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INTERNET-DRAFT                                             L. Coene(Ed)
Internet Engineering Task Force                                 Siemens
Issued:  January 2002
Expires: July 2002





  Telephony Signalling Transport over SCTP applicability statement
  <draft-ietf-sigtran-signalling-over-sctp-applic-03.txt>


Status of this Memo



    This document is an Internet-Draft and is in full conformance with
    all provisions of Section 10 of RFC2026. Internet-Drafts are working
    documents of the Internet Engineering Task Force (IETF), its areas,
    and its working groups.  Note that other groups may also distribute
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    Internet-Drafts are draft documents valid for a maximum of six
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Abstract


    This document describes the applicability of the Stream Control
    Transmission Protocol (SCTP)[RFC2960] for transport of telephony
    signalling information over IP infrastructure. Special
    considerations for using SCTP to meet the requirements of
    transporting telephony signalling [RFC2719] are discussed.












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                            Table of contents




   Telephony signalling over SCTP Applicability statement .........   ii
   Chapter 1: Introduction ........................................    2
   Chapter 1.1: Terminology .......................................    2
   Chapter 1.2: Contributors ......................................    3
   Chapter 1.3: Overview .........................................     3
   Chapter 2: Applicability of telephony signalling transport
   using SCTP .....................................................    4
   Chapter 3: Issues for transporting Telephony signalling
   information over SCTP ..........................................    4
   Chapter 3.1: Congestion control ................................    4
   Chapter 3.2: Detection of failures .............................    5
   Chapter 3.2.1: Retransmission TimeOut (RTO) calculation ........    5
   Chapter 3.2.2: Heartbeat .......................................    5
   Chapter 3.2.3: Maximum Number of retransmissions ...............    5
   Chapter 3.3:  Shorten end-to-end message delay .................    6
   Chapter 3.4: Bundling considerations ...........................    6
   Chapter 3.5: Stream Usage ......................................    6
   Chapter 4: Security considerations .............................    6
   Chapter 5: References and related work .........................    7
   Chapter 6: Acknowledgments .....................................    7
   Chapter 7: Author's address ....................................    7




1 INTRODUCTION


    Transport of telephony signalling requires special
    considerations. In order to use SCTP, special care must be taken to
    meet the performance, timing and failure management requirements.



1.1 Terminology


    The following terms are commonly identified in related work:


    Association: SCTP connection between two endpoints.

    Stream: A uni-directional logical channel established within an
    association, within which all user messages are delivered in
    sequence except for those submitted to the unordered delivery
    service.



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1.2 Contributors


    The following people contributed to the document: L. Coene(Editor),
    M.  Tuexen, G. Verwimp, J. Loughney, R.R. Stewart, Qiaobing Xie,
    M. Holdrege, M.C. Belinchon, A. Jungmaier, and L. Ong.


1.3 Overview


    SCTP provides a general purpose, reliable transport between two
    endpoints.

    The following functions are provided by SCTP:

    - Reliable Data Transfer

    - Multiple streams to help avoid head-of-line blocking

    - Ordered and unordered data delivery on a per-stream basis

    - Bundling and fragmentation of user data

    - Congestion and flow control

    - Support continuous monitoring of reachability

    - Graceful termination of association

    - Support of multi-homing for added reliability

    - Protection against blind denial-of-service attacks

    - Protection against blind masquerade attacks

    Telephony Signalling transport over IP normally uses the following
    architecture:

                    Telephony Application
                              |
             +------------------------------------+
             |   Signalling Adaptation module     |
             +------------------------------------+
                              |
             +------------------------------------+
             |Stream Control Transmission Protocol|
             |             (SCTP)                 |
             +------------------------------------+
                              |
               Internet Protocol (IPv4/IPv6)

    Figure 1.1: Telephony signalling transport protocol stack


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    The components of the protocol stack are :

    (1) Adaptation modules are used when the telephony application needs
    to preserve an existing primitive interface. (e.g. management
    indications, data operation primitives, ... for a particular
    user/application protocol).

    (2) SCTP, specially configured to meet the telephony application
    performance requirements.

    (3) The standard Internet Protocol.



2  Applicability of Telephony Signalling transport using SCTP


    SCTP can be used as the transport protocol for telephony
    applications.  Message boundaries are preserved during data
    transport and so no message delineation is needed. The user data can
    be delivered by the order of transmission within a stream(in
    sequence delivery) or the order of arrival.

    SCTP can be used to provide redundancy and fault tolerance at the
    transport layer and below. Telephony applications needing this level
    of fault tolerance can make use of SCTP's multi-homing support.

    SCTP can be used for telephony applications where head-of-line
    blocking is a concern. Such an application should use multiple
    streams to provide independent ordering of telephony signalling
    messages.



3 Issues for transporting telephony signalling over SCTP



3.1 Congestion Control


    The basic mechanism of congestion control in SCTP have been
    described in [RFC2960]. SCTP congestion control sometimes conflicts
    with the timing requirements of telephony signalling transport.

    In an engineered network (e.g. a private intranet), in which network
    capacity and maximum traffic is very well understood, some telephony
    signalling applications may choose to relax the congestion control
    rules in order to satisfy the timing requirements.  But this should
    be done without destabilising the network, otherwise this would lead
    to potential congestion collapse of the network.

    Some telephony signalling applications may have their own congestion

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    control and flow control techniques. These techniques may interact
    with the congestion control procedures in SCTP.  Additionally,
    telephony applications may use SCTP stream based flow control
    [SCTPFLOW].



3.2 Detection of failures

    Telephony systems often must achieve high availability in operation.
    For example, they are often required to be able to preserve stable
    calls during a component failure. Therefore error situations at the
    transport layer and below must be detected very fast so that the
    application can take approriate steps to recover and preserve the
    stable calls. This poses special requirements on SCTP to discover
    unreachablility of a destination address or a peer.



3.2.1 Retransmission TimeOut (RTO) calculation

    The SCTP protocol parameter RTO.Min value has a direct impact on the
    calculation of the RTO itself. Some telephony applications want to
    lower the value of the RTO.Min to less than 1 second. This would
    allow the message sender to reach the maximum
    number-of-retransmission threshold faster in the case of network
    failures. However, lowering RTO.Min may have a negative impact on
    network behaviour [ALLMAN99].

    In some rare cases, telephony applications might not want to use the
    exponential timer back-off concept in RTO calculation in order to
    speed up failure detection. The danger of doing this is that, when
    network congestion occurs, not backing off the timer may worsen the
    congestion situation. Therefore, this strategy should never be used
    in public Internet.

    It should be noted that not using delayed SACK will also help faster
    failure detection.



3.2.2 Heartbeat

    For faster detection of (un)availability of idle paths, the
    telephony application may consider lowering the SCTP parameter
    HB.interval. It should be noted this will result in a higher traffic
    load.



3.2.3 Maximum number of retransmissions

    Setting Path.Max.Retrans and Association.Max.Retrans SCTP parameters
    to lower values will speed up both destination address and peer

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    failure detection. However, if these values are set too low, the
    probability of false detections will increase.



3.3 Shorten end-to-end message delay

    Telephony applications often require short end-to-end message
    delays.  The methods described in section 3.2.1 on lowering RTO and
    not using delayed SACK may be considered.



3.4 Bundling considerations

    Bundling small telephony signalling messages at transmission helps
    improve the bandwidth usage efficiency of the network. On the
    downside, bundling may introduce additional delay to some of the
    messages. This should be taken into consideration when end-to-end
    delay is a concern.



3.5 Stream Usage

    Telephony signalling traffic is often composed of multiple,
    independent message sequences. It is highly desirable to transfer
    those independent message sequences in separate SCTP streams. This
    reduces the probability of head-of-line blocking in which the
    retransmission of a lost message affects the delivery of other
    messages not belonging to the same message sequence.



4 Security considerations


    SCTP only tries to increase the availability of a network. SCTP does
    not contain any protocol mechanisms which are directly related to
    user message authentication, integrity and confidentiality
    functions. For such features, it depends on the IPSEC protocols and
    architecture and/or on security features of its user protocols.

    Mechanisms for reducing the risk of blind denial-of-service attacks
    and masquerade attacks are built into SCTP protocol. See RFC2960,
    section 11 for detailed information.

    Currently the IPSEC working group is investigating the support of
    multihoming by IPSEC protocols. At the present time to use IPSEC,
    one must use 2 * N * M security associations if one endpoint uses N
    addresses and the other M addresses.




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5 References and related work


    [RFC2960] Stewart, R. R., Xie, Q., Morneault, K., Sharp, C. , ,
    Schwarzbauer, H. J., Taylor, T., Rytina, I., Kalla, M., Zhang,
    L. and Paxson, V, "Stream Control Transmission Protocol", RFC2960,
    October 2000.


    [RFCOENE] Coene, L., Tuexen, M., Verwimp, G., Loughney, J., Stewart,
    R.  R., Xie, Q., Holdrege, M., Belinchon, M.C., and Jungmayer, A.,
    "Stream Control Transmission Protocol Applicability statement",
    <draft-ietf-sigtran-sctp-applicability-03.txt>, December 2000. Work
    In Progress.


    [RFC2719] Ong, L., Rytina, I., Garcia, M., Schwarzbauer, H., Coene,
    L., Lin, H., Juhasz, I., Holdrege, M., Sharp, C., "Framework
    Architecture for Signalling Transport", RFC2719, October 1999


    [SCTPFLOW] Stewart, R., Ramalho, M., Xie, Q., Conrad, P. and Rose,
    M., "SCTP Stream based flow control", September 2000, Work in
    Progress.


    [ALLMAN99] Allman, M. and Paxson, V., "On Estimating End-to-End
    Network Path Properties", Proc. SIGCOMM'99, 1999.



6 Acknowledgments

    This document was initially developed by a design team consisting of
    Lode Coene, John Loughney, Michel Tuexen, Randall R. Stewart,
    Qiaobing Xie, Matt Holdrege, Maria-Carmen Belinchon, Andreas
    Jungmaier, Gery Verwimp and Lyndon Ong.


    The authors wish to thank Renee Revis, H.J. Schwarzbauer, T. Taylor,
    G.  Sidebottom, K. Morneault, T. George, M. Stillman and many others
    for their invaluable comments.



7 Author's Address


Lode Coene                  Phone: +32-14-252081
Siemens Atea                EMail: lode.coene@siemens.atea.be
Atealaan 34
B-2200    Herentals
Belgium


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Expires: July 2002


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