Internet DRAFT - draft-ietf-sigtran-signalling-over-sctp-applicability
draft-ietf-sigtran-signalling-over-sctp-applicability
INTERNET-DRAFT L. Coene
Internet Engineering Task Force M. Tuexen
Issued: December 2000 G. Verwimp
Expires: June 2001 Siemens
J. Loughney
Nokia
R.R. Stewart
Cisco
Qiaobing Xie
Motorola
M.C. Belinchon
I. Rytina
Ericsson
L. Ong
Nortel Networks
Telephony Signalling Transport over SCTP applicability statement
<draft-ietf-sigtran-signalling-over-sctp-applicability-02.txt>
Status of this Memo
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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 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
1 INTRODUCTION
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-
vice.
1.2 Overview
SCTP provides a general purpose, reliable transport between two end-
points.
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
architecture:
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
arrival.
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
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 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.
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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: lode.coene@siemens.atea.be
Atealaan 34
B-2200 Herentals
Belgium
John Loughney Phone: +358-9-43761
Nokia Research Center EMail: john.loughney@nokia.com
Itamerenkatu 11-13
FIN-00180 Helsinki
Finland
Michel Tuexen Phone: +49-89-722-47210
Siemens AG EMail: Michael.Tuexen@icn.siemens.de
Hofmannstr. 51
81359 Munich
Germany
Randall R. Stewart Phone: +1-815-477-2127
24 Burning Bush Trail. EMail: rrs@cisco.com
Crystal Lake, IL 60012
USA
Qiaobing Xie Phone: +1-847-632-3028
Motorola, Inc. EMail: qxie1@email.mot.com
1501 W. Shure Drive
Arlington Heights, IL 60004
USA
Maria-Carmen Belinchon Phone: +34-91-339-3535
Ericsson Espana S. A. EMail: Maria.C.Belinchon@ericsson.com
Network Communication Services
Retama 7, 5th floor
Madrid, 28045
Spain
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Ian Rytina EMail:ian.rytina@ericsson.com
Ericsson Australia
37/360 Elizabeth Street
Melbourne, Victoria 3000
Australia
Lyndon Ong Phone: -
Nortel Networks EMail: long@nortelnetworks.com
4401 Great America Parkway
Santa Clara, CA 95054
USA
Gery Verwimp Phone: +32-14-253424
Siemens Atea EMail: gery.verwimp@siemens.atea.be
Atealaan 34
B-2200 Herentals
Belgium
Expires: June 2001
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