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>
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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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