Internet Engineering Task Force SIP WG
Internet Draft J. Rosenberg
dynamicsoft
H. Schulzrinne
Columbia U.
G. Camarillo
Ericsson
draft-ietf-sip-sctp-04.txt
November 19, 2003
Expires: May, 2004
The Stream Control Transmission Protocol as a
Transport for the Session Initiation Protocol
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
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Abstract
This document specifies a mechanism for usage of SCTP (the Stream
Control Transmission Protocol) as the transport between SIP entities.
SCTP is a new protocol which provides several features that may prove
beneficial for transport between SIP entities which exchange a large
amount of messages, including gateways and proxies. As SIP is
transport independent, support of SCTP is a relatively
straightforward process, nearly identical to support for TCP.
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Table of Contents
1 Introduction ........................................ 3
2 Terminology ......................................... 3
3 Potential Benefits .................................. 3
3.1 Advantages over UDP ................................. 3
3.2 Advantages over TCP ................................. 4
4 Usage of SCTP ....................................... 5
4.1 Mapping of SIP Transactions into Streams ............ 6
5 Locating a SIP server ............................... 7
6 Security Considerations ............................. 7
7 Conclusion .......................................... 7
8 Author's Addresses .................................. 7
9 Normative References ................................ 8
10 Informative References .............................. 8
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1 Introduction
The Stream Control Transmission Protocol (SCTP) [1] has been designed
as a new transport protocol for the Internet (or intranets), at the
same layer as TCP and UDP. SCTP has been designed with the transport
of legacy SS7 signaling messages in mind. We have observed that many
of the features designed to support transport of such signaling are
also useful for the transport of SIP (the Session Initiation
Protocol) [2], which is used to initiate and manage interactive
sessions on the Internet.
SIP itself is transport-independent, and can run over any reliable or
unreliable message or stream transport. However, procedures are only
defined for transport over UDP and TCP. In order to encourage
experimentation and evaluation of the appropriateness of SCTP for SIP
transport, this document defines transport of SIP over SCTP.
Note that this is not a proposal that SCTP be adopted as the primary
or preferred transport for SIP; substantial evaluation of SCTP,
deployment, and experimentation can take place before that happens.
This draft is targeted at encouraging such experimentation by
enabling it in SIP.
2 Terminology
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 [3].
3 Potential Benefits
Coene et. al. present some of the key benefits of SCTP [4]. We
summarize some of these benefits here and analyze how they relate to
SIP (a more detailed analysis can be found in [5].)
3.1 Advantages over UDP
All the advantages that SCTP has over UDP regarding SIP transport are
also shared by TCP. Below there is a list of the general advantages
that a connection-oriented transport protocol such as TCP or SCTP has
over a connection-less transport protocol such as UDP.
Fast Retransmit: SCTP can quickly determine the loss of a
packet, as a result of its usage of SACK and a mechanism
which sends SACK messages faster than normal when losses
are detected. The result is that losses of SIP messages can
be detected much faster than when SIP is run over UDP
(detection will take at least 500ms, if not more). Note
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that TCP SACK does exist as well, and TCP also has a fast
retransmit option. Over an existing connection, this
results in faster call setup times under conditions of
packet loss, which is very desirable. This is probably the
most significant advantage to SCTP for SIP transport.
Congestion Control: SCTP maintains congestion control over the
entire association. For SIP, this means that the aggregate
rate of messages between two entities can be controlled.
When SIP is run over TCP, the same advantages are afforded.
However, when run over UDP, SIP provides less effective
congestion control. That is because congestion state
(measured in terms of the UDP retransmit interval) is
computed on a transaction by transaction basis, rather than
across all transactions. Congestion control performance is
thus similar to opening N parallel TCP connections, as
opposed to sending N messages over 1 TCP connection.
Transport layer fragmentation: SCTP and TCP provide transport
layer fragmentation. If a SIP message is larger than the
MTU size it is fragmented at the transport layer. When UDP
is used fragmentation occurs at the IP layer. IP
fragmentation increases the likelihood of having packet
losses and make it difficult (when not impossible) NAT and
firewall traversal. This feature will become important if
the size of SIP messages grows dramatically.
3.2 Advantages over TCP
We have shown the advantages of SCTP and TCP over UDP. We now analyze
the advantages of SCTP over TCP.
Head of the Line: SCTP is message based as opposed to TCP that
is stream based. This allows SCTP to separate different
signalling messages at the transport layer. TCP just
understands bytes. Assembling received bytes to form
signalling messages is performed at the application layer.
Therefore, TCP always delivers an ordered stream of bytes
to the application. On the other hand, SCTP can deliver
signalling messages to the application as soon as they
arrive (when using the unordered service). The loss of a
signalling message does not affect the delivery of the rest
of the messages. This avoids the head of line blocking
problem in TCP, which occurs when multiple higher layer
connections are multiplexed within a single TCP connection.
A SIP transaction can be considered an application layer
connection. Between proxies there are multiple transactions
running. The loss of a message in one transaction should
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not adversely effect the ability of a different transaction
to send a message. Thus, if SIP is run between entities
with many transactions occurring in parallel, SCTP can
provide improved performance over SIP over TCP (but not SIP
over UDP; but SIP over UDP is not ideal from a congestion
control standpoint, see above).
Easier Parsing: Another advantage of message based protocols
such as SCTP and UDP over stream based protocols such as
TCP is that they allow easier parsing of messages at the
application layer. There is not need of establishing
boundaries (typically using Content-Length headers) between
different messages. However, this advantage is almost
negligible.
Multihoming: An SCTP connection can be associated with multiple
IP addresses on the same host. Data is always sent over one
of the addresses, but should it become unreachable, data
sent to one can migrate to a different address. This
improves fault tolerance; network failures making one
interface of the server unavailable do not prevent the
service from continuing to operate. SIP servers are likely
to have substantial fault tolerance requirements. It is
worth noting that because SIP is message-oriented, and not
stream oriented, the existing SRV procedures defined in [2]
can accomplish the same goal, even when SIP is run over
TCP. In fact, SRV records allow the "connection" to fail
over to a separate host. Since SIP proxies can run
statelessly, failover can be accomplished without data
synchronization between the primary and its backups. Thus,
the multihoming capabilities of SCTP provide marginal
benefits.
It is important to note that most of the benefits of SCTP for SIP
occur under loss conditions. Therefore, under a zero loss condition,
SCTP transport of SIP should perform on par with TCP transport.
Research is needed to evaluate under what loss conditions the
improvements in setup times and throughput will be observed. The
purpose of this draft is to enable such experimentation in order to
provide concrete data on its applicability to SIP.
4 Usage of SCTP
Usage of SCTP requires no additional headers or syntax in SIP. Below
we show an example of a SIP URL with a transport parameter and an
example of a via header field. In both examples SCTP is the transport
protocol.
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sip:Bob.Johnson@example.com;transport=sctp
Via: SIP/2.0/SCTP ws1234.example.com:5060
Rules for sending a request over SCTP are identical to TCP. The only
difference is that an SCTP sender has to choose a particular stream
within an association in order to send the request (see Section 4.1).
Note that no SCTP identifier needs to be defined for SIP messages.
Therefore, the Payload Protocol Indentifier in SCTP DATA chunks
transporting SIP messages MUST be set to zero.
The SIP transport layers of both peers are responsible to manage the
persistent SCTP connection between them. On the sender side the core
or a client (or server) transaction generates a request (or response)
and passes it to the transport layer. The transport sends the request
to the peer's transaction layer. The peer's transaction layer is
responsible of delivering the incoming request (or response) to the
proper existing server (or client) transaction. If no server (or
client) transaction exists for the incoming message the transport
layer passes the request (or response) to the core, which may decide
to construct a new server (or client) transaction.
4.1 Mapping of SIP Transactions into Streams
SIP transactions need to be mapped into SCTP streams in a way that
avoids Head Of the Line (HOL) blocking. Among all the different ways
of performing this mapping that fulfill this requirement, we have
chosen the simplest one; a SIP entity SHOULD send every SIP message
(request or response) over stream zero with the unordered flag set.
On the receiving side, a SIP entity MUST be ready to receive SIP
messages over any stream.
Note that previous versions of this document proposed to
use SCTP stream IDs as lightweight SIP transaction
identifiers. That proposal has been withdrawn because SIP
now provides a transaction identifier in the branch
parameter of the Via entries. This transaction identifier,
missing in the previous SIP spec [6], makes it unnecessary
to use the SCTP stream IDs to demultiplex SIP traffic.
Some applications introduce an extra layer between the transport
layer and SIP (e.g., compression and/or encryption). This extra layer
sometimes requires ordered delivery of messages from the transport
layer (e.g., TLS [7]). In this case, it is RECOMMENDED that SIP
messages belonging to the same transaction are sent over the same
stream and messages belonging to different transactions are sent over
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different streams. Note that if both sides of the association follow
this recommendation, if a request arrives over a particular stream,
the server is free to return responses over a different stream. This
way, both sides manage the available streams in the sending
direction, independently of the streams chosen by the other side to
send a particular SIP message. This avoids undesirable collisions
when seizing a particular stream.
5 Locating a SIP server
The primary issue when sending a request is determining whether the
next hop server supports SCTP, so that an association can be opened.
SIP entities follow normal SIP procedures to discover [8] a server
that supports SCTP.
6 Security Considerations
No extra security risk outside these specified by [2] are foreseen.
7 Conclusion
This draft has presented a discussion on the applicability of SCTP to
SIP transport, and provided a mechanism for allowing two SCTP-capable
entities to use an SCTP association to exchange SIP traffic.
8 Author's Addresses
Jonathan Rosenberg
dynamicsoft
200 Executive Drive
Suite 120
West Orange, NJ 07052
email: jdrosen@dynamicsoft.com
Henning Schulzrinne
Columbia University
M/S 0401
1214 Amsterdam Ave.
New York, NY 10027-7003
email: schulzrinne@cs.columbia.edu
Gonzalo Camarillo
Ericsson
Advanced Signalling Research Lab.
FIN-02420 Jorvas
Finland
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Phone: +358 9 299 3371
Fax: +358 9 299 3052
Email: Gonzalo.Camarillo@ericsson.com
9 Normative References
[1] R. J. Stewart, Q. Xie, K. Morneault, C. Sharp, H. Schwarzbauer,
T. Taylor, I. Rytina, and M. Kalla, "Stream control transmission
protocol," RFC 2960, Internet Engineering Task Force, Oct. 2000.
[2] J. Rosenberg, H. Schulzrinne, G. Camarillo, A. R. Johnston, J.
Peterson, R. Sparks, M. Handley, and E. Schooler, "SIP: session
initiation protocol," RFC 3261, Internet Engineering Task Force, June
2002.
[3] S. Bradner, "Key words for use in RFCs to indicate requirement
levels," RFC 2119, Internet Engineering Task Force, Mar. 1997.
10 Informative References
[4] L. Coene, "Stream control transmission protocol applicability
statement," RFC 3257, Internet Engineering Task Force, Apr. 2002.
[5] G. Camarillo, H. Schulzrinne, and R. Kantola, "Evaluation of
transport protocols for the session initiation protocol," IEEE
Network , vol. 17, no. 5, 2003.
[6] M. Handley, H. Schulzrinne, E. Schooler, and J. Rosenberg, "SIP:
session initiation protocol," RFC 2543, Internet Engineering Task
Force, Mar. 1999.
[7] T. Dierks and C. Allen, "The TLS protocol version 1.0," RFC 2246,
Internet Engineering Task Force, Jan. 1999.
[8] J. Rosenberg and H. Schulzrinne, "Session initiation protocol
(SIP): locating SIP servers," RFC 3263, Internet Engineering Task
Force, June 2002.
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