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Versions: 00 01                                                         
Internet Engineering Task Force                              Koichi Yano
Internet Draft                                          Matthew Podolsky
draft-podolsky-avt-rtprx-01.txt                           Steven McCanne
March 10, 2000
Expires:  September 10, 2000

           RTP Profile for RTCP-based Retransmission Request
                          for Unicast session.

Status of this Memo

   This document is an Internet-Draft and is in full conformance with
   all provisions of Section 10 of RFC2026.

   Internet-Drafts are working documents of the Internet Engineering
   Task Force (IETF), its areas, and its working groups.  Note that
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   This document specifies a new RTP profile for retransmission of lost
   packets of unicast multimedia sessions.  We refer to this profile as
   "RTP/RX".  This profile follows RFC 1889 as it is for data exchange.
   Specifically for unicast session, it changes the RTCP interval and
   defines a new RTCP packet type for retransmission requests.  This
   document also describes how retransmission requests and
   retransmission data may be sent within RTP.

1.  Conventions and Acronyms

   SHOULD NOT, RECOMMENDED, MAY, and OPTIONAL, when they appear in this
   document, are to be interpreted as described in [2].

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2. Introduction

   RTP was designed as a flexible protocol capable of transporting real-
   time data over multicast or unicast.  It has been widely deployed and
   used extensively for transmitting real-time (or "near" real-time)
   multimedia signals, commonly called media streams.  And although
   considerable effort went into the design of RTP so that it could
   support multi-party interactive conferencing over multicast, a common
   use of it today is for one-way non-interactive transmission of media
   streams.  Because the communication is non-interactive, extra
   playback buffering and delay may be tolerable, which in turn enables
   receivers to request and receive retransmissions of lost RTP packets
   before their scheduled play-out times.  For example, popular
   commercial products like RealNetworks' G2 Player and Microsoft's
   NetShow use RTP as the underlying transport protocol for their media
   streams, and use retransmissions in unicast delivery sessions.

   However, as noted in [4], no standard exists for requesting the
   retransmission of streaming media.  As a result, various independent
   and incompatible research and commercial schemes have been developed
   for retransmission of RTP streams [3, 5]. This purpose of this note
   is to define a standard retransmission request framework for use with
   unicast RTP streams.

3.  RTCP Packet Forms and Protocol Behavior

   The section "RTP Profiles and Payload Format Specification" of RFC
   1889 enumerates a number of items that can be specified or modified
   in a profile.  This profile, RTP/RX, follows all of the default
   and/or recommended aspects of the Audio and Video Profile,
   RTP/AVP[6], except for the items which are described below.  This
   section addresses the items which are modified by this profile.

   RTCP packet types:
      A new RTCP packet types, RTCP_NACK, is defined for requesting a
      retransmission.  The packet format is specified in section 4.2.

   RTCP report interval:
      In order to allow immediate RTCP response to packet loss, the RTCP
      report interval restriction in RFC 1889 should be relaxed.
      However, in order to prevent RTCP packets from flooding the
      network, the bandwidth constraint of RFC 1889 should be preserved.
      Recommendations about the RTCP transmission frequency are
      specified in section 4.1.

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4. Control packets for unicast session

   This note proposes extending the RTCP specification to accommodate
   retransmission requests.  RTCP is a reasonable place for putting
   control packets into because it is a well-established control
   protocol and no extra connections need to be established.  However,
   the RTCP interval specified in [1] is too large for timely
   retransmission requests.  Thus, the current RTCP minimum transmission
   interval should be relaxed.

4.1. RTCP Transmission Bandwidth

   The RTCP transmission interval specified in [1] is not frequent
   enough to allow efficient control of a unicast session needing to
   request retransmissions of lost packets.  RFC 1889 recommends a
   minimum interval of 5 seconds between control packets.  In order to
   make RTCP applicable for unicast retransmission, a packet-granularity
   RTCP interval is necessary.

   The minimum interval restriction SHOULD be removed for unicast RTP
   sessions so that retransmission requests are allowed to be sent
   immediately after loss detection.  In order to prevent  RTCP packets
   from causing network congestion, it is RECOMMENDED that the average
   bandwidth allocated to RTCP be fixed at 5% of the overall session
   bandwidth.  The recommended default values are 1/4 of the RTCP
   bandwidth (1.25%) for the data sender and the remainder of the
   bandwidth (3.75%) for the receiver, following the conventions of RFC

   If the size of RTP payload packets is 1 KBytes and the size of RTCP
   packets is 40 bytes, the 3.75% limitation is large enough for a
   receiver to send an RTCP packet at every other data packet interval.
   The interval is frequent enough to allow timely retransmission

   The original RTCP packets (SR/RR) also SHOULD be sent based on RFC
   1889.  In order to regulate the sending rate of RTCP packets within
   the RTCP bandwidth, the deployment of a traffic shaping scheme such
   as a token backet is RECOMMENDED. The token rate should be set at
   (rtcp_bw / rtcp_size), where rtcp_bw is the RTCP bandwidth and
   rtcp_size is the size of RTCP packets.  The bucket size should be
   determined according to the playback buffer size.  As long as a token
   is in the bucket, the RTCP packet is sent immediately, and the long-
   term RTCP rate is regulated within the RTCP bandwidth limit.

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4.2. Packet format for requesting retransmission

   This section defines a packet format for a negative acknowledgment
   (NACK) packet, which is sent from the receiver to the sender and
   which asks for the retransmission of one or more RTP packets.  The
   format of NACK packets is as follows:

    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|   RC    | PT=RTCP_NACK  |          length               |
   |                  SSRC of packet sender                        |
   |                  SSRC_1                                       |
   |             FSN               |            BLP                |
   |                  SSRC_2                                       |
   |             FSN               |            BLP                |
   |                                                               |

   The various fields V, P, RC, SSRC and length are defined in the RTP
   specification [1].  The FSN and BLP fields have similar definitions
   to those given for the RTCP_NACK packet specified in [3].  We
   summarize the meaning of all of the fields below:

   version (V): 2 bits
      This field identifies the RTP version.  The current version is 2.

   padding (P): 1 bit
      If set, the padding bit indicates that the RTCP NACK packet
      contains additional padding octets at the end which are not part
      of the retransmission control information but are included in the
      length field.

   report count (RC): 5 bits
      The number of NACK blocks for different SSRC.

   packet type (PT): 8 bits
      This is the RTCP packet type which identifies the packet as being
      an RTCP NACK.

   length: 16 bits
      The length of this RTCP NACK packet in 32-bit words minus one,
      including the header and any padding.  This conforms with the

Yano, Podolsky, McCanne                                         [Page 4]

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      definition of the length field used in RTCP sender and receiver
      reports [1].

   SSRC: 16 bits
      The synchronization source identifier for the originator of this
      NACK packet.

   SSRC_n: 16 bits
      The SSRC identifier of the source to which the information in this
      NACK report block pertains.

   first sequence number (FSN): 16 bits
      The FSN corresponds to an RTP sequence number of a media stream.
      A NACK packet with a FSN field set to N is to be interpreted as a
      signal that the receiver has not received RTP data packet N, and
      that the receiver is requesting the retransmission of packet N.

   bitmask of following lost packets (BLP): 16 bits
      The BLP allows for the reporting of losses of the 16 RTP packets
      immediately following the RTP packet indicated by the FSN.  The
      BLP's definition is identical to that given in [3].  Denoting the
      BLP's least significant bit as bit 1, and its most significant bit
      as bit 16, then bit i of the bitmask is set to 1 if the sender has
      not received RTP packet number FSN+i (modulo 2^16) and the
      receiver is requesting its retransmission; bit i is set to 0
      otherwise.  Note that the sender MUST NOT assume that a receiver
      has received a packet because its bitmask was set to 0.  For
      example, the BLP is set to 0x00001 if the packet corresponding to
      the FSN and the following packet have been lost.  However, the
      sender cannot infer that packets FSN+2 through FSN+16 have been
      received simply because bits 2 through 16 of the BLP are 0; all
      the sender knows is that their retransmission has not been

   Note that the receiver may detect RTP packet loss via sequence number
   gaps, timer expirations, or other methods; however, details of the
   loss detection scheme are outside of the scope of this document.

5. Examples

5.1  A simple retransmission request system

   The simplest way to realize loss recovery using retransmission is
   that the receiver decides whether a NACK packet should be sent or not
   and the sender replies all retransmission requests.  The receiver
   sends NACKs when it determines that one or more data packets have
   been lost and that their playback time has not yet passed.

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   This scheme can be realized without an extra control packet between
   the sender and the receiver. However it could results in many late
   packets that arrive after play out time because it does not consider
   transmission delay.

5.2.  Receiver-based decisions on requesting retransmissions

   Before making a retransmission request at time Tr of packet N
   (scheduled for play-out at time Tp[N]), the receiver checks if

   Tr + eRTT < Tp[N]

   where eRTT is an estimate of the round trip time (RTT).  If true the
   receiver assumes the retransmission will arrive in time, and it sends
   the request for packet N.  If false the receiver does not send a
   retransmission request.  The above check can be made more general to
   account for receiver decoding delays and the sender's request
   response time.  RTT can be measured elapsed time between each NACK
   and the retransmission packet.  eRTT is calculated from the RTT
   measurements through Karn's algorithm[7].

5. References

   [1] Schulzrinne, H., Casner, S., Frederick, R., and V. Jacobson,
   "RTP: A transport protocol for real-time applications," RFC 1889,
   January 1996.

   [2] S. Bradner, "Key words for use in RFCs to indicate requirement
   levels," Request for Comments (Best Current Practice) 2119, Internet
   Engineering Task Force, Mar. 1997.

   [3] Turletti, T. and Huitema, C., "RTP Payload Format for H.261 Video
   Streams," RFC 2032, October 1996.

   [4] Perkins, C. and Hodson, O., "Options for Repair of Streaming
   Media" RFC 2354, June 1998.

   [5] R. X. Xu, A. C. Myers, H. Zhang, and R. Yavatkar.  Resilient
   multicast support for continuous media applications.  In Proceedings
   of the 7th International Workshop on Network and Operating Systems
   Support for Digital Audio and Video (NOSSDAV'97), Washington
   University in St. Louis, Missouri, May 1997.

   [6] Schulzrinne, H., "RTP Profile for Audio and Video Conferences
   with Minimal Control," RFC 1890, January 1996.

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   [7] P. Karn and C. Partridge, "Improving Round-Trip Time Estimates in
   Reliable Transport Protocol."  ACM Transactions on Computer Systems
   (TOCS), Vol. 9, No. 4, pp. 364-373, November 1991


   Matthew Podolsky
   UC Berkeley Computer Science Dept.
   Phone: 510-642-8905
   Email: podolsky@eecs.berkeley.edu
   URL: http://www.eecs.berkeley.edu/~podolsky

   Koichi Yano
   UC Berkeley Computer Science Dept.
   Phone: 510-642-9513
   Email: yano@cs.berkeley.edu
   URL: http://www.cs.berkeley.edu/~yano

   Steven McCanne
   FastForward Networks
   Email: mccanne@cs.berkeley.edu

   This draft was created in March 10, 2000.
   It expires  September 10, 2000.

Yano, Podolsky, McCanne                                         [Page 7]