INTERNET-DRAFT                                                 L. Coene
Internet Engineering Task Force                               M. Tuexen
Issued:  December 2000                                       G. Verwimp
Expires: June 2001                                              Siemens
                                                            J. Loughney
                                                           R.R. Stewart
                                                           Qiaobing Xie
                                                         M.C. Belinchon
                                                              I. Rytina
                                                                 L. Ong
                                                        Nortel Networks

     Telephony Signalling Transport over SCTP applicability statement

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 months
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   The list of current Internet-Drafts can be accessed at The list of Internet-
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   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 transport over SCTP Applicability state-
   ment ...........................................................   ii
   Chapter 1: Introduction ........................................    2
   Chapter 1.1: Terminology .......................................    2
   Chapter 1.2: Overview ..........................................    3
   Chapter 2: Applicability of Telephony Signalling transport
   using SCTP .....................................................    4
   Chapter 3: Issues for transporting Telephony signalling
   information over SCTP ..........................................    5
   Chapter 3.1: Congestion control ................................    5
   Chapter 3.2: Detection of failures .............................    5
   Chapter 3.2.1: Retransmission TimeOut (RTO) calculation ........    5
   Chapter 3.2.2: Heartbeat .......................................    6
   Chapter 3.2.3: Maximum Number of retransmissions ...............    6
   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 .............................    7
   Chapter 5: References and related work .........................    7
   Chapter 6: Acknowledgments .....................................    8
   Chapter 7: Authors address .....................................    8


     Transport of telephony signalling requires special considerations.
In order to use SCTP, special care must be taken to meet the perfor-
mance, 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

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     association, within which all user messages are delivered in
     sequence except for those submitted to the unordered delivery ser-

1.2 Overview

SCTP provides a general purpose, reliable transport between two end-

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

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     Telephony Signalling transport over IP normally uses the  following

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

        Figure 1.1: Telephony signalling transport protocol stack

     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 indica-
     tions, data operation primitives, ... for a particular
     user/application protocol).

(2)  SCTP, specially configured to meet the telephony application per-
     formance 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

SCTP can be used to provide redundancy and fault tolerance at the tran-
sport 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

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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 poten-
tial congestion collapse of the network.

Some telephony signalling applications may have their own congestion
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 des-
tination 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

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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

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 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

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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 mes-
sage 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 mul-
tihoming 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.

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 Architec-
     ture 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.

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6 Acknowledgments

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:
Atealaan 34
B-2200    Herentals

John Loughney               Phone: +358-9-43761
Nokia Research Center       EMail:
Itamerenkatu 11-13
FIN-00180    Helsinki

Michel Tuexen               Phone: +49-89-722-47210
Siemens AG                  EMail:
Hofmannstr. 51
81359 Munich

Randall R. Stewart          Phone: +1-815-477-2127
24 Burning Bush Trail.      EMail:
Crystal Lake, IL 60012

Qiaobing Xie                Phone: +1-847-632-3028
Motorola, Inc.              EMail:
1501 W. Shure Drive
Arlington Heights, IL 60004

Maria-Carmen Belinchon      Phone: +34-91-339-3535
Ericsson Espana S. A.       EMail:
Network Communication Services
Retama 7, 5th floor
Madrid, 28045

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Ian Rytina        
Ericsson Australia
37/360 Elizabeth Street
Melbourne, Victoria 3000

Lyndon Ong                  Phone: -
Nortel Networks             EMail:
4401 Great America Parkway
Santa Clara, CA 95054

Gery Verwimp                Phone: +32-14-253424
Siemens Atea                EMail:
Atealaan 34
B-2200    Herentals

Expires: June 2001

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