One document matched: draft-hiller-rohc-gehco-00.txt



 Robust Header Compression                                     Tom Hiller
 Internet Draft                                               Pete McCann
 Document: draft-hiller-rohc-gehco-00.txt             Lucent Technologies
 Category: Standards Track                                    August 2000


                 Good Enough Header COmpression (GEHCO)


 Status of this Memo

    This document is an Internet-Draft and is in full conformance
    withall provisions of Section 10 of RFC2026 [1].

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

    The list of current Internet-Drafts can be accessed at
    http://www.ietf.org/ietf/1id-abstracts.txt

    The list of Internet-Draft Shadow Directories can be accessed at
    http://www.ietf.org/shadow.html.

    For potential updates to the above required-text see:
    http://www.ietf.org/ietf/1id-guidelines.txt

 1.      Abstract

    The Robust Header Compression Working Group has embarked upon the
    development and standardization of header compression schemes that
    perform well over links with high error rates and long round-trip
    times. The goal is that the schemes must perform well for cellular
    links built using technologies such as WCDMA, EDGE, and CDMA-2000.
    One way the group plans to achieve this is by maintaining
    connections with other standardization organizations developing
    cellular technology for IP, such as 3GPP and 3GPP-2, to ensure that
    its output fulfills their requirements and will be put to good use.
    Of particular importance to the 3GPP2 community is the spectral
    efficiency of IP/UDP/RTP header compression for the voice over IP
    application to mobiles. Currently, the ROHC Working Group is
    focusing on compression schemes where the uncompressed header is
    identical to the original header seen by the compressor. This may
    be categorized as transparent compression. Transparent compression
    results in additional header bytes being transported over-the-air.
    3GPP2 carriers have indicated that no additional bytes may be
    transported over-the-air for voice over IP, that is, that voice
    over IP must have spectral efficiency comparable to legacy circuit
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    transport over-the-air. This draft proposes a specific scheme, Good
    Enough Header Compression (GEHCO) that does not exactly reproduce
    the IP/UDP/RTP header, but is otherwise "good enough" for voice
    over IP applications over cellular links found in 3GPP2 3G wireless
    networks.

    This particular scheme is applicable to certain cellular links that
    synchronously transport vocoded or other multimedia payloads so
    that the compressor may be certain that the compressed frame will
    be delivered with a fixed and predictable delay. One example of
    such a cellular link is the cdma2000 air interface with cdma2000
    vocoders. A summary of the cdma2000 link characteristics is
    provided herein.



 2.      Conventions used in this document

    The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
    "SHOULD", "SHOULD NOT", "RECOMMENDED",  "MAY", and "OPTIONAL" in
    this document are to be interpreted as described in RFC-2119 [2].

 3.      Introduction

    With the development of SIP [3], it is likely that new services
    will be created that require support of SIP in mobile endpoints. In
    concert, a need will then arise to support the transport of RTP
    packets to and from an endpoint.  Voice sessions transmit short
    frames of data, usually on a continuous basis according to known
    activity factors. The frequency and length of the voice packets
    requires compression of RTP packet overhead in order to justify
    such service relative to not using VOIP (or IP based multimedia) at
    all. Such compression schemes rely on a lower data link layer for
    framing (e.g. to provide a length) and while they can reduce the
    impact of header overhead, they still require some additional
    bandwidth compared to the existing circuit voice channel to carry
    the compressed header bytes and feedback packets.

    The Robust Header Compression Working Group has embarked upon the
    development and standardization of header compression schemes that
    perform well over links with high error rates and long round-trip
    times. The goal is that the schemes must perform well for cellular
    links built using technologies such as WCDMA, EDGE, and cdma2000.
    The ROHC group plans to achieve its goals by maintaining
    connections with other standardization organizations developing
    cellular technology for IP, such as 3GPP and 3GPP2.  Of particular
    importance to the 3GPP2 community is the spectral efficiency of
    IP/UDP/RTP header compression for voice-over-IP application to
    mobiles.


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    A number of RTP compression algorithms have emerged recently in the
    IETF. A few are:

         * Enhanced CRTP [3]
         * ROCCO [4]
         * ACE [5]

    This list is not exhaustive. These algorithms are able to reduce
    the IP/UDP/RTP overhead to one or two bytes. They are also able to
    tolerate some loss of packets and still maintain local
    decompression state without complete retransmission of a new
    IP/UDP/RTP header. In addition to the one or two bytes of header
    overhead some amount of bandwidth is necessary for feedback between
    decompressor and compressor to retransmit a partial or complete
    header when the local state cannot be repaired.

    Transparent compression results in additional header bytes being
    transported over-the-air. 3GPP2 carriers have indicated that voice
    over IP must have spectral efficiency comparable to legacy circuit
    transport over-the-air in order for them to deploy voice-over-IP to
    mobiles. To achieve such spectral efficiency, IP/UDP/RTP header
    compression must not transport any  additional bytes over-the-air.
    Such a scheme will not exactly reproduce all bits of the IP/UDP/RTP
    header, however, the behavior of the reproduced header is "good
    enough" for voice-over-IP applications over cellular links. Such
    header compression may be categorized as non-transparent header
    compression.

    This draft first discusses the characteristics of the cdma2000 link
    layer that makes GEHCO possible.  It then reviews the impact of
    other proposed IP/UDP/RTP header compression schemes on the
    spectral capacity in the context of low bit rate vocoders.  The
    additional impact of the data link layer such as PPP is also
    considered. .  Finally, the draft discusses a particular approach
    for non-transparent header compression called Good Enough Header
    COmpression (GEHCO).

 4       cdma2000 Link Characteristics

    The cdma2000 link synchronously transports physical frames every
    20ms. The number of bytes in a physical frame varies based on
    signaling negotiation between the user and the radio network as
    well as other criteria, such as backlog (for the forward direction
    of the network to mobile). The physical frames are sent in over-
    the-air connections. Currently, six such connections may exist
    simultaneously. The connections feature two modes, one in which the
    physical frames are subject to retransmission, and another in which
    they are not. The over-the-air connections are referred to as Radio
    Link Protocol (RLP) instances. The mode with retransmission serves
    to improve the error rate and thereby improve the throughput of
    TCP. It has variable delay. The one without retransmission behaves
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    synchronously and each 20ms frame has a fixed  number of bits such
    that the physical frame will be delivered in 20ms. There are four
    types of physical frames: full rate (172 bits), half rate (80
    bits), quarter rate (40 bits) and eighth rate (16 bits). One reason
    for these different "rate" frames is variable vocoding. Depending
    on speech activity and patterns, the vocoder puts out a number of
    bytes that exactly fits one of these four physical frame sizes.

    The over-the-air connections have priority. Voice payloads are sent
    in over-the-air connections with high priority in the mode without
    any retransmission. It is possible to say with certainty for these
    kinds of frames that if a physical frame is transmitted it will be
    delivered 20ms later, if it is received at all. We consider the
    frame to not be "received" if it has uncorrectable errors.

 5.      Transparent Compression and Spectral Efficiency

    3GPP2 vendors and service providers are developing a vocoder
    referred to as Selectable-Mode-Vocoder (SMV) which outputs encoded
    voice frames every 20ms. Like other low bit rate vocoders, this one
    does not output constant bit rate. Table 1 [7] shows the
    percentages of times that the vocoder is outputting various
    payloads (measured in terms of numbers of bits in a given payload).
    (Note that the activities sum to 100% so that the vocoder outputs a
    voice frame every 20ms).

         Rate            Activity %      Payload (bits)

         Full            20              172
         Half            20              80
         Quarter         10              40
         Eighth          50              16

            Table 1: Activity of a 3GPP-2 Vocoder

    As can be seen, even one or two bytes of data per vocoded frame
    represents a considerable taxation on the spectral capacity of the
    system.

    In the current 3GPP2 wireless packet data architecture, byte
    oriented HDLC framing provides packet delimitation. 3GPP2 selected
    byte oriented HDLC [8] and PPP [9] due to its wide deployment in
    operating systems commonly found in laptops and other portable
    devices, as well as its ability to flexibly negotiate data link
    configurations.  PPP overhead may be compressed by having the
    mobile negotiate the address/control/protocol fields to one byte,
    and negotiate no CRC via RFC 1570 [10]. This reduces PPP overhead
    to two bytes of overhead per vocoder frame, that is one protocol ID
    byte and one flag between frames.


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    Assuming one overhead byte for IP/UDP/RTP transparent compression
    schemes, and two bytes for PPP, the compressed overhead header
    bytes will exceed the payload 50% of the time. Even at peak vocoder
    output, the compressed headers amount to a 16% spectral tax [11].
    If two bytes of IP/UDP/RTP compression control bytes are needed,
    the overhead is even worse. This quick analysis assumes that there
    is no need for IP packets to be synchronized with the physical
    frame.  It also ignores payload expansion due to HDLC escaping.

    Since the taxation is a result of the shortness of the vocoded
    voice frame, one might try to aggregate voice frames into a single
    IP/UDP/RTP packet. If more vocoded frames could be combined within
    one RTP packet, then loss of spectrum capacity would
    correspondingly decrease.  Given that many of the packets are very
    short, aggregating more vocoder frames (20ms frames) does not
    necessarily create PPP frames larger than the basic rate, 172-bit
    payload.  This approach does imply additional delay end to end. The
    number of voice frames aggregated into a VOIP packet should be
    adjusted dynamically based on the coding rate. Dynamic aggregation
    requires a scheme for delimiting the vocoded frames within the RTP
    payload which might consume three to five bits per packet. This
    approach also requires more study to determine if users can, for
    example, tolerate the added 20ms delay incurred by aggregating two
    short vocoder frames into one PPP frame. Such an analysis should
    probably consider the case when there is an IP multimedia
    conference bridge involved. For this case, each direction then
    requires two voice encodes and decodes so that the bridge can sum
    the speech from the call participants, which gives rise to
    additional delay.

 6.      Non-Transparent Header Compression

    A different method to improve spectral efficiency may be to simply
    not transmit the RTP or PPP headers with the vocoded frames, but
    instead construct IP/UDP/RTP headers at the decompressor based on
    state context information forwarded to the decompressor by the
    compressor. The IP/UDP/RTP header reconstructed by the decompressor
    does not match the original IP/UDP/RTP sent packet bit for bit.
    Some fields such as RTP sequence counts and UDP CRC may be altered
    from an end-to-end point of view. Non-transparent header
    compression results in no loss of spectrum capacity (except for a
    couple of control packets involved in arranging the non-transparent
    compression).  This form of compression has also been called
    _header stripping and regeneration._

    The compressor removes headers before transmitting and the
    decompressor creates IP/UDP/RTP header information including RTP
    sequence counts. Static header information such as IP address and
    UDP port will exactly match those sent by the transmitter.  Other
    header information such as RTP sequence counts may not, hence the
    category _non transparent compression".
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    There are a variety of methods to implement non-transparent header
    compression.  This contribution will focus upon header compression
    using link (PPP) control mechanisms we refer to as _Good Enough
    Header COmpression (GEHCO)". Compressed voice (e.g. SMV) is sent
    with no headers (i.e. no PPP, IP, UDP, or RTP) over-the-air in
    physical frames and the decompressor regenerates all IP/UDP/RTP
    headers. This includes RTP sequence counts and UDP checksum. The
    compressor's job is simply to delete the various IP/UDP/RTP headers
    and pass the compressed voice to air framing hardware for
    transmission over the air interface. The decompressor's job is to
    wrap the compressed voice in an IP/UDP/RTP packet, creating
    sequence numbers and checksums as necessary.

    Depending on the vocoder used, the decompressor may wrap and
    forward erred data in RTP packets.  Regardless, the decompressor
    increments a sequence count for each received physical frame.  The
    receiver may then use the payload bits or the sequence numbers to
    detect missing packets and possibly mask the error.

    The compressor and decompressor communicate via an over-the-air
    connection established to transport the voice frames. As explained
    above, the cdma2000 link can support an over-the air connection
    that transports payloads synchronously in physical frames every
    20ms. The type of over-the-air connection for voice has priority
    and the vocoder output will dictate the size of the physical frame.
    This connection inherently provides framing for the vocoder voice
    frames, that is,we are not using a transparent data link layer like
    byte HDLC for framing. This connection is separate from the over-
    the-air connection over which usual PPP frames flow. That
    connection has lower priority and the physical frames on that
    connection will be subject to retransmission. There will be one RTP
    flow per over-the-air connection, since otherwise it is impossible
    to distinguish flows given no headers exist with the payloads. The
    physical basis and control (e.g. establishment and release) of
    connections for the usual PPP frames and for the voice frames is
    outside the scope of this document.

    In Section 7, we refer to the over-the-air connection that
    transports voice payloads as the "vocoder connection" and the one
    that transports PPP frames as the "PPP connection".

    In the network to mobile station direction (forward direction), the
    RTP packets received from the IP network will experience variable
    delays. The gateway (PPP termination point) will buffer some
    packets to compensate for jitter in the IP network. Because the
    over-the air connection that transports the RTP payloads is
    synchronous, does not feature retransmission, and has high
    priority, the gateway can simply play out the payloads of the RTP
    packets over the vocoder connection. These physical frames will be
    delivered exactly 20ms later. The decompressor in the mobile then
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    will assign timestamps that increment by 20ms each frame and
    deliver the packets to the IP layer in the mobile station.

    The loss of RTP packets in the network, or possibly significant
    delays due to transient congestion, could cause buffer under-run at
    the gateway.  In this case the compressor has no frame to output on
    the over-the-air link used to carry the voice frames.  One solution
    is to have the compressor output a "NO DATA" frame over-the-air to
    the decompressor. This would be part of the air interface
    definition and would be analogous to an approach pursued in 3GPP
    [12] for this same purpose.

    The decompressed IP/UDP/RTP packets are not bit for bit identical
    end-to-end but there is no impact to the service. The term _good
    enough_ means that the compression/decompression scheme is adequate
    for transport of voice sessions even though the headers are not
    reconstructed exactly. For example, RTP sequence counts sent by the
    mobile can differ from those sent by the network to other endpoints
    involved in a given multimedia call if the receiver does not start
    counting physical frames precisely at the same time the transmitter
    begins to send physical frames.   While different RTP sequence
    counts and time stamps will imply different UDP checksums, the
    receiver will be able to make use of all generated RTP sequence
    counts and time stamps.

    The IP Identification field is generated from information supplied
    to the decompressor by the compressor, and with proper engineering
    of the sender's IP stack it will maintain the uniqueness property
    that is important for proper functioning of IP fragmentation, ICMP
    processing, and multi-path routing.  However, even without such
    engineering, duplicate Identification fields will be rare and those
    that cause problems even more rare.  This is especially true since
    voice RTP packets will be quite small and should never be
    fragmented.




 7.      Good Enough Header Compression (GEHCO)

    The next two sections consider the identification of an end-to-end
    bit flow between link (e.g. PPP) entities and the actual GEHCO
    procedures.  The establishment of the over-the-air connection is
    properly not a GEHCO function and is dependent on the particular
    link layer protocol in use. GEHCO only requires an abstraction that
    allows link entities to determine the identity of a bit flow.

 7.1 Link Layer Connections

    In order to make use of a connection that carries compressed voice
    or other multimedia without any link or higher layer headers
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    between mobile station and a PPP peer in the network, the mobile
    station and network PPP peer must agree upon a connection
    identifier. This is so that both ends are able to identify a given
    bit flow with a given RTP flow.

    We assume that the network and mobile will use separate over-the-
    air connections for purposes of transporting PPP, and for
    transporting compressed voice. Typically in a cellular environment,
    the connection for the PPP frames will feature improved error rates
    via retransmission at the physical layer. However, due to latency
    requirements, the connection that transports the vocoded frames
    will not feature such retransmission.

    In the mobile station, the PPP implementation must have separate
    service access points for each connection (i.e. PPP frame
    connection and vocoder frame connection).  This will allow the PPP
    implementation to direct compressed IP/UDP/RTP packets to the
    vocoder frame connection and normal data traffic to the PPP frame
    connection.  In a relay model phone connected to a laptop, these
    connections are accessed via multiple access points, possibly
    supported by advanced modem interface protocols such as V.80.
    Similarly, the PPP implementation in the network side PPP peer must
    have separate service access points for each such connection.

 7.2 Compression and Decompression Operation

    This section describes how the mobile station and network (i.e. the
    radio gateway) agree to use GEHCO and establish auxiliary
    connections as described above. The mobile station and network must
    negotiate GEHCO as part of PPP establishment (e.g. as an extension
    to PPP's IPCP or IPv6CP). This negotiation would be carried out
    over the PPP frame connection.  Once negotiated, the two PPP
    implementations could dynamically establish and delete compression
    contexts for VOIP sessions.  These contexts could be sent as "full
    headers" [13] over PPP frame connection, and the CID (i.e.
    Connection ID from [13]) should be set equal to some identifier
    that identifies the over-the-air connection that will be used to
    transport the compressed packets for the session.  The full headers
    need to be sent using a new PPP Protocol Type to distinguish them
    from ordinary (transparent) header compression that may be taking
    place over the same PPP session.  Below we refer to this new
    protocol type as GEHCO_FULL_HEADER.  No additional compression
    (i.e., PPP payload compression) or transformation would be
    performed on the compressed RTP packets.  The vocoder frame
    connection would carry voice frames exactly as for normal circuit
    voice.  This implies that the vocoder has a real-time relationship
    to the physical frame rate just as in circuit voice, and the mobile
    host must be properly engineered to deliver voice traffic to the
    physical layer at the physical frame rate.  For now we assume that
    this will be in the form of IP/UDP/RTP packets delivered to PPP,
    although some architectures may provide a direct link between the
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    vocoder and the physical layer.  Similarly, the network side PPP
    peer must be able to guarantee that compressed voice frames are
    delivered in a way that enables real-time transmission over the
    air.


    The GEHCO compressor and decompressor are part of the PPP layer in
    both the mobile station and network side gateway. The compressor
    maintains a table of active RTP sessions that consists of source
    and destination IP address, source and destination UDP port,
    current RTP sequence number, and connection ID. When the compressor
    receives an IP/UDP/RTP packet from the IP layer, it scans a table
    of entries to find a match based the packet's source and
    destination addresses and ports. If the compressor does not find a
    matching entry, and if an appropriate over-the-air connection
    exists to carry the compressed voice, the compressor sends a PPP
    control frame of type GEHCO_FULL_HEADER to the decompressor over
    the PPP frame connection. If an appropriate over-the-air connection
    does not exist the peer may take steps to create one.  For future
    packets that match, the compressor removes the IP/UDP/RTP headers
    and places the compressed voice in the RTP payload onto the vocoder
    frame connection associated with this RTP flow.

    As explained above, the network side compressor will maintain a
    buffer to compensate for to incoming packet jitter from the IP
    network. The mobile decompressor then creates timestamps per the
    20ms physical frame rate and delivers packets to the IP layer.  For
    the compressor on the mobile side, we do not expect any real jitter
    from the mobile's vocoder since the vocoder is integral to the
    mobile station. If there is a small level of jitter (perhaps due to
    small variable delays in the mobile station) the compressor could
    likewise maintain a small buffer to compensate. At the network side
    the decompressor assigns timestamps that increment very 20ms. If a
    compressor buffer underruns (perhaps due to uncorrectable errors or
    significant delays in the IP network), the compressor outputs a
    radio frame that has "NULL DATA".  The decompressor would not
    produce an RTP frame for such NULL DATA frames.


    The GEHCO_FULL_HEADER message contains a complete IP/UDP/RTP
    packet, including:

         * IP source and destination address
         * Initial RTP sequence number
         * UDP source and destination port
         * Initial IP Identification field
         * CID (Connection ID)

    By way of an example, a convenient and trivial Connection ID for
    the case of cdma2000 is the Radio Link Protocol (RLP) instance.
    However, there is no real constraint other than that the system is
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    able to identify a particular radio connection with a bit flow
    received by a link layer entity on the network side. The
    GEHCO_FULL_HEADER is distinct from the RFC2509 FULL_HEADER to keep
    the space of CIDs used here which correspond to vocoder frame
    connection CIDs separate from RFC2509 CIDs. The decompressor in the
    PPP link layer stores the above information in a similar table. The
    mobile station and network side PPP peer use the CIDs as indices to
    the table.

    The decompressor sends a GEHCO_FULL_HEADER_ACK in response with the
    CID. The decompressor will then access the RTP payload and wrap the
    compressed voice in an IP/UDP/RTP packet and pass it to the IP
    layer for further processing or transport. The initial starting RTP
    sequence number is equal to that in the GEHCO_FULL_HEADER control
    frame, and is incremented by 1 for each decompressed frame. The IP
    address and UDP port are determined from the table.  The
    decompressor also inserts an RTP timestamp according to the value
    of a local clock at the time the frame was received.  Any other
    fields such as the IP Identification field and any Checksums are
    set to appropriate values by the decompressor.

    The initial IP Identification field is determined from the
    GEHCO_FULL_HEADER and is incremented by 1 for each decompressed
    frame.  To ensure uniqueness of IP Identification values, the RTP
    sender should assign IDs in increasing order on a per-flow basis.
    This might mean that the sender allocates a certain range of IDs
    for use by a flow; when the range is exhausted the sender will
    start a new range and the compressor will optionally send a new
    GEHCO_FULL_HEADER with the new start value.  The compressor is
    specifically NOT REQUIRED to do so, however, as duplicate IDs are
    expected to be rare even when the initial range is exhausted.

    While the compressor waits for the GEHCO_FULL_HEADER_ACK, the
    compressor places RTP payloads into the designated physical
    connection. This allows the decompressor to create RTP packets as
    soon as it receives the GEHCO_FULL_HEADER control frame. The
    compressor periodically sends a GEHCO_FULL_HEADER frame until it
    receives a GEHCO_FULL_HEADER_ACK.

    The mobile station or network may reclaim the UDP port and IP
    address for a new RTP flow that happens to use the same IP address
    and UDP port. Reclaiming could happen long after a given call has
    terminated or perhaps even during a call if the SIP control so
    determines to use the same parameters for a different call.  If
    this happens the compressor need only send a new GEHCO_FULL_HEADER
    frame with the new information for the new call.

    The compressor and decompressor physically will have a maximum
    number of table entries.  When the table size is exceeded, the
    oldest table entry is overwritten.

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 7.3 Protocol

    GEHCO uses the Configuration Option of Section 2.1 of RFC 2509 [13]
    and requires a new field for the RTP-compression sub-option of
    Section 2.2 of RFC 2509.  That is, we propose that the RTP-
    compression sub-option be extended with additional bytes indicating
    which RTP compression schemes are in use.

                 0                   1
                 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5
                +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
                |     Type      |    Length     |    Scheme  ...
                +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

             Type
                1

             Length
                Variable

             Scheme
                Sequence of CRTP, ROHC, GEHCO (values TBD)


    Section 3.3.1 of RFC 2508 [14] specifies that a FULL_HEADER packet
    is a PPP frame with a full header and the context ID placed in the
    IP and UDP length field. Two context formats are specified, one an
    8 bit and the other a 16 bit length ID. Section 3.3.1 specifies the
    placement of the Context ID in those length fields.

    The GEHCO_FULL_HEADER carries only one RTP flow in a given
    connection. Therefore, the connection ID we propose above fulfills
    a similar function as the Context ID does in RFC 2508. We propose
    that the connection ID be placed in the IP and UDP fields, that
    there be two such connection ID options, and that the placement of
    the connection ID be the same as that of the Context ID as
    specified in RFC 2508.  GEHCO does not require that a sequence
    number be carried in the GEHCO_FULL_HEADER.

 7.4 RTCP

    RTP has a control protocol that occasionally sends control packets.
    In GEHCO, these would be transported over the PPP frame connection
    in PPP frames. The taxation is negligible because very few packets
    are sent.

 7.5 GEHCO and Cell Phones

    The decompressor was depicted above as recreating RTP packets from
    the compressed voice (or IP multimedia) and passing that packet up
    the IP stack.  In an integrated cell phone, actual creation of
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    these packets may not be required because there would be no need to
    add IP/UDP/RTP headers to pass up an IP stack just to be taken off
    again before delivery to a vocoder.  In a laptop or palmtop, etc.,
    the story is different because of an existing IP stack with
    applications that must be accommodated.
    Cell phones in general will have an IP stack to support the
    integral browser and SIP (and RTCP) protocols. The compressed voice
    need not be placed into RTP format within the phone.

 7.6 Simultaneous Use of Transparent and Non-Transparent Compression

    This section would not apply to VOIP header compression in cell
    phones since non-transparent compression alone will suffice.

    Usually one would expect only one RTP compression scheme to be
    active on a given mobile (i.e. a laptop) at any time. For laptops,
    if it is necessary to allow transparent and non-transparent header
    compression methods to be simultaneously active, then the PPP layer
    would not know whether to use a transparent scheme (like ROCCO,
    ACE, CRTP, etc.) or GEHCO for a given flow.  There are a couple of
    solutions. One would be to for the mobile application to have an
    API to the PPP layer that indicates the type on a per flow basis.
    The mobile would then negotiate with the PPP peer the appropriate
    type of compression to use for the flow. The other approach would
    be for ROHC to agree upon RTP Payload types to be used with non-
    transparent compression.  The mobile and network PPP peers then
    just use this to determine when to use GEHCO.

 8. Security Considerations

    Non transparent compression will be used over radio links that
    feature strong encryption; such encryption is optional and users
    who defer it are more susceptible to attack. The identity of users
    of these radio links will be authenticated via a private key in
    both the radio realm and the IETF based AAA realm. In general, it
    is very difficult in inject frames onto the radio links; as other
    drafts on compression have pointed out, if a hacker is able to
    inject frames onto the radio links, the problems this creates far
    exceed just those associated with compression and decompression.

 9. References

    [1] Bradner, S., "The Internet Standards Process -- Revision 3",
    BCP 9, RFC 2026, October 1996.

    [2] Bradner, S., "Key words for use in RFCs to Indicate Requirement
    Levels", BCP 14, RFC 2119, March 1997.

    [3] RFC 2543, SIP: Session Initiation Protocol, Handley,
    Schulzrinne, Schooler, Rosenberg, March 1999

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    [4] Enhancements to IP/UDP/RTP Header Compression (draft-koren-avt-
    crtp-enhance-01.txt), Koren, Casner, Ruddy, Thompson, Tweedly,
    Geevarghese, March, 2000 (IETF draft)

    [5] Robust Checksum-based header COmpression (ROCCO) (draft-
    jonsson-robust-hc-04.txt), Lars-Erik Jonsson, Mikael Degermark,
    Hans Hannu,Krister Svanbro, March 10, 2000 (IETF draft)

    [6] Adaptive Header ComprEssion (ACE) for Real-Time Multimedia
    (draft-ace-robust-hc-01.txt), Le, Clanton, Liu, Zheng, March 2000
    (IETF draft)

    [7] Over-the-Air Voice over IP (VoIP) Issues and Recommended Phases
    (ALLIP-20000323-006_QUA_VoIP_Issues), Hsu, Odenwalder, Chen,
    Tiedemann, 3GPP2 All IP, March 2000

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

    [9] RFC 1662, PPP in HDLC-like Framing, W. Simpson, July 1994

    [10] RFC 1570, PPP LCP Extensions, W. Simpson, January 1994

    [11] End-to-End QoS for VOIP and Multimedia in 3GPP2 ALL IP
    Networks (SC-ALLIP-20000426-009_LUC_End-to_End), Tom Hiller, Pete
    McCann, 3GPP2 All IP, April 2000

    [12] Tdoc SMG2 981/00, Voice Over GERAN/UTRAN PS Domain, Nortel
    Networks, ETSI SMG2 #36, see Table 3

    [13] RFC2509, IP Header Compression over PPP, Engan, Casner,
    Bormann, February 1999

    [14] RFC2508, Compressing IP/UDP/RTP Headers for Low-Speed Serial
    Links, Casner, Jacobson, February 1999


 10.  Acknowledgments

    The author wish to acknowledge the input of Qualcomm's Ray Hsu et
    al who initially identified the spectral tax associated with
    IP/UDP/RTP transparent compression schemes, and Mark Lipford of
    SPCS and Mark Munson of Verizon for their business insights on
    spectral capacity, and encouragement to pursue non transparent
    compression.

 11. Author's Addresses

    Tom Hiller
    Lucent Technologies
    263 Shuman Drive
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    Naperville, IL. USA 60137
    Phone: 630-979-7673
    Email: tom.hiller@lucent.com

    Pete McCann
    Lucent Technologies
    263 Shuman Drive
    Naperville, IL. USA 60137
    Phone: 630-713-9359
    Email: mccap@lucent.com









































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PAFTECH AB 2003-20262026-04-22 15:18:23