Internet Engineering Task Force                                   SIP WG
Internet Draft                           Rosenberg/Schulzrinne/Camarillo
draft-rosenberg-sip-sctp-01.txt         dynamicsoft,Columbia U.,Ericsson
May 18, 2001
Expires: November, 2001

                      SCTP as a Transport for SIP


   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 working documents as Internet-

   Internet-Drafts are draft documents valid for a maximum of six months
   and may be updated, replaced, or obsoleted by other documents at any
   time.  It is inappropriate to use Internet-Drafts as reference
   material or to cite them other than as "work in progress".

   The list of current Internet-Drafts can be accessed at

   To view the list Internet-Draft Shadow Directories, see


   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.

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

Rosenberg/Schulzrinne/Camarillo                               [Page 1]

Internet Draft        SCTP as a Transport for SIP           May 18, 2001

   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 Potential Benefits

   Coene et. al. present some of the key benefits of SCTP [3]. We
   summarize some of these benefits here and analyze how they relate to

2.1 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
             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 connection. 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

Rosenberg/Schulzrinne/Camarillo                               [Page 2]

Internet Draft        SCTP as a Transport for SIP           May 18, 2001

             (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.

2.2 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
             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

Rosenberg/Schulzrinne/Camarillo                               [Page 3]

Internet Draft        SCTP as a Transport for SIP           May 18, 2001

             different messages. However, this advantage is almost

        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

   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.

3 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. In both examples SCTP is the transport

        Via: SIP/2.0/SCTP

   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.

   It is important noting that a receiver uses SIP headers such as
   Call-ID to demultiplex requests (or responses) that belong to
   different transactions and are received over the same STCP
   association. The stream id MUST NOT be used for this purpose.

Rosenberg/Schulzrinne/Camarillo                               [Page 4]

Internet Draft        SCTP as a Transport for SIP           May 18, 2001

        This way a receiver is not affected by the way a particular
        sender maps transactions into streams. The receiver will
        always be able to properly demultiplex incoming SIP traffic
        without knowing the mapping policy of the sender.

3.1 Mapping of transactions into streams

   The more efficient mapping of transactions into streams consists of
   sending requests belonging to the same call over the same ordered
   stream. Requests belonging to different calls will be mapped into
   different streams. It is RECOMMENDED that some kind of stateless hash
   be used so that requests for the same call all be mapped into the
   same stream.

        Note that if the number of calls handled by the SIP entity
        is larger than the number of available streams some calls
        would have to share the same stream. This would result in
        the head of the line blocking problem described previously.

   Responses are mapped in the same way - responses that belong to the
   same call are sent over the same oreered stream. However, final
   responses should be transmitted with the unordered flag set. This
   prevents lost provisional responses from delaying the delivery of
   final responses.

   Some implementors might think that this way of mapping transactions
   into streams is somehow complicated. It is possible to perform this
   mapping in a much simpler way. Senders can take advantage of the
   ordering of requests that SIP performs at the application layer. That
   is, SIP does not send a new request until the last transaction has
   completed. Therefore, all requests and responses can be sent with the
   unordered flag set over any stream. Effectively, a single stream can
   be used for all SIP traffic. This way of performing the mapping is
   almost as effective as the one described previously and it has the
   advantage of being simpler.

        The scenarios where this second way of performing the
        mapping might result in reordering of requests/responses
        represent corner cases that do not justify the increase in
        the complexity of the first solution.

   It is RECOMMENDED to use the second approach because it combines
   simplicity and efficiency.

4 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.

Rosenberg/Schulzrinne/Camarillo                               [Page 5]

Internet Draft        SCTP as a Transport for SIP           May 18, 2001

   This draft assumes that SRV records are the primary vehicle for such
   determinations. RFC2543bis [4] describes the process that an entity
   (UAC or proxy) that wishes to send a request to a particular URI MUST

   The format of the SRV RR as described in [5] is shown below:

        _Service._Proto.Name TTL Class Priority weight Port Target

   When SRV records are to be used, the service to use when querying for
   the SRV record is "sip" and the transport protocol is "sctp". So, a
   SIP client that wants to discover a SIP server that supports SCTP for
   the domain does a lookup of

   SCTP associations that were opened by proxies as the result of a
   successful SRV query SHOULD remain open after the transaction
   completes. The amount of time after completion of a transaction,
   before which the connection is closed, is configurable.

        The rule here for SRV provides a neat way to differentiate
        among connections between proxies, and between proxies and
        UAs and UAs and proxies. You really only want and need long
        lived connections between proxies. It is very likely that
        only proxies have SRV record entries.

5 Conclusion

   This draft has presented a discussion on the applicability of SCTP to
   SIP transport, and provided an experimental mechanism for allowing
   two SCTP-capable entities to establish and use an SCTP connection.

6 Author's Addresses

   Jonathan Rosenberg
   200 Executive Drive
   Suite 120
   West Orange, NJ 07052

   Henning Schulzrinne
   Columbia University
   M/S 0401
   1214 Amsterdam Ave.
   New York, NY 10027-7003

Rosenberg/Schulzrinne/Camarillo                               [Page 6]

Internet Draft        SCTP as a Transport for SIP           May 18, 2001


   Gonzalo Camarillo
   Advanced Signalling Research Lab.
   FIN-02420 Jorvas
   Phone: +358 9 299 3371
   Fax: +358 9 299 3052

7 Bibliography

   [1] R. Stewart, Q. Xie, K. Morneault, C. Sharp, H. Schwarzbauer, T.
   Taylor, I. Rytina, M. Kalla, L. Zhang, and V. Paxson, "Stream control
   transmission protocol," Request for Comments 2960, Internet
   Engineering Task Force, Oct.  2000.

   [2] M. Handley, H. Schulzrinne, E. Schooler, and J. Rosenberg, "SIP:
   session initiation protocol," Request for Comments 2543, Internet
   Engineering Task Force, Mar. 1999.

   [3] L. Coene et al.  , "Stream control transmission protocol
   applicability statement," Internet Draft, Internet Engineering Task
   Force, Apr. 2001.  Work in progress.

   [4] M. Handley, H. Schulzrinne, E. Schooler, and J. Rosenberg, "SIP:
   Session initiation protocol," Internet Draft, Internet Engineering
   Task Force, Nov. 2000.  Work in progress.

   [5] A. Gulbrandsen, P. Vixie, and L. Esibov, "A DNS RR for specifying
   the location of services (DNS SRV)," Request for Comments 2782,
   Internet Engineering Task Force, Feb. 2000.

                           Table of Contents

   1          Introduction ........................................    1
   2          Potential Benefits ..................................    2
   2.1        Advantages over UDP .................................    2

Rosenberg/Schulzrinne/Camarillo                               [Page 7]

Internet Draft        SCTP as a Transport for SIP           May 18, 2001

   2.2        Advantages over TCP .................................    3
   3          Usage of SCTP .......................................    4
   3.1        Mapping of transactions into streams ................    5
   4          Locating a SIP server ...............................    5
   5          Conclusion ..........................................    6
   6          Author's Addresses ..................................    6
   7          Bibliography ........................................    7

Rosenberg/Schulzrinne/Camarillo                               [Page 8]