





Network Working Group                                        B. Thompson
Request for Comments: 4170                                      T. Koren
BCP: 110                                                         D. Wing
Category: Best Current Practice                            Cisco Systems
                                                           November 2005


             Tunneling Multiplexed Compressed RTP (TCRTP)

Status of This Memo

   This document specifies an Internet Best Current Practices for the
   Internet Community, and requests discussion and suggestions for
   improvements.  Distribution of this memo is unlimited.

Copyright Notice

   Copyright (C) The Internet Society (2005).

Abstract

   This document describes a method to improve the bandwidth utilization
   of RTP streams over network paths that carry multiple Real-time
   Transport Protocol (RTP) streams in parallel between two endpoints,
   as in voice trunking.  The method combines standard protocols that
   provide compression, multiplexing, and tunneling over a network path
   for the purpose of reducing the bandwidth used when multiple RTP
   streams are carried over that path.























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Table of Contents

   1. Introduction ....................................................3
      1.1. Is Bandwidth Costly? .......................................3
      1.2. Overview of Protocols ......................................3
      1.3. Document Focus .............................................4
      1.4. Choice of Enhanced CRTP ....................................4
      1.5. Reducing TCRTP Overhead ....................................4
   2. Protocol Operation and Recommended Extensions ...................4
      2.1. Models .....................................................5
      2.2. Header Compression: ECRTP ..................................5
           2.2.1. Synchronizing ECRTP States ..........................5
           2.2.2. Out-of-Order Packets ................................6
      2.3. Multiplexing: PPP Multiplexing .............................6
           2.3.1. PPP Multiplex Transmitter Modifications for
                  Tunneling ...........................................7
           2.3.2. Tunneling Inefficiencies ............................8
      2.4. Tunneling: L2TP ............................................8
           2.4.1. Tunneling and DiffServ ..............................9
      2.5. Encapsulation Formats ......................................9
   3. Bandwidth Efficiency ...........................................10
      3.1. Multiplexing Gains ........................................10
      3.2. Packet Loss Rate ..........................................10
      3.3. Bandwidth Calculation for Voice and Video Applications ....10
           3.3.1. Voice Bandwidth Calculation Example ................12
           3.3.2. Voice Bandwidth Comparison Table ...................13
           3.3.3. Video Bandwidth Calculation Example ................13
           3.3.4. TCRTP over ATM .....................................14
           3.3.5. TCRTP over Non-ATM Networks ........................14
   4. Example Implementation of TCRTP ................................15
      4.1. Suggested PPP and L2TP Negotiation for TCRTP ..............17
      4.2. PPP Negotiation TCRTP .....................................17
           4.2.1. LCP Negotiation ....................................17
           4.2.2. IPCP Negotiation ...................................18
      4.3. L2TP Negotiation ..........................................19
           4.3.1. Tunnel Establishment ...............................19
           4.3.2. Session Establishment ..............................19
           4.3.3. Tunnel Tear Down ...................................20
   5. Security Considerations ........................................20
   6. Acknowledgements ...............................................21
   7. References .....................................................21
      7.1. Normative References ......................................21
      7.2. Informative References ....................................22








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1.  Introduction

   This document describes a way to combine existing protocols for
   compression, multiplexing, and tunneling to save bandwidth for some
   RTP applications.

1.1.  Is Bandwidth Costly?

   On certain links, such as customer access links, the cost of
   bandwidth is widely acknowledged to be a significant concern.
   protocols such as CRTP (Compressed RTP, [CRTP]) are well suited to
   help bandwidth inefficiencies of protocols such as VoIP over these
   links.

   Unacknowledged by many, however, is the cost of long-distance WAN
   links.  While some voice-over-packet technologies such as Voice over
   ATM (VoAAL2, [I.363.2]) and Voice over MPLS provide bandwidth
   efficiencies (because both technologies lack IP, UDP, and RTP
   headers), neither VoATM nor VoMPLS provide direct access to voice-
   over-packet services available to Voice over IP.  Thus, goals of WAN
   link cost reduction are met at the expense of lost interconnection
   opportunities to other networks.

   TCRTP solves the VoIP bandwidth discrepancy, especially for large,
   voice-trunking applications.

1.2.  Overview of Protocols

   Header compression is accomplished using Enhanced CRTP (ECRTP,
   [ECRTP]).  ECRTP is an enhancement to classical CRTP [CRTP] that
   works better over long delay links, such as the end-to-end tunneling
   links described in this document.  This header compression reduces
   the IP, UDP, and RTP headers.

   Multiplexing is accomplished using PPP Multiplexing [PPP-MUX].

   Tunneling PPP is accomplished by using L2TP [L2TPv3].

   CRTP operates link-by-link; that is, to achieve compression over
   multiple router hops, CRTP must be employed twice on each router --
   once on ingress, again on egress.  In contrast, TCRTP described in
   this document does not require any additional per-router processing
   to achieve header compression.  Instead, headers are compressed end-
   to-end, saving bandwidth on all intermediate links.







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1.3.  Document Focus

   This document is primarily concerned with bandwidth savings for Voice
   over IP (VoIP) applications over high-delay networks.  However, the
   combinations of protocols described in this document can be used to
   provide similar bandwidth savings for other RTP applications such as
   video, and bandwidth savings are included for a sample video
   application.

1.4.  Choice of Enhanced CRTP

   CRTP [CRTP] describes the use of RTP header compression on an
   unspecified link layer transport, but typically PPP is used.  For
   CRTP to compress headers, it must be implemented on each PPP link.  A
   lot of context is required to successfully run CRTP, and memory and
   processing requirements are high, especially if multiple hops must
   implement CRTP to save bandwidth on each of the hops.  At higher line
   rates, CRTP's processor consumption becomes prohibitively expensive.

   To avoid the per-hop expense of CRTP, a simplistic solution is to use
   CRTP with L2TP to achieve end-to-end CRTP.  However, as described in
   [ECRTP], CRTP is only suitable for links with low delay and low loss.
   However, once multiple router hops are involved, CRTP's expectation
   of low delay and low loss can no longer be met.  Further, packets can
   arrive out of order.

   Therefore, this document describes the use of Enhanced CRTP [ECRTP],
   which supports high delay, both packet loss, and misordering between
   the compressor and decompressor.

1.5.  Reducing TCRTP Overhead

   If only one stream is tunneled (L2TP) and compressed (ECRTP), there
   are little bandwidth savings.  Multiplexing is helpful to amortize
   the overhead of the tunnel header over many RTP payloads.  The
   multiplexing format proposed by this document is PPP multiplexing
   [PPP-MUX].  See Section 2.3 for details.

2.  Protocol Operation and Recommended Extensions

   This section describes how to combine three protocols: Enhanced CRTP,
   PPP Multiplexing, and L2TP Tunneling, to save bandwidth for RTP
   applications such as Voice over IP.








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2.1.  Models

   TCRTP can typically be implemented in two ways.  The most
   straightforward is to implement TCRTP in the gateways terminating the
   RTP streams:

       [voice gateway]---[voice gateway]
                       ^
                       |
                 TCRTP over IP

   Another way TCRTP can be implemented is with an external
   concentration device.  This device could be placed at strategic
   places in the network and could dynamically create and destroy TCRTP
   sessions without the participation of RTP-generating endpoints.

       [voice GW]\                                   /[voice GW]
       [voice GW]---[concentrator]---[concentrator]---[voice GW]
       [voice GW]/                                   \[voice GW]
                  ^                ^                ^
                  |                |                |
             RTP over IP     TCRTP over IP     RTP over IP

   Such a design also allows classical CRTP [CRTP] to be used on links
   with only a few active flows per link (where TCRTP isn't efficient;
   see Section 3):

       [voice GW]\                                   /[voice GW]
       [voice GW]---[concentrator]---[concentrator]---[voice GW]
       [voice GW]/                                   \[voice GW]
                  ^                ^                ^
                  |                |                |
           CRTP over IP     TCRTP over IP     RTP over IP

2.2.  Header Compression: ECRTP

   As described in [ECRTP], classical CRTP [CRTP] is not suitable over
   long-delay WAN links commonly used when tunneling, as proposed by
   this document.  Thus, ECRTP should be used instead of CRTP.

2.2.1.  Synchronizing ECRTP States

   When the compressor receives an RTP packet that has an unpredicted
   change in the RTP header, the compressor should send a COMPRESSED_UDP
   packet (described in [ECRTP]) to synchronize the ECRTP decompressor
   state.  The COMPRESSED_UDP packet updates the RTP context in the
   decompressor.




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   To ensure delivery of updates of context variables, COMPRESSED_UDP
   packets should be delivered using the robust operation described in
   [ECRTP].

   Because the "twice" algorithm described in [ECRTP] relies on UDP
   checksums, the IP stack on the RTP transmitter should transmit UDP
   checksums.  If UDP checksums are not used, the ECRTP compressor
   should use the CRTP Headers checksum described in [ECRTP].

2.2.2.  Out-of-Order Packets

   Tunneled transport does not guarantee ordered delivery of packets.
   Therefore, the ECRTP decompressor must operate correctly in the
   presence of out of order packets.

   The order of packets for RTP is determined by the RTP sequence
   number.  To add robustness in case of packet loss or packet
   reordering, ECRTP sends short deltas together with the full value
   when updating context variables, and repeats the updates in N
   packets, where N is an engineered constant tuned to the kind of pipe
   ECRTP is used for.

   By contrast, [ROHC] compresses out the sequence number and another
   layer is necessary for [ROHC] to handle out-of-order delivery of
   packets over a tunnel [REORDER].

2.3.  Multiplexing: PPP Multiplexing

   Both CRTP and ECRTP require a layer two protocol that allows
   identifying different protocols.  [PPP] is suited for this.

   When CRTP is used inside of a tunnel, the header compression
   associated with CRTP will reduce the size of the IP, UDP, and IP
   headers of the IP packet carried in the tunnel.  However, the tunnel
   itself has overhead due to its IP header and the tunnel header (the
   information necessary to identify the tunneled payload).  One way to
   reduce the overhead of the IP header and tunnel header is to
   multiplex multiple RTP payloads in a single tunneled packet.

   [PPP-MUX] describes an encapsulation that combines multiple PPP
   payloads into one multiplexed payload.  PPP multiplexing allows any
   supported PPP payload type to be multiplexed.  This multiplexed frame
   is then carried as a single PPPMUX payload in the IP tunnel.  This
   allows multiple RTP payloads to be carried in a single IP tunnel
   packet and allows the overhead of the uncompressed IP and tunnel
   headers to be amortized over multiple RTP payloads.





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   During PPP establishment of the TCRTP tunnel, only LCP and IPCP (for
   header compression) are required -- IP addresses do not need to be
   negotiated, nor is authentication necessary.  See Section 4.1 for
   details.

2.3.1.  PPP Multiplex Transmitter Modifications for Tunneling

   Section 1.2 of [PPP-MUX] describes an example transmitter procedure
   that can be used to implement a PPP Multiplex transmitter.  The
   transmission procedure described in this section includes a parameter
   MAX-SF-LEN that is used to limit the maximum size of a PPP Multiplex
   frame.

   There are two reasons for limiting the size of a PPP Multiplex frame.
   First, a PPPMUX frame should never exceed the Maximum Receive Unit
   (MRU) of a physical link.  Second, when a PPP session and its
   associated flow control are bound to a physical link, the MAX-SF-LEN
   parameter forms an upper limit on the amount of time a multiplex
   packet can be held before being transmitted.  When flow control for
   the PPP Multiplex transmitter is bound to a physical link, the clock
   rate of the physical link can be used to pull frames from the PPP
   Multiplex transmitter.

   This type of flow control limits the maximum amount of time a PPP
   multiplex frame can be held before being transmitted to MAX-SF-LEN /
   Link Speed.

   Tunnel interfaces are typically not bound to physical interfaces.
   Because of this, a tunnel interface has no well-known transmission
   rate associated with it.  This means that flow control in the PPPMUX
   transmitter cannot rely on the clock of a physical link to pull
   frames from the multiplex transmitter.  Instead, a timer must be used
   to limit the amount of time a PPPMUX frame can be held before being
   transmitted.  The timer along with the MAX-SF-LEN parameter should be
   used to limit the amount of time a PPPMUX frame is held before being
   transmitted.

   The following extensions to the PPPMUX transmitter logic should be
   made for use with tunnels.  The flow control logic of the PPP
   transmitter should be modified to collect incoming payloads until one
   of two events has occurred:

          (1)  a specific number of octets, MAX-SF-LEN, has arrived at
               the multiplexer, or

          (2)  a timer, called T, has expired.





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   When either condition is satisfied, the multiplexed PPP payload is
   transmitted.

   The purpose of MAX-SF-LEN is to ensure that a PPPMUX payload does not
   exceed the MTU size of any of the possible physical links that the
   tunnel can be associated with.  The value of MAX-SF-LEN should be
   less than or equal to the minimum of MRU-2 (maximum size of length
   field) and 16,383 (14 bits) for all possible physical interfaces that
   the tunnel may be associated with.

   The timer T provides an upper delay bound for tunnel interfaces.
   Timer T is reset whenever a multiplexed payload is sent to the next
   encapsulation layer.  The behavior of this timer is similar to AAL2's
   Timer_CU described in [I.363.2].  Each PPPMUX transmitter should have
   its own Timer T.

   The optimal values for T will vary depending upon the rate at which
   payloads are expected to arrive at the multiplexer and the delay
   budget for the multiplexing function.  For voice applications, the
   value of T would typically be 5-10 milliseconds.

2.3.2.  Tunneling Inefficiencies

   To get reasonable bandwidth efficiency using multiplexing within an
   L2TP tunnel, multiple RTP streams should be active between the source
   and destination of an L2TP tunnel.

   If the source and destination of the L2TP tunnel are the same as the
   source and destination of the ECRTP sessions, then the source and
   destination must have multiple active RTP streams to get any benefit
   from multiplexing.

   Because of this limitation, TCRTP is mostly useful for applications
   where many RTP sessions run between a pair of RTP endpoints.  The
   number of simultaneous RTP sessions required to reduce the header
   overhead to the desired level depends on the size of the L2TP header.
   A smaller L2TP header will result in fewer simultaneous RTP sessions
   being required to produce bandwidth efficiencies similar to CRTP.

2.4.  Tunneling: L2TP

   L2TP tunnels should be used to tunnel the ECRTP payloads end to end.
   L2TP includes methods for tunneling messages used in PPP session
   establishment, such as NCP.  This allows [IPCP-HC] to negotiate ECRTP
   compression/decompression parameters.






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2.4.1.  Tunneling and DiffServ

   RTP streams may be marked with Expedited Forwarding (EF) bits, as
   described in [EF-PHB].  When such a packet is tunneled, the tunnel
   header must also be marked for the same EF bits, as required by
   [EF-PHB].  It is important to not mix EF and non-EF traffic in the
   same EF-marked multiplexed tunnel.

2.5.  Encapsulation Formats

   The packet format for an RTP packet, compressed with RTP header
   compression as defined in ECRTP, is:

        +---------+---------+-------------+-----------------------+
        |         |   MSTI  |             |                       |
        | Context |         |     UDP     |                       |
        |   ID    |   Link  |   Checksum  |       RTP Data        |
        |         | Sequence|             |                       |
        |  (1-2)  |   (1)   |     (0-2)   |                       |
        +---------+---------+-------------+-----------------------+

   The packet format of a multiplexed PPP packet as defined by [PPP-MUX]
   is:

        +-------+---+------+-------+-----+   +---+------+-------+-----+
        | Mux   |P L|      |       |     |   |P L|      |       |     |
        | PPP   |F X|Len1  |  PPP  |     |   |F X|LenN  |  PPP  |     |
        | Prot. |F T|      | Prot. |Info1| ~ |F T|      | Prot. |InfoN|
        | Field |          | Field1|     |   |          |FieldN |     |
        | (1)   |1-2 octets| (0-2) |     |   |1-2 octets| (0-2) |     |
        +-------+----------+-------+-----+   +----------+-------+-----+

   The combined format used for TCRTP with a single payload is all of
   the above packets concatenated.  Here is an example with one payload:

        +------+-------+----------+-------+-------+-----+-------+----+
        | IP   | Mux   |P L|      |       |       | MSTI|       |    |
        |header| PPP   |F X|Len1  |  PPP  |Context|     | UDP   |RTP |
        | (20) | Proto |F T|      | Proto |  ID   | Link| Cksum |Data|
        |      | Field |          | Field1|       | Seq |       |    |
        |      | (1)   |1-2 octets| (0-2) | (1-2) | (1) | (0-2) |    |
        +------+-------+----------+-------+-------+-----+-------+----+
               |<------------- IP payload ------------------------->|
                       |<----- PPPmux payload --------------------->|

   If the tunnel contains multiplexed traffic, multiple "PPPMux
   payload"s are transmitted in one IP packet.




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3.  Bandwidth Efficiency

   The expected bandwidth efficiency attainable with TCRTP depends upon
   a number of factors.  These factors include multiplexing gain,
   expected packet loss rate across the network, and rates of change of
   specific fields within the IP and RTP headers.  This section also
   describes how TCRTP significantly enhances bandwidth efficiency for
   voice over IP over ATM.

3.1.  Multiplexing Gains

   Multiplexing reduces the overhead associated with the layer 2 and
   tunnel headers.  Increasing the number of CRTP payloads combined into
   one multiplexed PPP payload increases multiplexing gain.  As traffic
   increases within a tunnel, more payloads are combined in one
   multiplexed payload.  This will increase multiplexing gain.

3.2.  Packet Loss Rate

   Loss of a multiplexed packet causes packet loss for all of the flows
   within the multiplexed packet.

   When the expected loss rate in a tunnel is relatively low (less than
   perhaps 5%), the robust operation (described in [ECRTP]) should be
   sufficient to ensure delivery of state changes.  This robust
   operation is characterized by a parameter N, which means that the
   probability of more than N adjacent packets getting lost on the
   tunnel is small.

   A value of N=1 will protect against the loss of a single packet
   within a compressed session, at the expense of bandwidth.  A value of
   N=2 will protect against the loss of two packets in a row within a
   compressed session and so on.  Higher values of N have higher
   bandwidth penalties.

   The optimal value of N will depend on the loss rate in the tunnel.
   If the loss rate is high (above perhaps 5%), more advanced techniques
   must be employed.  Those techniques are beyond the scope of this
   document.

3.3.  Bandwidth Calculation for Voice and Video Applications

   The following formula uses the factors described above to model per-
   flow bandwidth usage for both voice and video applications.  These
   variables are defined:






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   SOV-TCRTP, unit: octet.  Per-payload overhead of ECRTP and the
          multiplexed PPP header.  This value does not include
          additional overhead for updating IP ID or the RTP Time Stamp
          fields (see [ECRTP] for details on IP ID).  The value assumes
          the use of the COMPRESSED_RTP payload type.  It consists of 1
          octet for the ECRTP context ID, 1 octet for COMPRESSED_RTP
          flags, 2 octets for the UDP checksum, 1 octet for PPP protocol
          ID, and 1 octet for the multiplexed PPP length field.  The
          total is 6 octets.

   POV-TCRTP, unit: octet.  Per-packet overhead of tunneled ECRTP.  This
          is the overhead for the tunnel header and the multiplexed PPP
          payload type.  This value is 20 octets for the IP header, 4
          octets for the L2TPv3 header and 1 octet for the multiplexed
          PPP protocol ID.  The total is 25 octets.

   TRANSMIT-LENGTH, unit: milliseconds.  The average duration of a
          transmission (such as a talk spurt for voice streams).

   SOV-TSTAMP, unit: octet.  Additional per-payload overhead of the
          COMPRESSED_UDP header that includes the absolute time stamp
          field.  This value includes 1 octet for the extra flags field
          in the COMPRESSED_UDP header and 4 octets for the absolute
          time stamp, for a total of 5 octets.

   SOV-IPID, unit: octet.  Additional per-payload overhead of the
          COMPRESSED_UDP header that includes the absolute IPID field.
          This value includes 2 octets for the absolute IPID.  This
          value also includes 1 octet for the extra flags field in the
          COMPRESSED_UDP header.  The total is 3 octets.

   IPID-RATIO, unit: integer values 0 or 1.  Indicates the frequency at
          which IPID will be updated by the compressor.  If IPID is
          changing randomly and thus always needs to be updated, then
          the value is 1.  If IPID is changing by a fixed constant
          amount between payloads of a flow, then IPID-RATIO will be 0.
          The value of this variable does not consider the IPID value at
          the beginning of a voice or video transmission, as that is
          considered by the variable TRANSMIT-LENGTH.  The
          implementation of the sending IP stack and RTP application
          controls this behavior.  See Section 1.1.

   NREP, unit: integer (usually a number between 1 and 3).  This is the
          number of times an update field will be repeated in ECRTP
          headers to increase the delivery rate between the compressor
          and decompressor.  For this example, we will assume NREP=2.

   PAYLOAD-SIZE, unit: octets.  The size of the RTP payload in octets.



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   MUX-SIZE, unit: count.  The number of PPP payloads multiplexed into
          one multiplexed PPP payload.

   SAMPLE-PERIOD, unit: milliseconds.  The average delay between
          transmissions of voice or video payloads for each flow in the
          multiplex.  For example, in voice applications the value of
          this variable would be 10ms if all calls have a 10ms sample
          period.

   The formula is:

     SOV-TOTAL = SOV-TCRTP + SOV-TSTAMP * (NREP * SAMPLE-PERIOD /
                 TRANSMIT-LENGTH) + SOV-IPID * IPID-RATIO

     BANDWIDTH = ((PAYLOAD-SIZE + SOV-TOTAL + (POV-TCRTP / MUX-SIZE)) *
                 8) / SAMPLE-PERIOD)

   The results are:

     BANDWIDTH, unit: kilobits per second.  The average amount of
               bandwidth used per voice or video flow.

     SOV-TOTAL = The total amount of per-payload overhead associated
                 with tunneled ECRTP.  It includes the per-payload
                 overhead of ECRTP and PPP, timestamp update overhead,
                 and IPID update overhead.

3.3.1.  Voice Bandwidth Calculation Example

   To create an example for a voice application using the above
   formulas, we will assume the following usage scenario.  Compressed
   voice streams using G.729 compression with a 20 millisecond
   packetization period.  In this scenario, VAD is enabled and the
   average talk spurt length is 1500 milliseconds.  The IPID field is
   changing randomly between payloads of streams.  There is enough
   traffic in the tunnel to allow 3 multiplexed payloads.  The following
   values apply:

        SAMPLE-PERIOD      = 20 milliseconds
        TRANSMIT-LENGTH    = 1500 milliseconds
        IPID-RATIO         = 1
        PAYLOAD-SIZE       = 20 octets
        MUX-SIZE           = 3

   For this example, per call bandwidth is 16.4 kbits/sec.  Classical
   CRTP over a single HDLC link using the same factors as above yields
   12.4 kbits/sec.




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   The effect of IPID can have a large effect on per call bandwidth.  If
   the above example is recalculated using an IPID-RATIO of 0, then the
   per call bandwidth is reduced to 13.8 kbits/sec.  Classical CRTP over
   a single HDLC link, using these same factors, yields 11.2 kbits/call.

3.3.2.  Voice Bandwidth Comparison Table

   The bandwidth values are as follows when using 5 simultaneous calls,
   no voice activity detection (VAD), G.729 with 20ms packetization
   interval, and not considering RTCP overhead:

       Normal VoIP over PPP:            124 kbps
       with classical CRTP on a link:    50 kbps (savings: 59%)
       with TCRTP over PPP:              62 kbps (savings: 50%)
       with TCRTP over AAL5:             85 kbps (savings: 31%)

3.3.3.  Video Bandwidth Calculation Example

   Since TCRTP can be used to save bandwidth on any type of RTP
   encapsulated flow, it can be used to save bandwidth for video
   applications.  This section documents an example of TCRTP-based
   bandwidth savings for MPEG-2 encoded video.

   To create an example for a video application using the above
   formulas, we will assume the following usage scenario.  RTP
   encapsulation of MPEG System and Transport Streams is performed as
   described in RFC 2250.  Frames for MPEG-2 encoded video are sent
   continuously, so the TRANSMIT-LENGTH variable in the bandwidth
   formula is essentially infinite.  The IPID field is changing randomly
   between payloads of streams.  There is enough traffic in the tunnel
   to allow 3 multiplexed payloads.  The following values apply:

        SAMPLE-PERIOD      = 2.8 milliseconds
        TRANSMIT-LENGTH    = infinite
        IPID-RATIO         = 1
        PAYLOAD-SIZE       = 1316 octets
        MUX-SIZE           = 3

   For this example, per flow bandwidth is 3.8 Mbits/sec.  MPEG video
   with no header compression, using the same factors as above, yields
   3.9 Mbits/sec.  While TCRTP does provide some bandwidth savings for
   video, the ratio of transmission headers to payload is so small that
   the bandwidth savings are insignificant.








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3.3.4.  TCRTP over ATM

   IP transport over AAL5 causes a quantizing effect on bandwidth
   utilization due to the packets always being multiples of ATM cells.

   For example, the payload size for G.729 using 10 millisecond
   packetization intervals is 10 octets.  This is much smaller than the
   payload size of an ATM cell (48 octets).  When classical CRTP [CRTP]
   is used on a link-by-link basis, the IP overhead to payload ratio is
   quite good.  However, AAL5 encapsulation and its cell padding always
   force the minimum payload size to be one ATM cell, which results in
   poor bandwidth utilization.

   Instead of wasting this padding, the multiplexing of TCRTP allows
   this previously wasted space in the ATM cell to contain useful data.
   This is one of the main reasons why multiplexing has such a large
   effect on bandwidth utilization with Voice over IP over ATM.

   This multiplexing efficiency of TCRTP is similar to AAL2 sub-cell
   multiplexing described in [I.363.2].  Unlike AAL2 sub-cell
   multiplexing, however, TCRTP's multiplexing efficiency isn't limited
   to only ATM networks.

3.3.5.  TCRTP over Non-ATM Networks

   When TCRTP is used with other layer 2 encapsulations that do not have
   a minimum PDU size, the benefit of multiplexing is not as great.

   Depending upon the exact overhead of the layer 2 encapsulation, the
   benefit of multiplexing might be slightly better or worse than link-
   by-link CRTP header compression.  The per-payload overhead of CRTP
   tunneling is either 4 or 6 octets.  If classical CRTP plus layer 2
   overhead is greater than this amount, TCRTP multiplexing will consume
   less bandwidth than classical CRTP when the outer IP header is
   amortized over a large number of payloads.

   The payload breakeven point can be determined by the following
   formula:

     POV-L2 * MUX-SIZE >= POV-L2 + POV-TUNNEL + POV-PPPMUX + SOV-PPPMUX
          * MUX-SIZE

   Where:

     POV-L2, unit: octet.  Layer 2 packet overhead: 5 octets for HDLC
          encapsulation





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     POV-TUNNEL, unit: octet.  Packet overhead due to tunneling: 24
          octets IP header and L2TPv3 header

     POV-PPPMUX, unit: octet.  Packet overhead for the multiplexed PPP
          protocol ID: 1 octet

     SOV-PPPMUX, unit: octet.  Per-payload overhead of PPPMUX, which is
          comprised of the payload length field and the ECRTP protocol
          ID.  The value of SOV-PPPMUX is typically 1, 2, or 3.

   If using HDLC as the layer 2 protocol, the breakeven point (using the
   above formula) is when MUX-SIZE = 7.  Thus 7 voice or video flows
   need to be multiplexed to make TCRTP as bandwidth-efficient as link-
   by-link CRTP compression.

4.  Example Implementation of TCRTP

   This section describes an example implementation of TCRTP.
   Implementations of TCRTP may be done in many ways as long as the
   requirements of the associated RFCs are met.

   Here is the path an RTP packet takes in this implementation:





























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         +-------------------------------+             ^
         |          Application          |             |
         +-------------------------------+             |
         |              RTP              |             |
         +-------------------------------+        Application and
         |              UDP              |            IP stack
         +-------------------------------+             |
         |              IP               |             |
         +-------------------------------+             V
                         |
                         |  IP forwarding
                         |
         +-------------------------------+             ^
         |             ECRTP             |             |
         +-------------------------------+             |
         |            PPPMUX             |             |
         +-------------------------------+          Tunnel
         |             PPP               |         Interface
         +-------------------------------+             |
         |             L2TP              |             |
         +-------------------------------+             |
         |              IP               |             |
         +-------------------------------+             V
                         |
                         |  IP forwarding
                         |
         +-------------------------------+             ^
         |            Layer 2            |             |
         +-------------------------------+          Physical
         |            Physical           |          Interface
         +-------------------------------+             V

   A protocol stack is configured to create an L2TP tunnel interface to
   a destination host.  The tunnel is configured to negotiate the PPP
   connection (using NCP IPCP) with ECRTP header compression and PPPMUX.
   IP forwarding is configured to route RTP packets to this tunnel.  The
   destination UDP port number could distinguish RTP packets from non-
   RTP packets.

   The transmitting application gathers the RTP data from one source,
   and formats an RTP packet.  Lower level application layers add UDP
   and IP headers to form a complete IP packet.

   The RTP packets are routed to the tunnel interface where headers are
   compressed, payloads are multiplexed, and then the packets are
   tunneled to the destination host.





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   The operation of the receiving node is the same as the transmitting
   node in reverse.

4.1.  Suggested PPP and L2TP Negotiation for TCRTP

   This section describes the necessary PPP and LT2P negotiations
   necessary for establishing a PPP connection and L2TP tunnel with L2TP
   header compression.  The negotiation is between two peers: Peer1 and
   Peer2.

4.2.  PPP Negotiation TCRTP

   The Point-to-Point Protocol is described in [PPP].

4.2.1.  LCP Negotiation

   Link Control Processing (LCP) is described in [PPP].

4.2.1.1.  Link Establishment

              Peer1                       Peer2
              -----                       -----
     Configure-Request (no options) ->
                                     <- Configure-Ack
                                     <- Configure-Request (no options)
     Configure-Ack                  ->

4.2.1.2.  Link Tear Down

        Terminate-Request              ->
                                        <- Terminate-Ack




















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4.2.2.  IPCP Negotiation

   The protocol exchange here is described in [IPHCOMP], [PPP], and
   [ECRTP].

              Peer1                       Peer2
              -----                       -----
     Configure-Request              ->
       Options:
       IP-Compression-Protocol
         Use protocol 0x61
         and sub-parameters
         as described in
         [IPCP-HC] and [ECRTP]
                                     <- Configure-Ack
                                     <- Configure-Request
                                          Options:
                                          IP-Compression-Protocol
                                            Use protocol 0x61
                                            and sub-parameters
                                            as described in
                                            [IPCP-HC] and [ECRTP]
     Configure-Ack                  ->




























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4.3.  L2TP Negotiation

   L2TP is described in [L2TPv3].

4.3.1.  Tunnel Establishment

              Peer1                       Peer2
              -----                       -----
     SCCRQ                          ->
       Mandatory AVP's:
       Message Type
       Protocol Version
       Host Name
       Framing Capabilities
       Assigned Tunnel ID
                                     <- SCCRP
                                          Mandatory AVP's:
                                          Message Type
                                          Protocol Version
                                          Host Name
                                          Framing Capabilities
                                          Assigned Tunnel ID
     SCCCN                          ->
     Mandatory AVP's:
       Message Type
                                     <- ZLB

4.3.2.  Session Establishment

              Peer1                       Peer2
              -----                       -----
     ICRQ                           ->
       Mandatory AVP's:
       Message Type
       Assigned Session ID
       Call Serial Number
                                         <- ICRP
                                          Mandatory AVP's:
                                          Message Type
                                          Assigned Session ID
     ICCN                           ->
       Mandatory AVP's:
       Message Type
       Tx (Connect Speed)
       Framing Type
                                     <- ZLB





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4.3.3.  Tunnel Tear Down

              Peer1                       Peer2
              -----                       -----
     StopCCN                        ->
       Mandatory AVP's:
       Message Type
       Assigned Tunnel ID
       Result Code
                                     <- ZLB

5.  Security Considerations

   This document describes a method for combining several existing
   protocols that implement compression, multiplexing, and tunneling of
   RTP streams.  Attacks on the component technologies of TCRTP include
   attacks on RTP/RTCP headers and payloads carried within a TCRTP
   session, attacks on the compressed headers, attacks on the
   multiplexing layer, or attacks on the tunneling negotiation or
   transport.  The security issues associated individually with each of
   those component technologies are addressed in their respective
   specifications, [ECRTP], [PPP-MUX], [L2TPv3], along with the security
   considerations for RTP itself [RTP].

   However, there may be additional security considerations arising from
   the use of these component technologies together.  For example, there
   may be an increased risk of unintended misdelivery of packets from
   one stream in the multiplex to another due to a protocol malfunction
   or data error because the addressing information is more condensed.
   This is particularly true if the tunnel is transmitted over a link-
   layer protocol that allows delivery of packets containing bit errors,
   in combination with a tunnel transport layer option that does not
   checksum all of the payload.

   The opportunity for malicious misdirection may be increased, relative
   to that for a single RTP stream transported by itself, because
   addressing information must be unencrypted for the header compression
   and multiplexing layers to function.

   The primary defense against misdelivery is to make the data unusable
   to unintended recipients through cryptographic techniques.  The basic
   method for encryption provided in the RTP specification [RTP] is not
   suitable because it encrypts the RTP and RTCP headers along with the
   payload.  However, the RTP specification also allows alternative
   approaches to be defined in separate profile or payload format
   specifications wherein only the payload portion of the packet would
   be encrypted; therefore, header compression may be applied to the
   encrypted packets.  One such profile, [SRTP], provides more



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   sophisticated and complete methods for encryption and message
   authentication than the basic approach in [RTP].  Additional methods
   may be developed in the future.  Appropriate cryptographic protection
   should be incorporated into all TCRTP applications.

6.  Acknowledgements

   The authors would like to thank the authors of RFC 2508, Stephen
   Casner and Van Jacobson, and the authors of RFC 2507, Mikael
   Degermark, Bjorn Nordgren, and Stephen Pink.

   The authors would also like to thank Dana Blair, Alex Tweedley, Paddy
   Ruddy, Francois Le Faucheur, Tim Gleeson, Matt Madison, Hussein
   Salama, Mallik Tatipamula, Mike Thomas, Mark Townsley, Andrew
   Valencia, Herb Wildfeuer, J. Martin Borden, John Geevarghese, and
   Shoou Yiu.

7.  References

7.1.  Normative References

   [PPP-MUX] Pazhyannur, R., Ali, I., and C. Fox, "PPP Multiplexing",
             RFC 3153, August 2001.

   [ECRTP]   Koren, T., Casner, S., Geevarghese, J., Thompson, B., and
             P. Ruddy, "Enhanced Compressed RTP (CRTP) for Links with
             High Delay, Packet Loss and Reordering", RFC 3545, July
             2003.

   [CRTP]    Casner, S. and V. Jacobson, "Compressing IP/UDP/RTP Headers
             for Low-Speed Serial Links", RFC 2508, February 1999.

   [IPHCOMP] Degermark, M., Nordgren, B., and S. Pink, "IP Header
             Compression", RFC 2507, February 1999.

   [IPCP-HC] Engan, M., Casner, S., Bormann, C., and T. Koren, "IP
             Header Compression over PPP", RFC 3544, July 2003.

   [RTP]     Schulzrinne, H.,  Casner, S., Frederick, R., and V.
             Jacobson, "RTP: A Transport Protocol for Real-Time
             Applications", STD 64, RFC 3550, July 2003.

   [L2TPv3]  Lau, J., Townsley, M., and I. Goyret, "Layer Two Tunneling
             Protocol - Version 3 (L2TPv3)", RFC 3931, March 2005.

   [I.363.2] ITU-T, "B-ISDN ATM Adaptation layer specification: Type 2
             AAL", I.363.2, September 1997.




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   [EF-PHB]  Davie, B., Charny, A., Bennet, J.C., Benson, K., Le Boudec,
             J., Courtney, W., Davari, S., Firoiu, V., and D. Stiliadis,
             "An Expedited Forwarding PHB (Per-Hop Behavior)", RFC 3246,
             March 2002.

   [PPP]     Simpson, W., "The Point-to-Point Protocol (PPP)", STD 51,
             RFC 1661, July 1994.

7.2.  Informative References

   [SRTP]    Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K.
             Norrman, "The Secure Real-time Transport Protocol (SRTP)",
             RFC 3711, March 2004.

   [REORDER] G. Pelletier, L. Jonsson, K. Sandlund, "RObust Header
             Compression (ROHC): ROHC over Channels that can Reorder
             Packets", Work in Progress, June 2004.

   [ROHC]    Bormann, C., Burmeister, C., Degermark, M., Fukushima, H.,
             Hannu, H., Jonsson, L-E., Hakenberg, R., Koren, T., Le, K.,
             Liu, Z., Martensson, A., Miyazaki, A., Svanbro, K., Wiebke,
             T., Yoshimura, T., and H. Zheng, "RObust Header Compression
             (ROHC): Framework and four profiles: RTP, UDP, ESP, and
             uncompressed ", RFC 3095, July 2001.



























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Authors' Addresses

   Bruce Thompson
   170 West Tasman Drive
   San Jose, CA  95134-1706
   United States of America

   Phone: +1 408 527 0446
   EMail: brucet@cisco.com


   Tmima Koren
   170 West Tasman Drive
   San Jose, CA  95134-1706
   United States of America

   Phone: +1 408 527 6169
   EMail: tmima@cisco.com


   Dan Wing
   170 West Tasman Drive
   San Jose, CA  95134-1706
   United States of America

   EMail: dwing@cisco.com

























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Full Copyright Statement

   Copyright (C) The Internet Society (2005).

   This document is subject to the rights, licenses and restrictions
   contained in BCP 78, and except as set forth therein, the authors
   retain all their rights.

   This document and the information contained herein are provided on an
   "AS IS" basis and THE CONTRIBUTOR, THE ORGANIZATION HE/SHE REPRESENTS
   OR IS SPONSORED BY (IF ANY), THE INTERNET SOCIETY AND THE INTERNET
   ENGINEERING TASK FORCE DISCLAIM ALL WARRANTIES, EXPRESS OR IMPLIED,
   INCLUDING BUT NOT LIMITED TO ANY WARRANTY THAT THE USE OF THE
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   WARRANTIES OF MERCHANTABILITY OR FITNESS FOR A PARTICULAR PURPOSE.

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   The IETF invites any interested party to bring to its attention any
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Acknowledgement

   Funding for the RFC Editor function is currently provided by the
   Internet Society.







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