was received contains only a
        single primary multicast stream, the BRS SHALL always use the
        SSRC of the RTP stream associated with the FT_Ap in the RAMS-I
        message(s) regardless of the media sender SSRC requested in the
        RAMS-R message.  In such cases the 'ssrc' attribute MAY be
        omitted from the media description.  If the requested SSRC and
        the actual media sender SSRC do not match, the BRS MUST
        explicitly populate the correct media sender SSRC in the initial
        RAMS-I message (see <a href="#section-7.3">Section 7.3</a>).

        The FT_Ap could also be associated with an RTP session that
        carries two or more primary multicast streams.  If the RTP_Rx
        wants to issue a collective request to receive the whole primary
        multicast RTP session, it does not need the 'ssrc' attributes to
        be described in the media description.

        If the FT_Ap is associated with two or more RTP sessions,
        RTP_Rx's request will be ambiguous.  To avoid any ambiguity,
        each FT_Ap MUST be only associated with a single RTP session.

        If the RTP_Rx is willing to rapidly acquire only a subset of the
        primary multicast streams, the RTP_Rx MUST list all the media
        sender SSRC(s) denoting the stream(s) it wishes to acquire in
        the RAMS-R message.  Upon receiving such a message, the BRS MAY
        accept the request for all or a subset of the media sender
        SSRC(s) that match the RTP stream(s) it serves.  The BRS MUST
        reject all other requests with an appropriate response code.

        *  Reject Responses:  The BRS MUST take into account any
           limitations that may have been specified by the RTP_Rx in the
           RAMS-R message when making a decision regarding the request.
           If the RTP_Rx has requested to acquire the whole primary
           multicast RTP session but the BRS cannot provide a rapid



<span class="grey">Ver Steeg, et al.            Standards Track                   [Page 18]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-19" ></span>
<span class="grey"><a href="./rfc6285">RFC 6285</a>                          RAMS                         June 2011</span>


           acquisition service for any of the primary multicast streams,
           the BRS MUST reject the request via a single RAMS-I message
           with a collective reject response code, which is defined as
           510 in <a href="#section-11.6">Section 11.6</a> and whose media sender SSRC field is set
           to one of SSRCs served by this FT_Ap.  Upon receiving this
           RAMS-I message, the RTP_Rx abandons the rapid acquisition
           attempt and can immediately join the multicast session by
           sending an SFGMP Join message towards its upstream multicast
           router.

           In all other cases, the BRS MUST send a separate RAMS-I
           message with the appropriate 5xx-level response code (as
           defined in <a href="#section-11.6">Section 11.6</a>) for each primary multicast stream
           that has been requested by the RTP_Rx but cannot be served by
           the BRS.  There could be multiple reasons why the BRS has
           rejected the request; however, the BRS chooses the most
           appropriate response code to inform the RTP_Rx.

           Upon receiving a reject response indicating a transient
           problem such as insufficient BRS or network resources, if the
           RTP_Rx wants to retry sending the same request, the RTP_Rx
           MUST follow the RTCP timer rules of [<a href="./rfc4585" title="&quot;Extended RTP Profile for Real-time Transport Control Protocol (RTCP)-Based Feedback (RTP/AVPF)&quot;">RFC4585</a>] to allow the
           transient problems to go away.  However, if the reject
           response indicates a non-transient problem (such as the ones
           reported by response codes 504, 505, and 506), the RTP_Rx
           MUST NOT attempt a retry.  Different response codes have
           different scopes.  Refer to <a href="#section-7.3.1">Section 7.3.1</a> for details.

           The BRS can employ a policing mechanism against the broken
           RTP_Rx implementations that are not following these rules.
           See <a href="#section-10">Section 10</a> for details.

        *  Accept Responses:  The BRS MUST send at least one separate
           RAMS-I message with the appropriate response code (either
           zero indicating a private response or response code 200
           indicating success as listed in <a href="#section-11.6">Section 11.6</a>) for each
           primary multicast stream that has been requested by the
           RTP_Rx and will be served by the BRS.  Such RAMS-I messages
           comprise fields that can be used to describe the individual
           unicast burst streams.  When there is a RAMS-R request for
           multiple primary multicast streams, the BRS MUST include all
           the individual RAMS-I messages corresponding to that RAMS-R
           request in the same compound RTCP packet if these messages
           fit in the same packet.  Note that if the BRS is sending only
           the preamble information but not the whole unicast burst
           stream, it will not accept the request but will send a
           response code 511 instead, as defined in <a href="#section-11.6">Section 11.6</a>.




<span class="grey">Ver Steeg, et al.            Standards Track                   [Page 19]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-20" ></span>
<span class="grey"><a href="./rfc6285">RFC 6285</a>                          RAMS                         June 2011</span>


           The RAMS-I message carries the RTP sequence number of the
           first packet transmitted in the respective RTP stream to
           allow the RTP_Rx to detect any missing initial packet(s).
           When the BRS accepts a request for a primary multicast
           stream, this field MUST always be populated in the RAMS-I
           message(s) sent for this particular primary multicast stream.
           It is RECOMMENDED that the BRS sends a RAMS-I message at the
           start of the burst so that the RTP_Rx can quickly detect any
           missing initial packet(s).

        It is possible that the RAMS-I message for a primary multicast
        stream can get delayed or lost, and the RTP_Rx can start
        receiving RTP packets before receiving a RAMS-I message.  An
        RTP_Rx implementation MUST NOT assume it will quickly receive
        the initial RAMS-I message.  For redundancy purposes, it is
        RECOMMENDED that the BRS repeats the RAMS-I messages multiple
        times as long as it follows the RTCP timer rules defined in
        [<a href="./rfc4585" title="&quot;Extended RTP Profile for Real-time Transport Control Protocol (RTCP)-Based Feedback (RTP/AVPF)&quot;">RFC4585</a>].

   4.   Unicast Burst:  For the primary multicast stream(s) for which
        the request is accepted, the BRS starts sending the unicast
        burst(s) that comprises one or more RTP retransmission packets
        sent in the unicast burst RTP session.  In some applications,
        the BRS can send preamble information data to the RTP_Rx in
        addition to the requested burst to prime the media decoder(s).
        However, for the BRS to send the preamble information in a
        particular format, explicit signaling from the RTP_Rx is
        required.  In other words, the BRS MUST NOT send preamble
        information in a particular format unless the RTP_Rx has
        signaled support for that format in the RAMS-R message through a
        public or private extension as defined in <a href="#section-7.1">Section 7.1</a>.

        The format of this preamble data is RTP-payload specific and not
        specified here.

   5.   Updated Request:  The RTP_Rx MAY send an updated RAMS-R message
        (as unicast feedback in the primary multicast RTP session) with
        a different value for one or more fields of an earlier RAMS-R
        message.  The BRS MUST be able to detect whether a burst is
        already planned for or being transmitted to this particular
        RTP_Rx for this particular media sender SSRC.  If there is an
        existing burst planned for or being transmitted, the newly
        arriving RAMS-R is an updated request; otherwise, it is a new
        request.  It is also possible that the RTP_Rx has sent an
        improperly formatted RAMS-R message or used an invalid value in
        the RAMS-R message.  If notified by the BRS using a 4xx-level





<span class="grey">Ver Steeg, et al.            Standards Track                   [Page 20]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-21" ></span>
<span class="grey"><a href="./rfc6285">RFC 6285</a>                          RAMS                         June 2011</span>


        response code (as defined in <a href="#section-11.6">Section 11.6</a>) and only after
        following the timing rules of [<a href="./rfc4585" title="&quot;Extended RTP Profile for Real-time Transport Control Protocol (RTCP)-Based Feedback (RTP/AVPF)&quot;">RFC4585</a>], the RTP_Rx MAY resend
        its corrected request.

        The BRS determines the identity of the requesting RTP_Rx by
        looking at its canonical name identifier (CNAME item in the
        source description (SDES)).  Thus, the RTP_Rx MUST choose a
        method that ensures it uses a unique CNAME identifier.  Such
        methods are provided in [<a href="./rfc6222" title="&quot;Guidelines for Choosing RTP Control Protocol (RTCP) Canonical Names (CNAMEs)&quot;">RFC6222</a>].  In addition to one or more
        fields with updated values, an updated RAMS-R message may also
        include the fields whose values did not change.

        Upon receiving an updated request, the BRS can use the updated
        values for sending/shaping the burst or refine the values and
        use the refined values for sending/shaping the burst.
        Subsequently, the BRS MAY send an updated RAMS-I message in the
        unicast burst RTP session to indicate the changes it made.

        It is an implementation-dependent decision for an RTP_RX whether
        and when to send an updated request.  However, note that the
        updated request messages can get delayed and arrive at the BRS
        after the initial unicast burst was completed.  In this case,
        the updated request message can trigger a new unicast burst, and
        by then if the RTP_Rx has already started receiving multicast
        data, a congestion is likely to occur.  In this case, the RTP_Rx
        can detect this only after a delay, and then it can try to
        terminate the new burst.  However, in the meantime, the RTP_Rx
        can experience packet loss or other problems.  This and other
        similar corner cases SHOULD be carefully examined if updated
        requests are to be used.

   6.   Updated Response:  The BRS can send more than one RAMS-I message
        in the unicast burst RTP session, e.g., to update the value of
        one or more fields in an earlier RAMS-I message.  The updated
        RAMS-I messages might or might not be a direct response to a
        RAMS-R message.  The BRS can also send updated RAMS-I messages
        to signal the RTP_Rx in real time to join the SSM session when
        the BRS had already sent an initial RAMS-I message, e.g., at the
        start of the burst.  The RTP_Rx depends on the BRS to learn the
        join time, which can be conveyed by the first RAMS-I message or
        can be sent/revised in a later RAMS-I message.  If the BRS is
        not capable of determining the join time in the initial RAMS-I
        message, the BRS MUST send another RAMS-I message (with the join
        time information) later.

   7.   Multicast Join Signaling:  The RAMS-I message allows the BRS to
        signal explicitly when the RTP_Rx needs to send the SFGMP Join
        message.  The RTP_Rx SHOULD use this information from the most



<span class="grey">Ver Steeg, et al.            Standards Track                   [Page 21]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-22" ></span>
<span class="grey"><a href="./rfc6285">RFC 6285</a>                          RAMS                         June 2011</span>


        recent RAMS-I message unless it has more accurate information.
        If the request is accepted, this information MUST be conveyed in
        at least one RAMS-I message, and its value MAY be updated by
        subsequent RAMS-I messages.

        There can be missing packets if the RTP_Rx joins the multicast
        session too early or too late.  For example, if the RTP_Rx
        starts receiving the primary multicast stream while it is still
        receiving the unicast burst at a high excess bitrate, this can
        result in an increased risk of packet loss.  Or, if the RTP_Rx
        joins the multicast session some time after the unicast burst is
        finished, there can be a gap between the burst and multicast
        data (a number of RTP packets might be missing).  In both cases,
        the RTP_Rx can issue retransmission requests (via RTCP NACK
        messages sent as unicast feedback in the primary multicast RTP
        session) [<a href="./rfc4585" title="&quot;Extended RTP Profile for Real-time Transport Control Protocol (RTCP)-Based Feedback (RTP/AVPF)&quot;">RFC4585</a>] to the FT entity of the RS to fill the gap.
        The BRS might or might not respond to such requests.  When it
        responds and the response causes significant changes in one or
        more values reported earlier to the RTP_Rx, an updated RAMS-I
        SHOULD be sent to the RTP_Rx.

   8.   Multicast Receive:  After the join, the RTP_Rx starts receiving
        the primary multicast stream(s).

   9.   Terminate:  The BRS can know when it needs to ultimately stop
        the unicast burst based on its parameters.  However, the RTP_Rx
        may need to ask the BRS to terminate the burst prematurely or at
        a specific sequence number.  For this purpose, the RTP_Rx uses
        the RAMS Termination (RAMS-T) message sent as RTCP feedback in
        the unicast burst RTP session.  A separate RAMS-T message is
        sent for each primary multicast stream served by the BRS unless
        an RTCP BYE message has been sent in the unicast burst RTP
        session as described in Step 10.  For the burst requests that
        were rejected by the BRS, there is no need to send a RAMS-T
        message.

        If the RTP_Rx wants to terminate a burst prematurely, it MUST
        send a RAMS-T message for the SSRC of the primary multicast
        stream it wishes to terminate.  This message is sent in the
        unicast burst RTP session.  Upon receiving this message, the BRS
        MUST terminate the unicast burst.  If the RTP_Rx requested to
        acquire the entire primary multicast RTP session but wants to
        terminate this request before it learns the individual media
        sender SSRC(s) via RAMS-I message(s) or RTP packets, the RTP_Rx
        cannot use RAMS-T message(s) and thus MUST send an RTCP BYE
        message in the unicast burst RTP session to terminate the
        request.




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<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-23" ></span>
<span class="grey"><a href="./rfc6285">RFC 6285</a>                          RAMS                         June 2011</span>


        Otherwise, the default behavior for the RTP_Rx is to send a
        RAMS-T message in the unicast burst RTP session immediately
        after it joins the multicast session and has started receiving
        multicast packets.  In that case, the RTP_Rx MUST send a RAMS-T
        message with the sequence number of the first RTP packet
        received in the primary multicast stream.  Then, the BRS MUST
        terminate the respective burst after it sends the unicast burst
        packet whose Original Sequence Number (OSN) field in the RTP
        retransmission payload header matches this number minus one.  If
        the BRS has already sent that unicast burst packet when the
        RAMS-T message arrives, the BRS MUST terminate the respective
        burst immediately.

        If an RTCP BYE message has not been issued yet as described in
        Step 10, the RTP_Rx MUST send at least one RAMS-T message for
        each primary multicast stream served by the BRS.  The RAMS-T
        message(s) MUST be sent to the BRS in the unicast burst RTP
        session.  Against the possibility of a message loss, it is
        RECOMMENDED that the RTP_Rx repeats the RAMS-T messages multiple
        times as long as it follows the RTCP timer rules defined in
        [<a href="./rfc4585" title="&quot;Extended RTP Profile for Real-time Transport Control Protocol (RTCP)-Based Feedback (RTP/AVPF)&quot;">RFC4585</a>].

        The binding between RAMS-T and ongoing bursts is achieved
        through RTP_Rx's CNAME identifier and packet sender and media
        sender SSRCs.  Choosing a globally unique CNAME makes sure that
        the RAMS-T messages are processed correctly.

   10.  Terminate with RTCP BYE:  When the RTP_Rx is receiving one or
        more burst streams, if the RTP_Rx becomes no longer interested
        in acquiring any of the primary multicast streams, the RTP_Rx
        SHALL issue an RTCP BYE message for the unicast burst RTP
        session and another RTCP BYE message for the primary multicast
        RTP session.  These RTCP BYE messages are sent to the BRS and FT
        logical entities, respectively.

        Upon receiving an RTCP BYE message, the BRS MUST terminate the
        rapid acquisition operation and cease transmitting any further
        burst packets and retransmission packets.  If support for
        [<a href="./rfc5506" title="&quot;Support for Reduced-Size Real-Time Transport Control Protocol (RTCP): Opportunities and Consequences&quot;">RFC5506</a>] has been signaled, the RTCP BYE message MAY be sent in
        a reduced-size RTCP packet.  Otherwise, <a href="./rfc3550#section-6.1">Section&nbsp;6.1 of [RFC3550]</a>
        mandates the RTCP BYE message always be sent with a sender or
        receiver report in a compound RTCP packet.  If no data has been
        received, an empty receiver report MUST be still included.  With
        the information contained in the receiver report, the RS can
        figure out how many duplicate RTP packets have been delivered to
        the RTP_Rx (note that this will be an upper-bound estimate as
        one or more packets might have been lost during the burst




<span class="grey">Ver Steeg, et al.            Standards Track                   [Page 23]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-24" ></span>
<span class="grey"><a href="./rfc6285">RFC 6285</a>                          RAMS                         June 2011</span>


        transmission).  The impact of duplicate packets and measures
        that can be taken to minimize the impact of receiving duplicate
        packets will be addressed in <a href="#section-6.4">Section 6.4</a>.

        Since an RTCP BYE message issued for the unicast burst RTP
        session terminates that session and ceases transmitting any
        further packets in that session, there is no need for sending
        explicit RAMS-T messages, which would only terminate their
        respective bursts.

   For the purpose of gathering detailed information about RTP_Rx's
   rapid acquisition experience, [<a href="#ref-MULTICAST-ACQ">MULTICAST-ACQ</a>] defines an RTCP
   Extended Report (XR) Block.  This report is designed to be payload-
   independent; thus, it can be used by any multicast application that
   supports rapid acquisition.

<span class="h3"><a class="selflink" id="section-6.3" href="#section-6.3">6.3</a>.  Synchronization of Primary Multicast Streams</span>

   When an RTP_RX acquires multiple primary multicast streams, it might
   need to synchronize them for playout.  This synchronization is
   achieved by the help of the RTCP sender reports [<a href="./rfc3550" title="&quot;RTP: A Transport Protocol for Real-Time Applications&quot;">RFC3550</a>].  If the
   playout will start before the RTP_Rx has joined the multicast
   session, the RTP_Rx needs to receive the information reflecting the
   synchronization among the primary multicast streams early enough so
   that it can play out the media in a synchronized fashion.

   The suggested approach is to use the RTP header extension mechanism
   [<a href="./rfc5285" title="&quot;A General Mechanism for RTP Header Extensions&quot;">RFC5285</a>] and convey the synchronization information in a header
   extension as defined in [<a href="./rfc6051" title="&quot;Rapid Synchronisation of RTP Flows&quot;">RFC6051</a>].  Per [<a href="./rfc4588" title="&quot;RTP Retransmission Payload Format&quot;">RFC4588</a>], "if the original
   RTP header carried an RTP header extension, the retransmission packet
   SHOULD carry the same header extension".  Thus, as long as the
   multicast source emits a header extension with the synchronization
   information frequently enough, there is no additional task that needs
   to be carried out by the BRS.  The synchronization information will
   be sent to the RTP_Rx along with the burst packets.  The frequent
   header extensions sent in the primary multicast RTP sessions also
   allow rapid synchronization of the RTP streams for the RTP receivers
   that do not support RAMS or that directly join the multicast session
   without running RAMS.  Thus, in RAMS applications, it is RECOMMENDED
   that the multicast sources frequently send synchronization
   information by using header extensions following the rules presented
   in [<a href="./rfc6051" title="&quot;Rapid Synchronisation of RTP Flows&quot;">RFC6051</a>].  The regular sender reports are still sent in the
   unicast session by following the rules of [<a href="./rfc3550" title="&quot;RTP: A Transport Protocol for Real-Time Applications&quot;">RFC3550</a>].








<span class="grey">Ver Steeg, et al.            Standards Track                   [Page 24]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-25" ></span>
<span class="grey"><a href="./rfc6285">RFC 6285</a>                          RAMS                         June 2011</span>


<span class="h3"><a class="selflink" id="section-6.4" href="#section-6.4">6.4</a>.  Burst Shaping and Congestion Control in RAMS</span>

   This section provides informative guidelines about how the BRS can
   shape the transmission of the unicast burst and how congestion can be
   dealt with in the RAMS process.  When the RTP_Rx detects that the
   unicast burst is causing severe congestion, it can prefer to send a
   RAMS-T message immediately to stop the unicast burst.

   A higher bitrate for the unicast burst naturally conveys the
   Reference Information and media content to the RTP_Rx faster.  This
   way, the RTP_Rx can start consuming the data sooner, which results in
   a faster acquisition.  A higher bitrate also represents a better
   utilization of the BRS resources.  As the burst may continue until it
   catches up with the primary multicast stream, the higher the bursting
   bitrate, the less data the BRS needs to transmit.  However, a higher
   bitrate for the burst also increases the chances for congestion-
   caused packet loss.  Thus, as discussed in <a href="#section-5">Section 5</a>, there has to be
   an upper bound on the bandwidth used by the burst.

   When the BRS transmits the unicast burst, it is expected to take into
   account all available information to prevent any packet loss that
   might take place during the bursting as a result of buffer overflow
   on the path between the RS and RTP_Rx and at the RTP_Rx itself.  The
   bursting bitrate can be determined by taking into account the
   following information, when available:

   (a)  Information obtained via the RAMS-R message, such as Max RAMS
        Buffer Fill Requirement and/or Max Receive Bitrate (see
        <a href="#section-7.2">Section 7.2</a>).

   (b)  Information obtained via RTCP receiver reports provided by the
        RTP_Rx in the retransmission session, allowing in-session
        bitrate adaptations for the burst.  When these receiver reports
        indicate packet loss, this can indicate a certain congestion
        state in the path from the RS to the RTP_Rx.

   (c)  Information obtained via RTCP NACKs provided by the RTP_Rx in
        the primary multicast RTP session, allowing in-session bitrate
        adaptations for the burst.  Such RTCP NACKs are transmitted by
        the RTP_Rx in response to packet loss detection in the burst.
        NACKs can indicate a certain congestion state on the path from
        the RS to RTP_Rx.

   (d)  There can be other feedback received from the RTP_Rx, e.g., in
        the form of Explicit Congestion Notification - Congestion
        Experienced (ECN-CE) markings [<a href="#ref-ECN-FOR-RTP">ECN-FOR-RTP</a>] that can influence
        in-session bitrate adaptation.




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<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-26" ></span>
<span class="grey"><a href="./rfc6285">RFC 6285</a>                          RAMS                         June 2011</span>


   (e)  Information obtained via updated RAMS-R messages, allowing in-
        session bitrate adaptations, if supported by the BRS.

   (f)  Transport protocol-specific information.  For example, when
        Datagram Congestion Control Protocol (DCCP) is used to transport
        the RTP burst, the ACKs from the DCCP client can be leveraged by
        the BRS / DCCP server for burst shaping and congestion control.

   (g)  Preconfigured settings for each RTP_Rx or a set of RTP_Rxs that
        indicate the upper-bound bursting bitrates for which no packet
        loss will occur as a result of congestion along the path of the
        RS to RTP_Rx.  For example, in managed IPTV networks, where the
        bottleneck bandwidth along the end-to-end path is known and
        where the network between the RS and this link is provisioned
        and dimensioned to carry the burst streams, the bursting bitrate
        does not exceed the provisioned value.  These settings can also
        be dynamically adapted using application-aware knowledge.

   The BRS chooses the initial burst bitrate as follows:

   o  When using RAMS in environments as described in (g), the BRS MUST
      transmit the burst packets at an initial bitrate higher than the
      nominal bitrate but within the engineered or reserved bandwidth
      limit.

   o  When the BRS cannot determine a reliable bitrate value for the
      unicast burst (using (a) through (g)), it is desirable for the BRS
      to choose an appropriate initial bitrate not above the nominal
      bitrate and increase it gradually unless a congestion is detected.

   In both cases, during the burst transmission, the BRS MUST
   continuously monitor for packet losses as a result of congestion by
   means of one or more of the mechanisms described in (b) through (f).
   When the BRS relies on RTCP receiver reports, sufficient bandwidth
   needs to be provided to RTP_Rx for RTCP transmission in the unicast
   burst RTP session.  To achieve a reasonable fast adaptation against
   congestion, it is recommended that the RTP_Rx sends a receiver report
   at least once every two RTTs between the RS and RTP_Rx.  Although the
   specific heuristics and algorithms that deduce a congestion state and
   how the BRS acts subsequently are outside the scope of this
   specification, the following two methods are the best practices:

   o  Upon detection of a significant amount of packet loss, which the
      BRS attributes to congestion, the BRS decreases the burst bitrate.
      The rate by which the BRS increases and decreases the bitrate for
      the burst can be determined by a TCP-friendly bitrate adaptation
      algorithm for RTP over UDP or, in the case of (f), by the
      congestion control algorithms defined in DCCP [<a href="./rfc5762" title="&quot;RTP and the Datagram Congestion Control Protocol (DCCP)&quot;">RFC5762</a>].



<span class="grey">Ver Steeg, et al.            Standards Track                   [Page 26]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-27" ></span>
<span class="grey"><a href="./rfc6285">RFC 6285</a>                          RAMS                         June 2011</span>


   o  If the congestion is persistent and the BRS has to reduce the
      burst bitrate to a point where the RTP_Rx buffer might underrun or
      the burst will consume too many resources, the BRS terminates the
      burst and transmits a RAMS-I message to RTP_Rx with the
      appropriate response code.  It is then up to RTP_Rx to decide when
      to join the multicast session.

   Even though there is no congestion experienced during the burst,
   congestion may anyway arise near the end of the burst as the RTP_Rx
   eventually needs to join the multicast session.  During this brief
   period, both the burst packets and the multicast packets can be
   simultaneously received by the RTP_Rx, thus enhancing the risk of
   congestion.

   Since the BRS signals the RTP_Rx when the BRS expects the RTP_Rx to
   send the SFGMP Join message, the BRS can have a rough estimate of
   when the RTP_Rx will start receiving multicast packets in the SSM
   session.  The BRS can keep on sending burst packets but reduces the
   bitrate accordingly at the appropriate instant by taking the bitrate
   of the whole SSM session into account.  If the BRS ceases
   transmitting the burst packets before the burst catches up, any gap
   resulting from this imperfect switch-over by the RTP_Rx can be later
   repaired by requesting retransmissions for the missing packets from
   the RS.  The retransmissions can be shaped by the BRS to make sure
   that they do not cause collateral loss in the primary multicast RTP
   session and the unicast burst RTP session.

<span class="h3"><a class="selflink" id="section-6.5" href="#section-6.5">6.5</a>.  Failure Cases</span>

   In this section, we examine the implications of losing the RAMS-R,
   RAMS-I, or RAMS-T messages and other failure cases.

   When the RTP_Rx sends a RAMS-R message to initiate a rapid
   acquisition but the message gets lost and the FT does not receive it,
   the RTP_Rx will get neither a RAMS-I message nor a unicast burst.  In
   this case, the RTP_Rx MAY resend the request when it is eligible to
   do so based on the RTCP timer rules defined in [<a href="./rfc4585" title="&quot;Extended RTP Profile for Real-time Transport Control Protocol (RTCP)-Based Feedback (RTP/AVPF)&quot;">RFC4585</a>].  Or, after
   a reasonable amount of time, the RTP_Rx can time out (based on the
   previous observed response times) and immediately join the SSM
   session.

   In the case where RTP_Rx starts receiving a unicast burst but does
   not receive a corresponding RAMS-I message within a reasonable amount
   of time, the RTP_Rx can either discard the burst data or decide not
   to interrupt the unicast burst and be prepared to join the SSM
   session at an appropriate time it determines or as indicated in a
   subsequent RAMS-I message (if available).  If the network is subject




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<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-28" ></span>
<span class="grey"><a href="./rfc6285">RFC 6285</a>                          RAMS                         June 2011</span>


   to packet loss, it is RECOMMENDED that the BRS repeats the RAMS-I
   messages multiple times based on the RTCP timer rules defined in
   [<a href="./rfc4585" title="&quot;Extended RTP Profile for Real-time Transport Control Protocol (RTCP)-Based Feedback (RTP/AVPF)&quot;">RFC4585</a>].

   In the failure cases where the RAMS-I message is lost or the RAMS-R
   message is lost and the RTP_Rx gives up, the RTP_Rx MUST still
   terminate the burst(s) it requested by following the rules described
   in <a href="#section-6.2">Section 6.2</a>.

   In the case where a RAMS-T message sent by the RTP_Rx does not reach
   its destination, the BRS can continue sending burst packets even
   though the RTP_Rx no longer needs them.  If an RTP_Rx is receiving
   burst packets it no longer needs after sending a RAMS-T message, it
   is RECOMMENDED that the RTP_Rx repeats the RAMS-T message multiple
   times based on the RTCP timer rules defined in [<a href="./rfc4585" title="&quot;Extended RTP Profile for Real-time Transport Control Protocol (RTCP)-Based Feedback (RTP/AVPF)&quot;">RFC4585</a>].  The BRS
   MUST be provisioned to terminate the burst when it can no longer send
   the burst packets faster than it receives the primary multicast
   stream packets.

   <a href="./rfc3550#section-6.3.5">Section&nbsp;6.3.5 of [RFC3550]</a> explains the rules pertaining to timing
   out an SSRC.  When the BRS accepts to serve the requested burst(s)
   and establishes the retransmission session, the BRS needs to check
   the liveness of the RTP_Rx via the RTCP messages and reports the
   RTP_Rx sends.  The default rules explained in [<a href="./rfc3550" title="&quot;RTP: A Transport Protocol for Real-Time Applications&quot;">RFC3550</a>] apply in RAMS
   as well.

<span class="h2"><a class="selflink" id="section-7" href="#section-7">7</a>.  Encoding of the Signaling Protocol in RTCP</span>

   This section defines the formats of the RTCP transport-layer feedback
   messages that are exchanged between the retransmission server (RS)
   and RTP receiver (RTP_Rx) during rapid acquisition.  These messages
   are referred to as the RAMS Messages.  They are payload-independent
   and MUST be used by all RTP-based multicast applications that support
   rapid acquisition regardless of the payload they carry.

   Payload-specific feedback messages are not defined in this document.
   However, further optional payload-independent and payload-specific
   information can be included in the exchange.

   The common packet format for the RTCP feedback messages is defined in
   <a href="./rfc4585#section-6.1">Section&nbsp;6.1 of [RFC4585]</a> and is also sketched in Figure 4.










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

     Figure 4: The Common Packet Format for the RTCP Feedback Messages

   Each feedback message has a fixed-length field for version, padding,
   feedback message type (FMT), packet type (PT), length, SSRC of packet
   sender, SSRC of media source, and a variable-length field for
   feedback control information (FCI).

   In the RAMS messages, the PT field is set to RTPFB (205) and the FMT
   field is set to RAMS (6).  Individual RAMS messages are identified by
   a sub-field called sub-feedback message type (SFMT).  Any Reserved
   field SHALL be set to zero and ignored.

   Depending on the specific scenario and timeliness/importance of a
   RAMS message, it can be desirable to send it in a reduced-size RTCP
   packet [<a href="./rfc5506" title="&quot;Support for Reduced-Size Real-Time Transport Control Protocol (RTCP): Opportunities and Consequences&quot;">RFC5506</a>].  However, unless support for [<a href="./rfc5506" title="&quot;Support for Reduced-Size Real-Time Transport Control Protocol (RTCP): Opportunities and Consequences&quot;">RFC5506</a>] has been
   signaled, compound RTCP packets MUST be used by following [<a href="./rfc3550" title="&quot;RTP: A Transport Protocol for Real-Time Applications&quot;">RFC3550</a>]
   rules.

   Following the rules specified in [<a href="./rfc3550" title="&quot;RTP: A Transport Protocol for Real-Time Applications&quot;">RFC3550</a>], all integer fields in the
   messages defined below are carried in network-byte order, that is,
   most significant byte (octet) first, also known as big-endian.
   Unless otherwise stated, numeric constants are in decimal (base 10).

<span class="h3"><a class="selflink" id="section-7.1" href="#section-7.1">7.1</a>.  Extensions</span>

   To improve the functionality of the RAMS method in certain
   applications, it can be desirable to define new fields in the RAMS
   Request, Information, and Termination messages.  Such fields MUST be
   encoded as Type-Length-Value (TLV) elements as described below and
   sketched in Figure 5:

   o  Type:  A single-octet identifier that defines the type of the
      parameter represented in this TLV element.






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   o  Length:  A two-octet field that indicates the length (in octets)
      of the TLV element excluding the Type and Length fields and the
      8-bit Reserved field between them.  This length does not include
      any padding that is required for alignment.

   o  Value:  Variable-size set of octets that contains the specific
      value for the parameter.

   In the extensions, the Reserved field SHALL be set to zero and
   ignored.  If a TLV element does not fall on a 32-bit boundary, the
   last word MUST be padded to the boundary using further bits set to
   zero.

   An RTP_Rx or BRS MAY include vendor-neutral and private extensions in
   any RAMS message.  The RTP_Rx or BRS MUST place such extensions after
   the mandatory fields and mandatory TLV elements (if any) and MAY
   place such extensions in any order.  The RTP_Rx or BRS MUST NOT
   include multiple TLV elements with the same Type value in a RAMS
   message.

   The support for extensions (unless specified otherwise) is OPTIONAL.
   An RTP_Rx or BRS receiving a vendor-neutral or private extension that
   it does not understand MUST ignore that extension.

      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
     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
     |     Type      |   Reserved    |            Length             |
     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
     :                             Value                             :
     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

                   Figure 5: Structure of a TLV Element

<span class="h4"><a class="selflink" id="section-7.1.1" href="#section-7.1.1">7.1.1</a>.  Vendor-Neutral Extensions</span>

   If the goal in defining new TLV elements is to extend the
   functionality in a vendor-neutral manner, they MUST be registered
   with IANA through the guidelines provided in <a href="#section-11.5">Section 11.5</a>.

   The current document defines several vendor-neutral extensions in the
   subsequent sections.

<span class="h4"><a class="selflink" id="section-7.1.2" href="#section-7.1.2">7.1.2</a>.  Private Extensions</span>

   It is desirable to allow vendors to use private extensions in a TLV
   format.  For interoperability, such extensions must not collide with
   each other.



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   A certain range of TLV Types (between and including 128 and 254 ) is
   reserved for private extensions (refer to <a href="#section-11.5">Section 11.5</a>).  IANA
   management for these extensions is unnecessary, and they are the
   responsibility of individual vendors.

   The structure that MUST be used for the private extensions is
   depicted in Figure 6.  Here, the enterprise numbers are as listed on
   <a href="http://www.iana.org">http://www.iana.org</a>.  This will ensure the uniqueness of the private
   extensions and avoid any collision.

      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
     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
     |      Type     |   Reserved    |            Length             |
     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
     |                       Enterprise Number                       |
     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
     :                             Value                             :
     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

                Figure 6: Structure of a Private Extension

<span class="h3"><a class="selflink" id="section-7.2" href="#section-7.2">7.2</a>.  RAMS Request</span>

   The RAMS Request (RAMS-R) message is identified by SFMT=1.  This
   message is sent as unicast feedback in the primary multicast RTP
   session by the RTP_Rx to request rapid acquisition of a primary
   multicast RTP session or one or more primary multicast streams
   belonging to the same primary multicast RTP session.  In the RAMS-R
   message, the RTP_Rx MUST set both the packet sender SSRC and media
   sender SSRC fields to its own SSRC since the media sender SSRC may
   not be known.  The RTP_Rx MUST provide explicit signaling as
   described below to request stream(s).  This minimizes the chances of
   accidentally requesting a wrong primary multicast stream.

   The FCI field MUST contain only one RAMS Request.  The FCI field has
   the structure depicted in Figure 7.

   The semantics of the RAMS-R message is independent of the payload
   type carried in the primary multicast RTP session.











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      0                   1                   2                   3
      0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
     |    SFMT=1     |                    Reserved                   |
     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
     :                  Requested Media Sender SSRC(s)               :
     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
     :      Optional TLV-encoded Fields (and Padding, if needed)     :
     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

          Figure 7: FCI Field Syntax for the RAMS Request Message

   o  Requested Media Sender SSRC(s):  Mandatory TLV element that lists
      the media sender SSRC(s) requested by the RTP_Rx.  The BRS MUST
      ignore the media sender SSRC specified in the header of the RAMS-R
      message.

      If the Length field is set to zero (i.e., no media sender SSRC is
      listed), it means that the RTP_Rx is requesting to rapidly acquire
      the entire primary multicast RTP session.  Otherwise, the RTP_Rx
      lists the individual media sender SSRCs in this TLV element and
      sets the Length field of the TLV element to 4*n, where n is the
      number of SSRC entries.

      Type:  1

   o  Min RAMS Buffer Fill Requirement (32 bits):  Optional TLV element
      that denotes the minimum milliseconds of data that the RTP_Rx
      desires to have in its buffer before allowing the data to be
      consumed by the application.

      The RTP_Rx can have knowledge of its buffering requirements.
      These requirements can be application and/or device specific.  For
      instance, the RTP_Rx might need to have a certain amount of data
      in its application buffer to handle transmission jitter and/or to
      be able to support error-control methods.  If the BRS is told the
      minimum buffering requirement of the receiver, the BRS can tailor
      the burst(s) more precisely, e.g., by choosing an appropriate
      starting point.  The methods used by the RTP_Rx to determine this
      value are application specific and thus out of the scope of this
      document.










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      If specified, the amount of backfill that will be provided by the
      unicast bursts and any payload-specific information MUST NOT be
      smaller than the specified value.  Otherwise, the backfill will
      not be able to build up the desired level of buffer at the RTP_Rx
      and can cause buffer underruns.

      Type:  2

   o  Max RAMS Buffer Fill Requirement (32 bits):  Optional TLV element
      that denotes the maximum milliseconds of data that the RTP_Rx can
      buffer without losing the data due to buffer overflow.

      The RTP_Rx can have knowledge of its buffering requirements.
      These requirements can be application or device specific.  For
      instance, one particular RTP_Rx might have more physical memory
      than another RTP_Rx and thus can buffer more data.  If the BRS
      knows the buffering ability of the receiver, the BRS can tailor
      the burst(s) more precisely.  The methods used by the receiver to
      determine this value are application specific and thus out of the
      scope of this document.

      If specified, the amount of backfill that will be provided by the
      unicast bursts and any payload-specific information MUST NOT be
      larger than this value.  Otherwise, the backfill may cause buffer
      overflows at the RTP_Rx.

      Type:  3

   o  Max Receive Bitrate (64 bits):  Optional TLV element that denotes
      the maximum bitrate (in bits per second) at which the RTP_Rx can
      process the aggregation of the unicast burst(s) and any payload-
      specific information that will be provided by the BRS.  The limits
      can include local receiver limits as well as network limits that
      are known to the receiver.

      If specified, the total bitrate of the unicast burst(s) plus any
      payload-specific information MUST NOT be larger than this value.
      Otherwise, congestion and packet loss are more likely to occur.

      Type:  4

   o  Preamble-only Allowed (0 bits):  Optional TLV element that
      indicates that the RTP_Rx is willing to receive only the preamble
      information for the desired primary multicast stream(s) in case
      the BRS cannot send the unicast burst stream(s).  (In such cases,
      the BRS will not accept the request, but will send a response code
      511 in the RAMS-I message as defined in <a href="#section-11.6">Section 11.6</a>.)  Note that




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      the RTP_Rx signals the particular preamble format(s) it supports
      through a public or a private extension in the same RAMS-R
      message.

      If this TLV element does not exist in the RAMS-R message, the BRS
      MUST NOT respond to the RAMS-R message by sending the preamble
      information only.  Since this TLV element does not carry a Value
      field, the Length field MUST be set to zero.

      Type:  5

   o  Supported Enterprise Number(s):  Optional TLV element that
      indicates the enterprise number(s) that the RTP_Rx implementation
      supports.  Similar to the private extensions, the enterprise
      numbers here are as listed on <a href="http://www.iana.org">http://www.iana.org</a>.  This TLV
      element, if exists, provides a hint to the BRS about which private
      extensions the RTP_Rx can potentially support.  Note that an
      RTP_Rx does not necessarily support all the private extensions
      under a particular enterprise number.  Unless the BRS explicitly
      knows which private extensions an RTP_Rx supports (through this or
      some out-of-band means), the BRS SHOULD NOT use private extensions
      in the RAMS-I messages.  The BRS SHOULD rather use only vendor-
      neutral extensions.  The Length field of this TLV element is set
      to 4*n, where n is the number of enterprise number entries.

      Type:  6

<span class="h3"><a class="selflink" id="section-7.3" href="#section-7.3">7.3</a>.  RAMS Information</span>

   The RAMS Information (RAMS-I) message is identified by SFMT=2.  This
   message is used to describe the unicast burst that will be sent for
   rapid acquisition.  It also includes other useful information for the
   RTP_Rx as described below.

   A separate RAMS-I message with the appropriate response code is sent
   in the unicast burst RTP session by the BRS for each primary
   multicast stream that has been requested by the RTP_Rx.  In each of
   these RAMS-I messages, the media sender SSRC and packet sender SSRC
   fields are both set to the SSRC of the BRS, which equals the SSRC of
   the respective primary multicast stream.

   The FCI field MUST contain only one RAMS Information message.  The
   FCI field has the structure depicted in Figure 8.

   The semantics of the RAMS-I message is independent of the payload
   type carried in the primary multicast RTP session.





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      0                   1                   2                   3
      0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
     |    SFMT=2     |      MSN      |          Response             |
     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
     :      Optional TLV-encoded Fields (and Padding, if needed)     :
     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

        Figure 8: FCI Field Syntax for the RAMS Information Message

   A RAMS-I message has the following fields:

   o  Message Sequence Number (MSN) (8 bits) :  Mandatory field that
      denotes the sequence number of the RAMS-I message for the
      particular media sender SSRC specified in the message header.  The
      MSN value SHALL be set to zero when a new RAMS request is
      received.  During rapid acquisition, the same RAMS-I message MAY
      be repeated for redundancy purposes without incrementing the MSN
      value.  If an updated RAMS-I message will be sent (either with new
      information or updated information), the MSN value SHALL be
      incremented by one.  In the MSN field, the regular wrapping rules
      apply.  Note that if the RTP_Rx has sent an updated request, it
      MUST check every RAMS-I message entirely, regardless of whether or
      not the MSN value has changed.

   o  Response (16 bits):  Mandatory field that denotes the response
      code for this RAMS-I message.  This document defines several
      initial response codes in <a href="#section-7.3.1">Section 7.3.1</a> and registers them with
      IANA in <a href="#section-11.6">Section 11.6</a>.  If a new vendor-neutral response code will
      be defined, it MUST be registered with IANA through the guidelines
      specified in <a href="#section-11.6">Section 11.6</a>.  If the new response code is intended
      to be used privately by a vendor, there is no need for IANA
      management.  Instead, the vendor MUST use the private extension
      mechanism (<a href="#section-7.1.2">Section 7.1.2</a>) to convey its message and MUST indicate
      this by putting zero in the Response field.

      When the RTP_Rx receives a RAMS-I message with a response code
      that it does not understand, the RTP_Rx MUST send a RAMS-T message
      immediately to the BRS.

   The following TLV elements have been defined for the RAMS-I messages:

   o  Media Sender SSRC (32 bits):  Optional TLV element that specifies
      the media sender SSRC of the unicast burst stream.  If the FT_Ap
      that received the RAMS-R message is associated with a single
      primary multicast stream but the requested media sender SSRC does
      not match the SSRC of the RTP stream associated with this FT_Ap,




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      the BRS includes this TLV element in the initial RAMS-I message to
      let the RTP_Rx know that the media sender SSRC has changed.  If
      the two SSRCs match, there is no need to include this TLV element.

      Type:  31

   o  RTP Seqnum of the First Packet (16 bits):  TLV element that
      specifies the RTP sequence number of the first packet that will be
      sent in the respective unicast RTP stream.  This allows the RTP_Rx
      to know whether one or more packets sent by the BRS have been
      dropped at the beginning of the stream.  If the BRS accepts the
      RAMS request, this element exists.  If the BRS rejects the RAMS
      request, this element SHALL NOT exist.

      Type:  32

   o  Earliest Multicast Join Time (32 bits):  TLV element that
      specifies the delta time (in ms) between the arrival of the first
      RTP packet in the unicast RTP stream (which could be a burst
      packet or a payload-specific packet) and the earliest time instant
      when an RTP_Rx MAY send an SFGMP Join message to join the
      multicast session.  A zero value indicated in this element means
      that the RTP_Rx MAY send the SFGMP Join message right away.  If
      the RTP_Rx does not want to wait until the earliest multicast join
      time, it MAY send a RAMS-T message, and after a reasonable amount
      of time, it MAY join the SSM session.

      If the RAMS request has been accepted, the BRS sends this element
      at least once so that the RTP_Rx knows when to join the multicast
      session.  If the burst request has been rejected as indicated in
      the Response field, this element MUST indicate a zero value.  In
      that case, it is up to the RTP_Rx when or whether to join the
      multicast session.

      When the BRS serves two or more bursts and sends a separate RAMS-I
      message for each burst, the join times specified in these RAMS-I
      messages SHOULD correspond to more or less the same time instant,
      and the RTP_Rx sends the SFGMP Join message based on the earliest
      join time.

      Type:  33

   o  Burst Duration (32 bits):  Optional TLV element that denotes the
      time the burst will last (in ms), i.e., the difference between the
      expected transmission times of the first and the last burst
      packets that the BRS is planning to send in the respective unicast
      RTP stream.  In the absence of additional stimulus, the BRS will




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      send a burst of this duration.  However, the burst duration can be
      modified by subsequent events, including changes in the primary
      multicast stream and reception of RAMS-T messages.

      The BRS MUST terminate the flow in the timeframe indicated by this
      burst duration or the duration established by those subsequent
      events even if it does not get a RAMS-T or a BYE message from the
      RTP_Rx.  It is OPTIONAL to send this element in a RAMS-I message
      when the burst request is accepted.  If the burst request has been
      rejected as indicated in the Response field, this element MAY be
      omitted or indicate a zero value.

      Type:  34

   o  Max Transmit Bitrate (64 bits):  Optional TLV element that denotes
      the maximum bitrate (in bits per second) that will be used by the
      BRS for the RTP stream associated with this RAMS-I message.

      Type:  35

<span class="h4"><a class="selflink" id="section-7.3.1" href="#section-7.3.1">7.3.1</a>.  Response Code Definitions</span>

   1xx-Level Response Codes:  These response codes are sent for
   informational purposes.

   o  100:  This is used when the BRS wants to update a value that was
      sent earlier to the RTP_Rx.

   2xx-Level Response Codes:  These response codes are sent to indicate
   success.

   o  200:  This is used when the server accepts the RAMS request.

   o  201:  This is used when the unicast burst has been completed and
      the BRS wants to notify the RTP_Rx.

   4xx-Level Response Codes:  These response codes are sent to indicate
   that the message sent by the RTP_Rx is either improperly formatted or
   contains an invalid value.  The rules the RTP_Rx needs to follow upon
   receiving one of these response codes are outlined in Step 5 in
   <a href="#section-6.2">Section 6.2</a>.  The 4xx-level response codes are also used as status
   codes in the Multicast Acquisition Report Block [<a href="#ref-MULTICAST-ACQ">MULTICAST-ACQ</a>] in
   order to collect RTP_Rx-based error conditions.

   o  400:  This is used when the RAMS-R message is improperly
      formatted.





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   o  401:  This is used when the minimum RAMS buffer fill requirement
      value indicated in the RAMS-R message is invalid.

   o  402:  This is used when the maximum RAMS buffer fill requirement
      value indicated in the RAMS-R message is invalid.

   o  403:  This is used when the maximum receive bitrate value
      indicated in the RAMS-R message is insufficient.

   o  404:  This is used when the RAMS-T message is improperly
      formatted.

   5xx-Level Response Codes:  These response codes are sent to indicate
   an error on the BRS side.  The rules the RTP_Rx needs to follow upon
   receiving one of these response codes are outlined in Step 3 in
   <a href="#section-6.2">Section 6.2</a>.  The 5xx-level response codes are also used as status
   codes in the Multicast Acquisition Report Block [<a href="#ref-MULTICAST-ACQ">MULTICAST-ACQ</a>] in
   order to collect BRS-based error conditions.

   o  500:  This is used when the BRS has experienced an internal error
      and cannot accept the RAMS request.

   o  501:  This is used when the BRS does not have enough bandwidth to
      send the unicast burst stream.

   o  502:  This is used when the BRS terminates the unicast burst
      stream due to network congestion.

   o  503:  This is used when the BRS does not have enough CPU resources
      to send the unicast burst stream.

   o  504:  This is used when the BRS does not support sending a unicast
      burst stream.

   o  505:  This is used when the requesting RTP_Rx is not eligible to
      receive a unicast burst stream.

   o  506:  This is used when RAMS functionality is not enabled for the
      requested multicast stream.

   o  507:  This is used when the BRS cannot find a valid starting point
      for the unicast burst stream that satisfies the RTP_Rx's
      requirements.

   o  508:  This is used when the BRS cannot find the essential
      reference information for the requested multicast stream.





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<span class="grey"><a href="./rfc6285">RFC 6285</a>                          RAMS                         June 2011</span>


   o  509:  This is used when the BRS cannot match the requested SSRC to
      any of the streams it is serving.

   o  510:  This is used when the BRS cannot serve the requested entire
      session.

   o  511:  This is used when the BRS sends only the preamble
      information but not the whole unicast burst stream.

   o  512:  This is used when the RAMS request is denied due to a policy
      for the requested multicast stream, the RTP_Rx, or this particular
      BRS.

<span class="h3"><a class="selflink" id="section-7.4" href="#section-7.4">7.4</a>.  RAMS Termination</span>

   The RAMS Termination (RAMS-T) message is identified by SFMT=3.

   The RAMS Termination is used to assist the BRS in determining when to
   stop the burst.  A separate RAMS-T message is sent in the unicast
   burst RTP session by the RTP_Rx for each primary multicast stream
   that has been served by the BRS.  Each of these RAMS-T message's
   media sender SSRC field SHALL BE populated with the SSRC of the media
   stream to be terminated.  If the media sender SSRC populated in the
   RAMS-T message does not match the SSRC of the burst served by the
   BRS, BRS SHALL ignore the RAMS-T message.

   If the RTP_Rx wants the BRS to stop a burst prematurely, it sends a
   RAMS-T message as described below.  Upon receiving this message, the
   BRS stops the respective burst immediately.  If the RTP_Rx wants the
   BRS to terminate all of the bursts, it needs to send all of the
   respective RAMS-T messages in a single compound RTCP packet.

   The default behavior for the RTP_Rx is to send a RAMS-T message
   immediately after it joined the multicast session and started
   receiving multicast packets.  In that case, the RTP_Rx includes the
   sequence number of the first RTP packet received in the primary
   multicast stream in the RAMS-T message.  With this information, the
   BRS can decide when to terminate the unicast burst.

   The FCI field MUST contain only one RAMS Termination.  The FCI field
   has the structure depicted in Figure 9.

   The semantics of the RAMS-T message is independent of the payload
   type carried in the primary multicast RTP session.







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<span class="grey"><a href="./rfc6285">RFC 6285</a>                          RAMS                         June 2011</span>


      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
     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
     |    SFMT=3     |                    Reserved                   |
     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
     :      Optional TLV-encoded Fields (and Padding, if needed)     :
     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

        Figure 9: FCI field syntax for the RAMS Termination message

   o  Extended RTP Seqnum of First Multicast Packet (32 bits):  Optional
      TLV element that specifies the extended RTP sequence number of the
      first packet received from the SSM session for a particular
      primary multicast stream.  The low 16 bits contain the sequence
      number of the first packet received from the SSM session, and the
      most significant 16 bits extend that sequence number with the
      corresponding count of sequence number cycles, which can be
      maintained according to the algorithm in <a href="./rfc3550#appendix-A.1">Appendix&nbsp;A.1 of
      [RFC3550]</a>.

      Type:  61

<span class="h2"><a class="selflink" id="section-8" href="#section-8">8</a>.  SDP Signaling</span>

<span class="h3"><a class="selflink" id="section-8.1" href="#section-8.1">8.1</a>.  Definitions</span>

   The syntax of the 'rtcp-fb' attribute has been defined in [<a href="./rfc4585" title="&quot;Extended RTP Profile for Real-time Transport Control Protocol (RTCP)-Based Feedback (RTP/AVPF)&quot;">RFC4585</a>].
   Here we add the following syntax to the 'rtcp-fb' attribute (the
   feedback type and optional parameters are all case sensitive):

   (In the following ABNF [<a href="./rfc5234" title="&quot;Augmented BNF for Syntax Specifications: ABNF&quot;">RFC5234</a>], rtcp-fb-nack-param is used as
   defined in [<a href="./rfc4585" title="&quot;Extended RTP Profile for Real-time Transport Control Protocol (RTCP)-Based Feedback (RTP/AVPF)&quot;">RFC4585</a>].)

         rtcp-fb-nack-param =/ SP "rai"

   The following parameter is defined in this document for use with
   'nack':

   o  'rai' stands for Rapid Acquisition Indication and indicates the
      use of RAMS messages as defined in <a href="#section-7">Section 7</a>.

   This document also defines a new media-level SDP attribute
   ('rams-updates') that indicates whether or not the BRS supports
   updated request messages.  This attribute is used in a declarative
   manner and no Offer/Answer Model behavior is specified.  If the BRS
   supports updated request messages and this attribute is included in
   the SDP description, the RTP_Rx can send updated requests.  The BRS
   might or might not be able to accept value changes in every field in



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<span class="grey"><a href="./rfc6285">RFC 6285</a>                          RAMS                         June 2011</span>


   an updated RAMS-R message.  However, if the 'rams-updates' attribute
   is not included in the SDP description, the RTP_Rx SHALL NOT send
   updated requests.  The RTP_Rx MAY still repeat its initial request
   without changes, though.

<span class="h3"><a class="selflink" id="section-8.2" href="#section-8.2">8.2</a>.  Requirements</span>

   The use of SDP to describe the RAMS entities normatively requires
   support for:

   o  The SDP grouping framework and flow identification (FID) semantics
      [<a href="./rfc5888" title="&quot;The Session Description Protocol (SDP) Grouping Framework&quot;">RFC5888</a>]

   o  The RTP/AVPF profile [<a href="./rfc4585" title="&quot;Extended RTP Profile for Real-time Transport Control Protocol (RTCP)-Based Feedback (RTP/AVPF)&quot;">RFC4585</a>]

   o  The RTP retransmission payload format [<a href="./rfc4588" title="&quot;RTP Retransmission Payload Format&quot;">RFC4588</a>]

   o  The RTCP extensions for SSM sessions with unicast feedback
      [<a href="./rfc5760" title="&quot;RTP Control Protocol (RTCP) Extensions for Single-Source Multicast Sessions with Unicast Feedback&quot;">RFC5760</a>]

   o  The 'multicast-rtcp' attribute [<a href="./rfc6128" title="&quot;RTP Control Protocol (RTCP) Port for Source- Specific Multicast (SSM) Sessions&quot;">RFC6128</a>]

   o  Multiplexing RTP and RTCP on a single port on both endpoints in
      the unicast session [<a href="./rfc5761" title="&quot;Multiplexing RTP Data and Control Packets on a Single Port&quot;">RFC5761</a>]

   o  The 'portmapping-req' attribute [<a href="./rfc6284" title="&quot;Port Mapping Between Unicast and Multicast RTP Sessions&quot;">RFC6284</a>]

   Support for the source-specific media attributes [<a href="./rfc5576" title="&quot;Source-Specific Media Attributes in the Session Description Protocol (SDP)&quot;">RFC5576</a>] can also
   be needed when the 'ssrc' attribute is to be used for the media
   descriptions.

<span class="h3"><a class="selflink" id="section-8.3" href="#section-8.3">8.3</a>.  Example and Discussion</span>

   This section provides a declarative SDP [<a href="./rfc4566" title="&quot;SDP: Session Description Protocol&quot;">RFC4566</a>] example (for the
   RTP_Rx side) for enabling rapid acquisition of multicast RTP
   sessions.

        v=0
        o=ali 1122334455 1122334466 IN IP4 rams.example.com
        s=Rapid Acquisition Example
        t=0 0
        a=group:FID 1 2
        a=rtcp-unicast:rsi
        m=video 41000 RTP/AVPF 98
        i=Primary Multicast Stream
        c=IN IP4 233.252.0.2/255
        a=source-filter:incl IN IP4 233.252.0.2 198.51.100.1
        a=rtpmap:98 MP2T/90000



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<span class="grey"><a href="./rfc6285">RFC 6285</a>                          RAMS                         June 2011</span>


        a=multicast-rtcp:42000
        a=rtcp:43000 IN IP4 192.0.2.1
        a=rtcp-fb:98 nack
        a=rtcp-fb:98 nack rai
        a=ssrc:123321 cname:iptv-ch32@rams.example.com
        a=rams-updates
        a=portmapping-req:54000 IN IP4 192.0.2.1
        a=mid:1
        m=video 51000 RTP/AVPF 99
        i=Unicast Retransmission Stream (Ret. and Rapid Acq. Support)
        c=IN IP4 192.0.2.1
        a=sendonly
        a=rtpmap:99 rtx/90000
        a=rtcp-mux
        a=rtcp:51500
        a=fmtp:99 apt=98;rtx-time=5000
        a=portmapping-req:55000
        a=mid:2

         Figure 10: Example SDP for a Single-Channel RAMS Scenario

   In this example SDP description, we have a primary multicast (source)
   stream and a unicast retransmission stream.  The source stream is
   multicast from a distribution source (with a source IP address of
   198.51.100.1) to the multicast destination address of 233.252.0.2 and
   port 41000.  The corresponding RTCP traffic is multicast to the same
   multicast destination address at port 42000.  Here, we are assuming
   that the multicast RTP and RTCP ports are carefully chosen so that
   different RTP and RTCP streams do not collide with each other.

   The feedback target (FT_Ap) residing on the RS (with an address of
   192.0.2.1) at port 43000 is declared with the "a=rtcp" line
   [<a href="./rfc3605" title="&quot;Real Time Control Protocol (RTCP) attribute in Session Description Protocol (SDP)&quot;">RFC3605</a>].  The support for the conventional retransmission is
   indicated through the "a=rtcp-fb:98 nack" line.  The RTP receiver(s)
   can report missing packets on the source stream to the feedback
   target and request retransmissions.  The SDP above includes the
   "a=sendonly" line for the media description of the retransmission
   stream since the retransmissions are sent in only one direction.

   The support for rapid acquisition is indicated through the "a=rtcp-
   fb:98 nack rai" line.  The parameter 'rtx-time' applies to both the
   conventional retransmissions and rapid acquisition.  However, when
   rapid acquisition is enabled, the standard definition for the
   parameter 'rtx-time' given in [<a href="./rfc4588" title="&quot;RTP Retransmission Payload Format&quot;">RFC4588</a>] is not followed.  Instead,
   when rapid acquisition support is enabled, 'rtx-time' specifies the
   time in milliseconds that the BRS keeps an RTP packet in its cache





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<span class="grey"><a href="./rfc6285">RFC 6285</a>                          RAMS                         June 2011</span>


   available for retransmission (measured from the time the packet was
   received by the BRS, not from the time indicated in the packet
   timestamp).

   Once an RTP_Rx has acquired an SDP description, it can ask for rapid
   acquisition before it joins a primary multicast RTP session.  To do
   so, it sends a RAMS-R message to the feedback target of that primary
   multicast RTP session.  If the FT_Ap is associated with only one RTP
   stream, the RTP_Rx does not need to learn the SSRC of that stream via
   an out-of-band method.  If the BRS accepts the rapid acquisition
   request, it will send a RAMS-I message with the correct SSRC
   identifier.  If the FT_Ap is associated with a multi-stream RTP
   session and the RTP_Rx is willing to request rapid acquisition for
   the entire session, the RTP_Rx again does not need to learn the SSRCs
   via an out-of-band method.  However, if the RTP_Rx intends to request
   a particular subset of the primary multicast streams, it must learn
   their SSRC identifiers and list them in the RAMS-R message.  Since
   this RTP_Rx has not yet received any RTP packets for the primary
   multicast stream(s), the RTP_Rx must in this case learn the SSRC
   value(s) from the 'ssrc' attribute of the media description
   [<a href="./rfc5576" title="&quot;Source-Specific Media Attributes in the Session Description Protocol (SDP)&quot;">RFC5576</a>].  In addition to the SSRC value, the 'cname' source
   attribute must also be present in the SDP description [<a href="./rfc5576" title="&quot;Source-Specific Media Attributes in the Session Description Protocol (SDP)&quot;">RFC5576</a>].

   Listing the SSRC values for the primary multicast streams in the SDP
   file does not create a problem in SSM sessions when an SSRC collision
   occurs.  This is because in SSM sessions, an RTP_Rx that observed an
   SSRC collision with a media sender must change its own SSRC [<a href="./rfc5760" title="&quot;RTP Control Protocol (RTCP) Extensions for Single-Source Multicast Sessions with Unicast Feedback&quot;">RFC5760</a>]
   by following the rules defined in [<a href="./rfc3550" title="&quot;RTP: A Transport Protocol for Real-Time Applications&quot;">RFC3550</a>].

   A feedback target that receives a RAMS-R message becomes aware that
   the RTP_Rx wants to rapidly catch up with a primary multicast RTP
   session.  If the necessary conditions are satisfied (as outlined in
   <a href="./rfc4585#section-7">Section&nbsp;7 of [RFC4585]</a>) and available resources exist, the BRS can
   react to the RAMS-R message by sending any transport-layer (and
   optional payload-specific, when allowed) feedback message(s) and
   starting the unicast burst.

   In this section, we considered the simplest scenario where the
   primary multicast RTP session carried only one stream and the RTP_Rx
   wanted to rapidly acquire this stream only.  Best practices for
   scenarios where the primary multicast RTP session carries two or more
   streams or the RTP_Rx wants to acquire one or more streams from
   multiple primary multicast RTP sessions at the same time are
   presented in [<a href="#ref-RAMS-SCENARIOS">RAMS-SCENARIOS</a>].







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<span class="grey"><a href="./rfc6285">RFC 6285</a>                          RAMS                         June 2011</span>


<span class="h2"><a class="selflink" id="section-9" href="#section-9">9</a>.  NAT Considerations</span>

   For a variety of reasons, one or more Network Address Port
   Translators (NAPT, hereafter simply called NAT) can exist between the
   RTP_Rx and RS.  NATs have a variety of operating characteristics for
   UDP traffic [<a href="./rfc4787" title="&quot;Network Address Translation (NAT) Behavioral Requirements for Unicast UDP&quot;">RFC4787</a>].  For a NAT to permit traffic from the BRS to
   arrive at the RTP_Rx, the NAT(s) must first either:

   a.  See UDP (or DCCP) traffic sent from the RTP_Rx (which is on the
       'inside' of the NAT) to the BRS (which is on the 'outside' of the
       NAT).  This traffic has the same transport address as the
       subsequent response traffic, or

   b.  Be configured to forward certain ports (e.g., using HTML
       configuration or Universal Plug and Play (UPnP) Internet Gateway
       Device (IGD) [<a href="#ref-UPnP-IGD" title="&quot;Universal Plug and Play (UPnP) Internet Gateway Device (IGD)&quot;">UPnP-IGD</a>]).  Details of this are out of the scope
       of this document.

   For both (a) and (b), the RTP_Rx is responsible for maintaining the
   NAT's state if it wants to receive traffic from the BRS on that port.
   For (a), the RTP_Rx MUST send UDP traffic to keep the NAT binding
   alive, at least every 30 seconds [<a href="./rfc4787" title="&quot;Network Address Translation (NAT) Behavioral Requirements for Unicast UDP&quot;">RFC4787</a>].  While (a) is more like
   an automatic/dynamic configuration, (b) is more like a manual/static
   configuration.

   When the RTP_Rx sends a request (RAMS-R) message to the FT as unicast
   feedback in the primary multicast RTP session and the request is
   received by the FT, a new unicast burst RTP session will be
   established between the BRS and RTP_Rx.

   While the FT and BRS ports on the RS are already signaled via out-of-
   band means (e.g., SDP), the RTP_Rx needs to convey the RTP and RTCP
   ports it wants to use on its side for the new session.  Since there
   are two RTP sessions (one multicast and one unicast) involved during
   this process and one of them is established upon a feedback message
   sent in the other one, this requires an explicit port mapping method.

   Applications using RAMS MUST support the method presented in
   [<a href="./rfc6284" title="&quot;Port Mapping Between Unicast and Multicast RTP Sessions&quot;">RFC6284</a>] both on the RS and RTP_Rx side to allow RTP receivers to
   use their desired ports and to support RAMS behind NAT devices.  The
   port mapping message exchange needs to take place whenever the RTP_Rx
   intends to make use of the RAMS protocol for rapidly acquiring a
   specific multicast RTP session prior to joining the associated SSM
   session.







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<span class="grey"><a href="./rfc6285">RFC 6285</a>                          RAMS                         June 2011</span>


<span class="h2"><a class="selflink" id="section-10" href="#section-10">10</a>.  Security Considerations</span>

   Applications that are using RAMS make heavy use of the unicast
   feedback mechanism described in [<a href="./rfc5760" title="&quot;RTP Control Protocol (RTCP) Extensions for Single-Source Multicast Sessions with Unicast Feedback&quot;">RFC5760</a>], the payload format defined
   in [<a href="./rfc4588" title="&quot;RTP Retransmission Payload Format&quot;">RFC4588</a>], and the port mapping solution specified in [<a href="./rfc6284" title="&quot;Port Mapping Between Unicast and Multicast RTP Sessions&quot;">RFC6284</a>].
   Thus, these applications are subject to the general security
   considerations discussed in those documents.  In particular, the
   threats and risks discussed in [<a href="./rfc5760" title="&quot;RTP Control Protocol (RTCP) Extensions for Single-Source Multicast Sessions with Unicast Feedback&quot;">RFC5760</a>] need to be considered
   carefully.  In this section, we give an overview of the guidelines
   and suggestions described in these specifications from a RAMS
   perspective.  We also discuss the security considerations that
   explicitly apply to applications using RAMS.

   First of all, much of the session description information is
   available in the SDP descriptions that are distributed to the media
   senders, retransmission servers, and RTP receivers.  Adequate
   security measures are RECOMMENDED to ensure the integrity and
   authenticity of the SDP descriptions so that transport addresses of
   the media senders, distribution sources, feedback targets, and other
   session-specific information can be protected.  See [<a href="./rfc4566" title="&quot;SDP: Session Description Protocol&quot;">RFC4566</a>] for
   details.

   Compared to an RTCP NACK message that triggers one or more
   retransmissions, a RAMS Request (RAMS-R) message can trigger a new
   burst stream to be sent by the retransmission server.  Depending on
   the application-specific requirements and conditions existing at the
   time of the RAMS-R reception by the retransmission server, the
   resulting burst stream can potentially contain a large number of
   retransmission packets.  Since these packets are sent faster than the
   nominal rate, RAMS consumes more resources on the retransmission
   servers, RTP receivers, and the network.  In particular, this can
   make denial-of-service (DoS) attacks more intense and hence more
   harmful than attacks that target ordinary retransmission sessions.

   As RAMS messages are sent as RTCP messages, counter-measures SHOULD
   be taken to prevent tampered or spoofed RTCP packets, following the
   suggestions given in [<a href="./rfc4588" title="&quot;RTP Retransmission Payload Format&quot;">RFC4588</a>].  Tampered RAMS-R messages can trigger
   inappropriate burst streams or alter the existing burst streams in an
   inappropriate way.  For example, if the Max Receive Bitrate field is
   altered by a tampered RAMS-R message, the updated burst can overflow
   the buffer at the receiver side or, oppositely, can slow down the
   burst to the point that it becomes useless.  Tampered RAMS
   Termination (RAMS-T) messages can terminate valid burst streams
   prematurely resulting in gaps in the received RTP packets.  RAMS
   Information (RAMS-I) messages contain fields that are critical for a
   successful rapid acquisition.  Any tampered information in the RAMS-I
   message can easily cause an RTP receiver to make wrong decisions.
   Consequently, the RAMS operation can fail.



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<span class="grey"><a href="./rfc6285">RFC 6285</a>                          RAMS                         June 2011</span>


   RTCP BYE messages are similar to RAMS-T messages in the sense that
   they can be used to stop an existing burst.  The CNAME of an RTP
   receiver is used to bind the RTCP BYE message to an existing burst.
   Thus, one should be careful if the CNAMEs are reasonably easy to
   guess and off-path attacks can be performed.  Also note that the
   CNAMEs might be redistributed to all participants in the multicast
   group (as in ASM or the simple feedback model of [<a href="./rfc5760" title="&quot;RTP Control Protocol (RTCP) Extensions for Single-Source Multicast Sessions with Unicast Feedback&quot;">RFC5760</a>]).

   The retransmission server has to consider if values indicated in a
   RAMS-R message are reasonable.  For example, a request demanding a
   large value of many seconds in the Min RAMS Buffer Fill Requirement
   element should, unless special use cases exist, likely be rejected
   since it is likely to be an attempt to prolong a DoS attack on the
   retransmission server, RTP receiver, and/or the network.  The Max
   Receive Bitrate could also be set to a very large value to try to get
   the retransmission server to cause massive congestion by bursting at
   a bitrate that will not be supported by the network.  An RTP_Rx
   should also consider if the values for the Earliest Multicast Join
   Time and Burst Duration indicated by the retransmission server in a
   RAMS-I message are reasonable.  For example, if the burst packets
   stop arriving and the join time is still significantly far into the
   future, this could be a sign of a man-in-the-middle attack where the
   RAMS-I message has been manipulated by an attacker.

   A basic mitigation against DoS attacks introduced by an attacker
   injecting tampered RAMS messages is source address validation
   [<a href="./rfc2827" title="&quot;Network Ingress Filtering: Defeating Denial of Service Attacks which employ IP Source Address Spoofing&quot;">RFC2827</a>].  Also, most of the DoS attacks can be prevented by the
   integrity and authenticity checks enabled by Secure RTP (SRTP)
   [<a href="./rfc3711" title="&quot;The Secure Real-time Transport Protocol (SRTP)&quot;">RFC3711</a>].  However, an attack can still be started by legitimate
   endpoints that send several valid RAMS-R messages to a particular
   feedback target in a synchronized fashion and in a very short amount
   of time.  Since a RAMS operation can temporarily consume a large
   amount of resources, a series of the RAMS-R messages can temporarily
   overload the retransmission server.  In these circumstances, the
   retransmission server can, for example, reject incoming RAMS requests
   until its resources become available again.  One means to ameliorate
   this threat is to apply a per-endpoint policing mechanism on the
   incoming RAMS requests.  A reasonable policing mechanism should
   consider application-specific requirements and minimize false
   negatives.

   In addition to the DoS attacks, man-in-the-middle and replay attacks
   will also be harmful.  RAMS-R messages do not carry any information
   that allows the retransmission server to detect duplication or replay
   attacks.  Thus, the possibility of a replay attack using a captured
   valid RAMS-R message exists unless a mitigation method such as Secure
   RTCP (SRTCP) is used.  Similarly, RAMS-T messages can be replayed.
   The RAMS-I has a sequence number that makes replay attacks less



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<span class="grey"><a href="./rfc6285">RFC 6285</a>                          RAMS                         June 2011</span>


   likely but not impossible.  Man-in-the-middle attacks that are
   capable of capturing, injecting, or modifying the RAMS messages can
   easily accomplish the attacks described above.  Thus, cryptographic
   integrity and authentication is the only reliable protection.  To
   protect the RTCP messages from man-in-the-middle and replay attacks,
   the RTP receivers and retransmission server SHOULD perform a Datagram
   Transport Layer Security (DTLS)-SRTP handshake [<a href="./rfc5764" title="&quot;Datagram Transport Layer Security (DTLS) Extension to Establish Keys for the Secure Real-time Transport Protocol (SRTP)&quot;">RFC5764</a>] over the
   RTCP channel.  Because there is no integrity-protected signaling
   channel between an RTP receiver and the retransmission server, the
   retransmission server MUST maintain a list of certificates owned by
   legitimate RTP receivers, or their certificates MUST be signed by a
   trusted Certificate Authority.  Once the DTLS-SRTP security is
   established, non-SRTCP-protected messages received from a particular
   RTP receiver are ignored by the retransmission server.  To reduce the
   impact of DTLS-SRTP overhead when communicating with different
   feedback targets on the same retransmission server, it is RECOMMENDED
   that RTP receivers and the retransmission server both support TLS
   Session Resumption without Server-side State [<a href="./rfc5077" title="&quot;Transport Layer Security (TLS) Session Resumption without Server-Side State&quot;">RFC5077</a>].  To help
   scale SRTP to handle many RTP receivers asking for retransmissions of
   identical data, implementors may consider using the same SRTP key for
   SRTP data sent to the receivers [<a href="#ref-SRTP-EKT" title="&quot;Encrypted Key Transport for Secure RTP&quot;">SRTP-EKT</a>] and be aware that such key
   sharing allows those receivers to impersonate the sender.  Thus,
   source address validation remains important.

   [<a id="ref-RFC4588">RFC4588</a>] RECOMMENDS that cryptography mechanisms be used for the
   retransmission payload format to provide protection against known
   plain-text attacks.  As discussed in [<a href="./rfc4588" title="&quot;RTP Retransmission Payload Format&quot;">RFC4588</a>], the retransmission
   payload format sets the timestamp field in the RTP header to the
   media timestamp of the original packet, and this does not compromise
   the confidentiality.  Furthermore, if cryptography is used to provide
   security services on the original stream, then the same services,
   with equivalent cryptographic strength, MUST be provided on the
   retransmission stream per [<a href="./rfc4588" title="&quot;RTP Retransmission Payload Format&quot;">RFC4588</a>].

   Finally, a retransmission server that has become subverted by an
   attacker is almost impossible to protect against as such a server can
   perform a large number of different actions to deny service to
   receivers.

<span class="h2"><a class="selflink" id="section-11" href="#section-11">11</a>.  IANA Considerations</span>

   The following contact information is used for all registrations in
   this document:

   Ali Begen
   abegen@cisco.com





<span class="grey">Ver Steeg, et al.            Standards Track                   [Page 47]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-48" ></span>
<span class="grey"><a href="./rfc6285">RFC 6285</a>                          RAMS                         June 2011</span>


   Note that the "RAMS" (value 2) in the Multicast Acquisition Method
   Registry refers to the method described in <a href="#section-6">Section 6</a> of this
   document.

<span class="h3"><a class="selflink" id="section-11.1" href="#section-11.1">11.1</a>.  Registration of SDP Attributes</span>

   This document registers a new attribute name in SDP.

        SDP Attribute ("att-field"):
        Attribute name:     rams-updates
        Long form:          Support for Updated RAMS Request Messages
        Type of name:       att-field
        Type of attribute:  Media level
        Subject to charset: No
        Purpose:            See this document
        Reference:          [<a href="./rfc6285">RFC6285</a>]
        Values:             None

<span class="h3"><a class="selflink" id="section-11.2" href="#section-11.2">11.2</a>.  Registration of SDP Attribute Values</span>

   This document registers a new value for the 'nack' attribute to be
   used with the 'rtcp-fb' attribute in SDP.  For more information about
   the 'rtcp-fb' attribute, refer to <a href="./rfc4585#section-4.2">Section&nbsp;4.2 of [RFC4585]</a>.

        Value name:     rai
        Long name:      Rapid Acquisition Indication
        Usable with:    nack
        Reference:      [<a href="./rfc6285">RFC6285</a>]

<span class="h3"><a class="selflink" id="section-11.3" href="#section-11.3">11.3</a>.  Registration of FMT Values</span>

   Within the RTPFB range, the following format (FMT) value is
   registered:

        Name:           RAMS
        Long name:      Rapid Acquisition of Multicast Sessions
        Value:          6
        Reference:      [<a href="./rfc6285">RFC6285</a>]

<span class="h3"><a class="selflink" id="section-11.4" href="#section-11.4">11.4</a>.  SFMT Values for RAMS Messages Registry</span>

   This document creates a new sub-registry for the sub-feedback message
   type (SFMT) values to be used with the FMT value registered for RAMS
   messages.  The registry is called the SFMT Values for RAMS Messages
   Registry.  This registry is managed by the IANA according to the
   Specification Required policy of [<a href="./rfc5226" title="">RFC5226</a>].





<span class="grey">Ver Steeg, et al.            Standards Track                   [Page 48]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-49" ></span>
<span class="grey"><a href="./rfc6285">RFC 6285</a>                          RAMS                         June 2011</span>


   The length of the SFMT field in the RAMS messages is a single octet,
   allowing 256 values.  The registry is initialized with the following
   entries:

   Value Name                                               Reference
   ----- -------------------------------------------------- ------------
   0     Reserved                                           [<a href="./rfc6285">RFC6285</a>]
   1     RAMS Request                                       [<a href="./rfc6285">RFC6285</a>]
   2     RAMS Information                                   [<a href="./rfc6285">RFC6285</a>]
   3     RAMS Termination                                   [<a href="./rfc6285">RFC6285</a>]
   4-254 Unassigned - Specification Required
   255   Reserved                                           [<a href="./rfc6285">RFC6285</a>]

   The SFMT values 0 and 255 are reserved for future use.

   Any registration for an unassigned SFMT value needs to contain the
   following information:

   o  Contact information of the one doing the registration, including
      at least name, address, and email.

   o  A detailed description of what the new SFMT represents and how it
      shall be interpreted.

   New RAMS functionality is intended to be introduced by using the
   extension mechanism within the existing RAMS message types not by
   introducing new message types unless it is absolutely necessary.

<span class="h3"><a class="selflink" id="section-11.5" href="#section-11.5">11.5</a>.  RAMS TLV Space Registry</span>

   This document creates a new IANA TLV space registry for the RAMS
   extensions.  The registry is called the RAMS TLV Space Registry.
   This registry is managed by the IANA according to the Specification
   Required policy of [<a href="./rfc5226" title="">RFC5226</a>].

   The length of the Type field in the TLV elements is a single octet,
   allowing 256 values.  The Type values 0 and 255 are reserved for
   future use.  The Type values between (and including) 128 and 254 are
   reserved for private extensions.












<span class="grey">Ver Steeg, et al.            Standards Track                   [Page 49]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-50" ></span>
<span class="grey"><a href="./rfc6285">RFC 6285</a>                          RAMS                         June 2011</span>


   The registry is initialized with the following entries:

   Type    Description                                     Reference
   ----    ----------------------------------------------- -------------
   0       Reserved                                        [<a href="./rfc6285">RFC6285</a>]
   1       Requested Media Sender SSRC(s)                  [<a href="./rfc6285">RFC6285</a>]
   2       Min RAMS Buffer Fill Requirement                [<a href="./rfc6285">RFC6285</a>]
   3       Max RAMS Buffer Fill Requirement                [<a href="./rfc6285">RFC6285</a>]
   4       Max Receive Bitrate                             [<a href="./rfc6285">RFC6285</a>]
   5       Request for Preamble Only                       [<a href="./rfc6285">RFC6285</a>]
   6       Supported Enterprise Number(s)                  [<a href="./rfc6285">RFC6285</a>]
   7-30    Unassigned - Specification Required
   31      Media Sender SSRC                               [<a href="./rfc6285">RFC6285</a>]
   32      RTP Seqnum of the First Packet                  [<a href="./rfc6285">RFC6285</a>]
   33      Earliest Multicast Join Time                    [<a href="./rfc6285">RFC6285</a>]
   34      Burst Duration                                  [<a href="./rfc6285">RFC6285</a>]
   35      Max Transmit Bitrate                            [<a href="./rfc6285">RFC6285</a>]
   36-60   Unassigned - Specification Required
   61      Extended RTP Seqnum of First Multicast Packet   [<a href="./rfc6285">RFC6285</a>]
   62-127  Unassigned - Specification Required
   128-254 Reserved for Private Use
   255     Reserved                                        [<a href="./rfc6285">RFC6285</a>]

   Any registration for an unassigned Type value needs to contain the
   following information:

   o  Contact information of the one doing the registration, including
      at least name, address, and email.

   o  A detailed description of what the new TLV element represents and
      how it shall be interpreted.

<span class="h3"><a class="selflink" id="section-11.6" href="#section-11.6">11.6</a>.  RAMS Response Code Space Registry</span>

   This document creates a new IANA TLV space registry for the RAMS
   response codes.  The registry is called the RAMS Response Code Space
   Registry.  This registry is managed by the IANA according to the
   Specification Required policy of [<a href="./rfc5226" title="">RFC5226</a>].

   The length of the Response field is two octets, allowing 65536 codes.
   However, in this document, the response codes have been classified
   and registered following an HTTP-style code numbering.  New response
   codes should be classified following the guidelines below:








<span class="grey">Ver Steeg, et al.            Standards Track                   [Page 50]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-51" ></span>
<span class="grey"><a href="./rfc6285">RFC 6285</a>                          RAMS                         June 2011</span>


   Code Level Purpose
   ---------- ---------------
   1xx        Informational
   2xx        Success
   3xx        Redirection
   4xx        RTP Receiver (RTP_Rx) Error
   5xx        Burst/Retransmission Source (BRS) Error

   The Response code 65535 is reserved for future use.

   The registry is initialized with the following entries:

   Code  Description                                        Reference
   ----- -------------------------------------------------- ------------
   0     A private response code is included in the message [<a href="./rfc6285">RFC6285</a>]

   100   Parameter update for RAMS session                  [<a href="./rfc6285">RFC6285</a>]

   200   RAMS request has been accepted                     [<a href="./rfc6285">RFC6285</a>]
   201   Unicast burst has been completed                   [<a href="./rfc6285">RFC6285</a>]

   400   Invalid RAMS-R message syntax                      [<a href="./rfc6285">RFC6285</a>]
   401   Invalid min buffer requirement in RAMS-R message   [<a href="./rfc6285">RFC6285</a>]
   402   Invalid max buffer requirement in RAMS-R message   [<a href="./rfc6285">RFC6285</a>]
   403   Insufficient max bitrate requirement in RAMS-R
         message                                            [<a href="./rfc6285">RFC6285</a>]
   404   Invalid RAMS-T message syntax                      [<a href="./rfc6285">RFC6285</a>]

   500   An unspecified BRS internal error has occurred     [<a href="./rfc6285">RFC6285</a>]
   501   BRS has insufficient bandwidth to start RAMS
         session                                            [<a href="./rfc6285">RFC6285</a>]
   502   Burst is terminated due to network congestion      [<a href="./rfc6285">RFC6285</a>]
   503   BRS has insufficient CPU cycles to start RAMS
         session                                            [<a href="./rfc6285">RFC6285</a>]
   504   RAMS functionality is not available on BRS         [<a href="./rfc6285">RFC6285</a>]
   505   RAMS functionality is not available for RTP_Rx     [<a href="./rfc6285">RFC6285</a>]
   506   RAMS functionality is not available for
         the requested multicast stream                     [<a href="./rfc6285">RFC6285</a>]
   507   BRS has no valid starting point available for
         the requested multicast stream                     [<a href="./rfc6285">RFC6285</a>]
   508   BRS has no reference information available for
         the requested multicast stream                     [<a href="./rfc6285">RFC6285</a>]
   509   BRS has no RTP stream matching the requested SSRC  [<a href="./rfc6285">RFC6285</a>]
   510   RAMS request to acquire the entire session
         has been denied                                    [<a href="./rfc6285">RFC6285</a>]
   511   Only the preamble information is sent              [<a href="./rfc6285">RFC6285</a>]
   512   RAMS request has been denied due to a policy       [<a href="./rfc6285">RFC6285</a>]




<span class="grey">Ver Steeg, et al.            Standards Track                   [Page 51]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-52" ></span>
<span class="grey"><a href="./rfc6285">RFC 6285</a>                          RAMS                         June 2011</span>


   Any registration for an unassigned Response code needs to contain the
   following information:

   o  Contact information of the one doing the registration, including
      at least name, address, and email.

   o  A detailed description of what the new Response code describes and
      how it shall be interpreted.

<span class="h2"><a class="selflink" id="section-12" href="#section-12">12</a>.  Contributors</span>

   Dave Oran, Magnus Westerlund, and Colin Perkins have contributed
   significantly to this specification by providing text and solutions
   to some of the issues raised during the development of this
   specification.

<span class="h2"><a class="selflink" id="section-13" href="#section-13">13</a>.  Acknowledgments</span>

   The following individuals reviewed earlier versions of this
   specification and provided helpful comments:  Joerg Ott, Roni Even,
   Dan Wing, Tony Faustini, Peilin Yang, Jeff Goldberg, Muriel
   Deschanel, Orit Levin, Guy Hirson, Tom Taylor, Xavier Marjou, Ye-Kui
   Wang, Zixuan Zou, Ingemar Johansson, Haibin Song, Ning Zong, Jonathan
   Lennox, Jose Rey, Sean Sheedy, and Keith Drage.

<span class="h2"><a class="selflink" id="section-14" href="#section-14">14</a>.  References</span>

<span class="h3"><a class="selflink" id="section-14.1" href="#section-14.1">14.1</a>.  Normative References</span>

   [<a id="ref-RFC2119">RFC2119</a>]  Bradner, S., "Key words for use in RFCs to Indicate
              Requirement Levels", <a href="https://www.rfc-editor.org/bcp/bcp14">BCP 14</a>, <a href="./rfc2119">RFC 2119</a>, March 1997.

   [<a id="ref-RFC2827">RFC2827</a>]  Ferguson, P. and D. Senie, "Network Ingress Filtering:
              Defeating Denial of Service Attacks which employ IP Source
              Address Spoofing", <a href="https://www.rfc-editor.org/bcp/bcp38">BCP 38</a>, <a href="./rfc2827">RFC 2827</a>, May 2000.

   [<a id="ref-RFC3376">RFC3376</a>]  Cain, B., Deering, S., Kouvelas, I., Fenner, B., and A.
              Thyagarajan, "Internet Group Management Protocol, Version
              3", <a href="./rfc3376">RFC 3376</a>, October 2002.

   [<a id="ref-RFC3550">RFC3550</a>]  Schulzrinne, H., Casner, S., Frederick, R., and V.
              Jacobson, "RTP: A Transport Protocol for Real-Time
              Applications", STD 64, <a href="./rfc3550">RFC 3550</a>, July 2003.

   [<a id="ref-RFC3605">RFC3605</a>]  Huitema, C., "Real Time Control Protocol (RTCP) attribute
              in Session Description Protocol (SDP)", <a href="./rfc3605">RFC 3605</a>,
              October 2003.




<span class="grey">Ver Steeg, et al.            Standards Track                   [Page 52]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-53" ></span>
<span class="grey"><a href="./rfc6285">RFC 6285</a>                          RAMS                         June 2011</span>


   [<a id="ref-RFC3711">RFC3711</a>]  Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K.
              Norrman, "The Secure Real-time Transport Protocol (SRTP)",
              <a href="./rfc3711">RFC 3711</a>, March 2004.

   [<a id="ref-RFC3810">RFC3810</a>]  Vida, R. and L. Costa, "Multicast Listener Discovery
              Version 2 (MLDv2) for IPv6", <a href="./rfc3810">RFC 3810</a>, June 2004.

   [<a id="ref-RFC4566">RFC4566</a>]  Handley, M., Jacobson, V., and C. Perkins, "SDP: Session
              Description Protocol", <a href="./rfc4566">RFC 4566</a>, July 2006.

   [<a id="ref-RFC4585">RFC4585</a>]  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)", <a href="./rfc4585">RFC 4585</a>,
              July 2006.

   [<a id="ref-RFC4588">RFC4588</a>]  Rey, J., Leon, D., Miyazaki, A., Varsa, V., and R.
              Hakenberg, "RTP Retransmission Payload Format", <a href="./rfc4588">RFC 4588</a>,
              July 2006.

   [<a id="ref-RFC4604">RFC4604</a>]  Holbrook, H., Cain, B., and B. Haberman, "Using Internet
              Group Management Protocol Version 3 (IGMPv3) and Multicast
              Listener Discovery Protocol Version 2 (MLDv2) for Source-
              Specific Multicast", <a href="./rfc4604">RFC 4604</a>, August 2006.

   [<a id="ref-RFC5077">RFC5077</a>]  Salowey, J., Zhou, H., Eronen, P., and H. Tschofenig,
              "Transport Layer Security (TLS) Session Resumption without
              Server-Side State", <a href="./rfc5077">RFC 5077</a>, January 2008.

   [<a id="ref-RFC5234">RFC5234</a>]  Crocker, D. and P. Overell, "Augmented BNF for Syntax
              Specifications: ABNF", STD 68, <a href="./rfc5234">RFC 5234</a>, January 2008.

   [<a id="ref-RFC5285">RFC5285</a>]  Singer, D. and H. Desineni, "A General Mechanism for RTP
              Header Extensions", <a href="./rfc5285">RFC 5285</a>, July 2008.

   [<a id="ref-RFC5506">RFC5506</a>]  Johansson, I. and M. Westerlund, "Support for Reduced-Size
              Real-Time Transport Control Protocol (RTCP): Opportunities
              and Consequences", <a href="./rfc5506">RFC 5506</a>, April 2009.

   [<a id="ref-RFC5576">RFC5576</a>]  Lennox, J., Ott, J., and T. Schierl, "Source-Specific
              Media Attributes in the Session Description Protocol
              (SDP)", <a href="./rfc5576">RFC 5576</a>, June 2009.

   [<a id="ref-RFC5760">RFC5760</a>]  Ott, J., Chesterfield, J., and E. Schooler, "RTP Control
              Protocol (RTCP) Extensions for Single-Source Multicast
              Sessions with Unicast Feedback", <a href="./rfc5760">RFC 5760</a>, February 2010.

   [<a id="ref-RFC5761">RFC5761</a>]  Perkins, C. and M. Westerlund, "Multiplexing RTP Data and
              Control Packets on a Single Port", <a href="./rfc5761">RFC 5761</a>, April 2010.



<span class="grey">Ver Steeg, et al.            Standards Track                   [Page 53]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-54" ></span>
<span class="grey"><a href="./rfc6285">RFC 6285</a>                          RAMS                         June 2011</span>


   [<a id="ref-RFC5764">RFC5764</a>]  McGrew, D. and E. Rescorla, "Datagram Transport Layer
              Security (DTLS) Extension to Establish Keys for the Secure
              Real-time Transport Protocol (SRTP)", <a href="./rfc5764">RFC 5764</a>, May 2010.

   [<a id="ref-RFC5888">RFC5888</a>]  Camarillo, G. and H. Schulzrinne, "The Session Description
              Protocol (SDP) Grouping Framework", <a href="./rfc5888">RFC 5888</a>, June 2010.

   [<a id="ref-RFC6051">RFC6051</a>]  Perkins, C. and T. Schierl, "Rapid Synchronisation of RTP
              Flows", <a href="./rfc6051">RFC 6051</a>, November 2010.

   [<a id="ref-RFC6128">RFC6128</a>]  Begen, A., "RTP Control Protocol (RTCP) Port for Source-
              Specific Multicast (SSM) Sessions", <a href="./rfc6128">RFC 6128</a>,
              February 2011.

   [<a id="ref-RFC6284">RFC6284</a>]  Begen, A., Wing, D., and T. Van Caenegem, "Port Mapping
              Between Unicast and Multicast RTP Sessions", <a href="./rfc6284">RFC 6284</a>,
              June 2011.

<span class="h3"><a class="selflink" id="section-14.2" href="#section-14.2">14.2</a>.  Informative References</span>

   [<a id="ref-ECN-FOR-RTP">ECN-FOR-RTP</a>]
              Westerlund, M., Johansson, I., Perkins, C., O'Hanlon, P.,
              and K. Carlberg, "Explicit Congestion Notification (ECN)
              for RTP over UDP", Work in Progress, May 2011.

   [<a id="ref-IC2009">IC2009</a>]   Begen, A., Glazebrook, N., and W. Ver Steeg, "Reducing
              Channel Change Times in IPTV with Real-Time Transport
              Protocol (IEEE Internet Computing)", May 2009.

   [<a id="ref-MULTICAST-ACQ">MULTICAST-ACQ</a>]
              Begen, A. and E. Friedrich, "Multicast Acquisition Report
              Block Type for RTP Control Protocol (RTCP) Extended
              Reports (XRs)", Work in Progress, May 2011.

   [<a id="ref-RAMS-SCENARIOS">RAMS-SCENARIOS</a>]
              Begen, A., "Considerations and Guidelines for Deploying
              the Rapid Acquisition of Multicast Sessions (RAMS)
              Method", Work in Progress, June 2011.

   [<a id="ref-RFC0768">RFC0768</a>]  Postel, J., "User Datagram Protocol", STD 6, <a href="./rfc768">RFC 768</a>,
              August 1980.

   [<a id="ref-RFC4787">RFC4787</a>]  Audet, F. and C. Jennings, "Network Address Translation
              (NAT) Behavioral Requirements for Unicast UDP", <a href="https://www.rfc-editor.org/bcp/bcp127">BCP 127</a>,
              <a href="./rfc4787">RFC 4787</a>, January 2007.






<span class="grey">Ver Steeg, et al.            Standards Track                   [Page 54]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-55" ></span>
<span class="grey"><a href="./rfc6285">RFC 6285</a>                          RAMS                         June 2011</span>


   [<a id="ref-RFC5226">RFC5226</a>]  Narten, T. and H. Alvestrand, "Guidelines for Writing an
              IANA Considerations Section in RFCs", <a href="https://www.rfc-editor.org/bcp/bcp26">BCP 26</a>, <a href="./rfc5226">RFC 5226</a>,
              May 2008.

   [<a id="ref-RFC5762">RFC5762</a>]  Perkins, C., "RTP and the Datagram Congestion Control
              Protocol (DCCP)", <a href="./rfc5762">RFC 5762</a>, April 2010.

   [<a id="ref-RFC6015">RFC6015</a>]  Begen, A., "RTP Payload Format for 1-D Interleaved Parity
              Forward Error Correction (FEC)", <a href="./rfc6015">RFC 6015</a>, October 2010.

   [<a id="ref-RFC6222">RFC6222</a>]  Begen, A., Perkins, C., and D. Wing, "Guidelines for
              Choosing RTP Control Protocol (RTCP) Canonical Names
              (CNAMEs)", <a href="./rfc6222">RFC 6222</a>, April 2011.

   [<a id="ref-SRTP-EKT">SRTP-EKT</a>] McGrew, D., Andreasen, F., Wing, D., and K. Fischer,
              "Encrypted Key Transport for Secure RTP", Work
              in Progress, March 2011.

   [<a id="ref-UPnP-IGD">UPnP-IGD</a>] UPnP Forum, "Universal Plug and Play (UPnP) Internet
              Gateway Device (IGD)", December 2010.































<span class="grey">Ver Steeg, et al.            Standards Track                   [Page 55]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-56" ></span>
<span class="grey"><a href="./rfc6285">RFC 6285</a>                          RAMS                         June 2011</span>


Authors' Addresses

   Bill Ver Steeg
   Cisco
   5030 Sugarloaf Parkway
   Lawrenceville, GA  30044
   USA

   EMail:  billvs@cisco.com


   Ali Begen
   Cisco
   181 Bay Street
   Toronto, ON  M5J 2T3
   Canada

   EMail:  abegen@cisco.com


   Tom Van Caenegem
   Alcatel-Lucent
   Copernicuslaan 50
   Antwerpen  2018
   Belgium

   EMail:  Tom.Van_Caenegem@alcatel-lucent.be


   Zeev Vax
   Magnum Semiconductor, Inc.
   591 Yosemite Drive
   Milpitas, CA  95035
   USA

   EMail:  zeev.vax@magnumsemi.com















Ver Steeg, et al.            Standards Track                   [Page 56]
</pre>