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Versions: 00                                                            
Internet Engineering Task Force                                   AVT WG
Internet Draft                                              Mark Handley
draft-ietf-avt-germ-00.txt                                           ISI
November 11, 1998
Expires: May, 1999


                     GeRM: Generic RTP Multiplexing

STATUS OF THIS MEMO

   This document is an Internet-Draft. Internet-Drafts are working
   documents of the Internet Engineering Task Force (IETF), its areas,
   and its working groups.  Note that other groups may also distribute
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   Distribution of this document is unlimited.

                                 ABSTRACT


         This document describes GeRM, an RTP payload format for
         generic multiplexing of multiple RTP streams.

         This document is a product of the Audio/Video Transport
         (AVT) working group of the Internet Engineering Task
         Force.  Comments are solicited and should be addressed to
         the working group's mailing list at rem-conf@es.net
         and/or the author.

1 Introduction

   When RTP[1] is used for end-to-end communication, each RTP data
   stream in a session should be send separately to a different UDP
   port.  This allows heterogeneous treatment of the streams by the
   network.  For example, in a multimedia conference, we may be willing
   to pay to make an RSVP[2] reservation for the audio, but unable to



Mark Handley                                                  [Page 1]


Internet Draft                    GeRM                 November 11, 1998


   reserve sufficient bandwidth for the video. Thus in the general case,
   we argue that multiplexing multiple RTP streams together should be
   avoided.

   However, there are circumstances when this general rule may not make
   a great deal of sense. If a stream is very low bandwidth, but needs
   low latency, the overhead of RTP packetisation may be too large. On
   slow modem links this can be overcome by using IP/UDP/RTP header
   compression [3], but this is a hop-by-hop compression scheme, and so
   is unsuitable for congested high-speed backbone links.

   MPEG 4 is an example of a codec that produces multiple elementary
   streams that comprise a single video stream. Many of these elementary
   streams are very low bandwidth. It makes little sense to packetise
   each of these elementary streams separately and send it to its own
   RTP/UDP port. Instead a network-aware multiplexing layer is required
   that can combine multiple elementary stream data units into a single
   RTP packet in a way that does not reduce resilience to packet
   loss[4].

   Another example is that of IP telephony gateways. In such a gateway,
   incoming PSTN calls are packetised over RTP and transmitted to a
   remote gateway, where they are turned back into PSTN calls again.
   Between any pair of gateways there may be many simultaneous telephone
   calls. If a relatively low bitrate codec is used such as GSM (approx
   14Kbps), each of these flows then gains its own IP, UDP and RTP
   headers comprising 40 bytes. With 20ms packetisation, the overhead is
   over 100%. In this case, many (if not all) of the flows expect the
   same network service. The header overhead can be significantly
   reduced if multiple unrelated flows are multiplexed together into a
   single RTP packet.

   In both these cases and other similar ones, we can design specific
   multiplexing protocols that satisfy one particular problem domain.
   Rather than do this, we propose a multiplexing protocol that attempts
   to be generic. Any pair of RTP flows with the same source and
   destination may be multiplexed together. The degree of compression
   depends on the similarity of the two flows, but the per-flow overhead
   is always less than a single RTP header (without IP or UDP), and is
   typically much better.

2 Specification

   The approach taken in GeRM is similar to that taken with IP/UDP/RTP
   header compression, in that only differences between one packet and
   the next are encoded. However, unlike IP/UDP/RTP header compression,
   GeRM does this by only encoding the differences between the RTP
   headers of the different payloads in the same multiplexed packet, and



Mark Handley                                                  [Page 2]


Internet Draft                    GeRM                 November 11, 1998


   all RTP header state is reinitialised in each new packet. As a result
   GeRM can function effectively across multiple network hops.

   The basic model is that a single IP packet contains multiple RTP
   headers each followed by its own payload. Each of these RTP headers
   followed by its payload is referred to as a sub-packet compresses
   each sub-packet header, so that fields which are predictable between
   one sub-header and the next sub-header within the same packet are not
   sent.

   Each multiplexed RTP packet has a full RTP header which contains the
   SSRC, Sequence number, Timestamp, etc corresponding to the first
   sub-packet payload, but the RTP payload type field is set to a value
   indicating this is a GeRM packet. The first sub-packet header will
   compress out completely except for the payload-type field and length
   because the full RTP header and the sub-packet header only differ in
   the payload-type.

   The second sub-packet header is then encoded based on predictable
   differences between the original RTP header for that sub-packet and
   the original RTP header for the first sub-packet.

   The third sub-packet header is then encoded off of the original RTP
   header for the second sub-packet and so forth.

   A regular RTP header has the following format:


       0                   1                   2                   3
       0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
      +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
      |V=2|P|X|  CC   |M|     PT      |       sequence number         |
      +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
      |                           timestamp                           |
      +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
      |           synchronization source (SSRC) identifier            |
      +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
      |            contributing source (CSRC) identifiers             |
      |                             ....                              |
      +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+



   A GeRM header consists of one byte followed by any RTP fields that
   are not predictable from the previous header. The parts of the RTP
   header corresponding to the bits of the GeRM header are as follows:





Mark Handley                                                  [Page 3]


Internet Draft                    GeRM                 November 11, 1998


      +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
      | V |P|X|  CC   |M|     PT      |       sequence number         |
      |0 0|0|0|0 0 0 0|1|2 2 2 2 2 2 2|3 3 3 3 3 3 3 3 3 3 3 3 3 3 3 3|
      +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
      |                           timestamp                           |
      |4 4 4 4 4 4 4 4 4 4 4 4 4 4 4 4 4 4 4 4 4 4 4 4 5 5 5 5 5 5 5 5|
      +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
      |           synchronization source (SSRC) identifier            |
      |6 6 6 6 6 6 6 6 6 6 6 6 6 6 6 6 6 6 6 6 6 6 6 6 6 6 6 6 6 6 6 6|
      +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
      |            contributing source (CSRC) identifiers             |
      |                         not compressed                        |
      +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+



   The GeRM Header is one byte:


       0  1  2  3  4  5  6  7
      +--+--+--+--+--+--+--+--+
      |B0|B1|B2|B3|B4|B5|B6|B7|
      +--+--+--+--+--+--+--+--+



   The meaning of these bits is:

   B0:

        -zero indicates that the first byte of the original RTP header
         remains unchanged from the original RTP header in the previous
         subpacket (or outer RTP header if there's no previous sub-
         packet in this packet). I.e, V, CC and P are unchanged.

        -one indicates that the first byte (byte 1) of the original RTP
         header immediately follows the GeRM header.

   B1: Contains the marker bit from the sub-packet's RTP header.

   B2:

        -zero indicates that the payload type remains unchanged.

        -one indicates that the payload type field follows the GeRM
         header and any byte 0 header that may be present. Although PT
         is a seven bit field, it is added as an eight bit field. Bit 0
         of this byte MUST be zero.



Mark Handley                                                  [Page 4]


Internet Draft                    GeRM                 November 11, 1998


   B3:

        -zero indicates that the sequence number remains unchanged.

        -one indicates that the 16 bit sequence number field follows the
         GeRM header and any byte 1 or PT header that may be present.

   B4:

        -zero indicates that the timestamp remains unchanged.

        -one indicates that the 32 bit timestamp field follows the GeRM
         header and any byte 1, PT or sequence number header that may be
         present.

   B5:

        -zero indicates that the most significant 24 bits of the SSRC
         remain unchanged.

        -one indicates that the most significant 24 bits of the SSRC
         follows the GeRM header and any byte 1, PT, sequence number or
         timestamp field that may be present.

   B6:

        -zero indicates that the least significant 8 bits of the SSRC
         are one higher than the preceding SSRC.

        -one indicates that the least significant 8 bits of the SSRC
         follows the GeRM header and any byte 1, PT, sequence number,
         timestamp or MSB SSRC header fields that may be present.

   B7:

        -zero indicates that the sub-packet length in bytes (ignoring
         the sub-packet header) is unchanged from the previous sub-
         packet.

        -one indicates that the sub-packet length (ignoring the sub-
         packet header) follows all the other GeRM headers as an 8-bit
         unsigned integer length field.

   Any CSRC fields present in the original RTP header then follow the
   GeRM headers.

3 Examples




Mark Handley                                                  [Page 5]


Internet Draft                    GeRM                 November 11, 1998


   In this section we attempt to characterise the likely behaviour of
   GeRM in some typical circumstances.

3.1 Arbitrary Streams, Same Payload Type

   Five RTP streams that originate at separate RTP sources (with SSRCs
   SSRC1 to SSRC5, sequence numbers SEQ1 to SEQ5, and timestamps T1 to
   T5) are being multiplexed together. They each use GSM compression,
   and the GSM codec uses fixed size frames.

   The compound packet is as follows:

     IP Header
     UDP Header
     RTP Header, V=0, P=0, CC=0, PT=>GERM, SEQ=SEQ1, TS=T1, SSRC=SSRC1
     GeRM Header, B0=0, B1=M1, B2=1, B3=0, B4=0, B5=0, B6=0, B7=1
       8-bit PT=>GSM
       8-bit length = length of GSM frame
     GSM payload 1
     GeRM Header, B0=0, B1=M2, B2=0, B3=1, B4=1, B5=1, B6=1, B7=0
       16-bit Sequence Number SEQ2
       32-bit Timestamp TS2
       24+8 bit SSRC SSRC2
     GSM payload 2
     ...
     GeRM Header, B0=0, B1=M5, B2=0, B3=1, B4=1, B5=1, B6=1, B7=0
       16-bit Sequence Number SEQ5
       32-bit Timestamp TS5
       24+8 bit SSRC SSRC5
     GSM payload 5



   The overhead is 40 bytes (IP+UDP+RTP) + 3 bytes (first sub-header) +
   11 bytes (each subsequent sub-header), or a total of 87 bytes, as
   opposed to 200 bytes for separate RTP packets.

   In some cases, having a multiplex stream sequence number in the outer
   RTP packet (rather than the first payload sequence number) might be
   desirable. This might be the case if we wish to add packet-level FEC
   to the multiplexed stream. In such a case, the sequence number of
   sub-packet 1 does not compress out, adding a further two bytes
   overhead.

3.2 Cooperating PSTN-IP gateways

   If several RTP streams coded with the same codec are ordinating at a
   PSTN->IP gateway and all terminate at the same IP->PSTN gateway, and



Mark Handley                                                  [Page 6]


Internet Draft                    GeRM                 November 11, 1998


   if we assume that an out-of-band signalling mechanism is used to
   communicate SSRC information at call setup time, then we can achieve
   significantly better compression.

   To do this we algorithmically generate the SSRC rather than
   allocating it randomly as specified in the RTP specification. This is
   acceptable in this context because only the remote gateway will ever
   see the SSRC.

   As consecutive flows arrive, they are given consecutive SSRCs, which
   in any event must be communicated as part of the call setup
   mechanism.  All the flows are digitised and compressed at the same
   time, so they share a common clock and hence common timestamps. If no
   silence suppression is performed, the sequence numbers can be
   consecutive too, but we do not assume this.

   As flows terminate, they will leave gaps in the SSRC space. New flows
   are then allocated the now unused SSRCs to attempt to keep the SSRC
   space as contiguous as possible.

   For the sake of example, we assume we have SSRCs 1 to 3 and 5 to 10
   in use, and that the flows with SSRC 5, 7 and 8 are being silence
   suppressed. This leaves us with flows 1,2,3,6,9 and 10 to transmit.


     IP Header
     UDP Header
     RTP Header, V=0, P=0, CC=0, PT=>GERM, SEQ=SEQ1, TS=T1, SSRC=SSRC1
     GeRM Header, B0=0, B1=M1, B2=1, B3=0, B4=0, B5=0, B6=0, B7=1
       8-bit PT=>GSM
       8-bit length = length of GSM frame
     GSM payload 1
     GeRM Header, B0=0, B1=M2, B2=0, B3=1, B4=0, B5=0, B6=0, B7=0
       16-bit Sequence Number SEQ2
     GSM payload 2
     GeRM Header, B0=0, B1=M3, B2=0, B3=1, B4=0, B5=0, B6=0, B7=0
       16-bit Sequence Number SEQ3
     GSM payload 3
     GeRM Header, B0=0, B1=M6, B2=0, B3=1, B4=0, B5=0, B6=1, B7=0
       16-bit Sequence Number SEQ6
       8-bit LSByte of SSRC 6
     GSM payload 6
     GeRM Header, B0=0, B1=M9, B2=0, B3=1, B4=0, B5=0, B6=1, B7=0
       16-bit Sequence Number SEQ9
       8-bit LSByte of SSRC 9
     GSM payload 9
     GeRM Header, B0=0, B1=M10, B2=0, B3=1, B4=0, B5=0, B6=0, B7=0
       16-bit Sequence Number SEQ10



Mark Handley                                                  [Page 7]


Internet Draft                    GeRM                 November 11, 1998


     GSM payload 10



   Thus the overhead is 40 bytes for the IP/UDP/RTP, 3 bytes for sub-
   header 1, 4 bytes each for sub-headers 2, 3, and 10, and 5 bytes for
   sub-headers 6 and 9. This totals 65 bytes against 240 bytes for
   separate IP/UDP/RTP headers per flow. Typically each new flow being
   included in the packet will require 4 to 5 bytes of overhead in
   addition to the compressed data itself.

   We might envisage ways in which sequence numbers of flows can also be
   manipulated as a flow returns from silence suppression (step the
   sequence number to match that of the flow with preceding SSRC) if we
   are sure that the flow will be removed from RTP at the next
   destination. This would reduce the per-flow overhead to between 1 and
   5 bytes depending on the effectiveness of this mapping. Whether this
   is worth pursuing is an open issue that providers may consider. It
   MUST NOT be done unless the destination knows to expect such
   behaviour and not treat it as loss.


   Appendix A: Author's Address

   Mark Handley
   Information Sciences Institute,
   University of Southern California,
   c/o MIT Laboratory for Computer Science,
   545 Technology Square,
   Cambridge, MA 02139,
   United States
   electronic mail: mjh@isi.edu

4 Bibliography

   [1]  H. Schulzrinne, S. Casner, R. Frederick and V. Jacobson "RTP: A
   Transport Protocol for Real-Time Applications" RFC 1889.  [2] R.
   Braden, Ed., L. Zhang, S. Berson, S. Herzog, S.  Jamin, "Resource
   ReSerVation Protocol (RSVP) -- Version 1 Functional Specification",
   RFC 2205 [3] S. Casner, V. Jacobson, "Compressing IP/UDP/RTP Headers
   for Low-Speed Serial Links", Internet Draft.  [4] M. Handley,
   "Guidelines for writers of RTP payload format specifications",
   Internet Draft.








Mark Handley                                                  [Page 8]