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The Stream Control Transmission Protocol (SCTP) as a Transport for the Session Initiation Protocol (SIP)

The information below is for an old version of the document that is already published as an RFC.
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This is an older version of an Internet-Draft that was ultimately published as RFC 4168.
Authors Henning Schulzrinne , Jonathan Rosenberg , Gonzalo Camarillo
Last updated 2015-10-14 (Latest revision 2005-02-04)
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IESG IESG state Became RFC 4168 (Proposed Standard)
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SIP                                                         J. Rosenberg
Internet-Draft                                             Cisco Systems
Expires: July 14, 2005                                    H. Schulzrinne
                                                     Columbia University
                                                            G. Camarillo
                                                        January 13, 2005

   The Stream Control Transmission Protocol (SCTP) as a Transport for
                 the Session Initiation Protocol (SIP)

Status of this Memo

   This document is an Internet-Draft and is subject to all provisions
   of section 3 of RFC 3667.  By submitting this Internet-Draft, each
   author represents that any applicable patent or other IPR claims of
   which he or she is aware have been or will be disclosed, and any of
   which he or she become aware will be disclosed, in accordance with
   RFC 3668.

   Internet-Drafts are working documents of the Internet Engineering
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   The list of current Internet-Drafts can be accessed at

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   This Internet-Draft will expire on July 14, 2005.

Copyright Notice

   Copyright (C) The Internet Society (2005).


   This document specifies a mechanism for usage of SCTP (the Stream
   Control Transmission Protocol) as the transport between SIP (Session
   Initiation Protocol) entities.  SCTP is a new protocol which provides

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   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.

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.  Transport Parameters . . . . . . . . . . . . . . . . . . . . .  5
   5.  SCTP Usage . . . . . . . . . . . . . . . . . . . . . . . . . .  6
     5.1   Mapping of SIP Transactions into SCTP Streams  . . . . . .  7
   6.  Locating a SIP Server  . . . . . . . . . . . . . . . . . . . .  8
   7.  Security Considerations  . . . . . . . . . . . . . . . . . . .  8
   8.  IANA Considerations  . . . . . . . . . . . . . . . . . . . . .  8
   9.  References . . . . . . . . . . . . . . . . . . . . . . . . . .  9
   9.1   Normative References . . . . . . . . . . . . . . . . . . . .  9
   9.2   Informative References . . . . . . . . . . . . . . . . . . .  9
       Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . 10
       Intellectual Property and Copyright Statements . . . . . . . . 11

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1.  Introduction

   The Stream Control Transmission Protocol (SCTP) [4] 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) [5], 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.  This document defines
   transport of SIP over SCTP.

2.  Terminology

   The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
   document are to be interpreted as described in RFC 2119 [1].

3.  Potential Benefits

   Coene et.  al.  present some of the key benefits of SCTP [10].  We
   summarize some of these benefits here and analyze how they relate to
   SIP (a more detailed analysis can be found in [12]).

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 500 ms, 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.

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   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 one 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 makes 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 which 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).

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   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 (Service Selection) procedures
      defined in [5] 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.

4.  Transport Parameters

   A SIP URIs can carry a transport parameter indicating the transport
   protocol to be used.  RFC 3261 defines the value "sctp" for SCTP but
   does not define the value for the transport parameter for TLS over
   SCTP.  Note that the value "tls", defined by RFC 3261, is intended
   for TLS over TCP.

   Here we define the value "tls-sctp" for the SIP URI transport
   parameter to be used for messages to be sent over TLS over SCTP [8].
   The updated augmented BNF (Backus-Naur Form) [2] for this parameter
   is the following (the original BNF for this parameter can be found in
   RFC 3261):

   transport-param   =  "transport="
                        ( "udp" / "tcp" / "sctp" / "tls" / "tls-sctp"
                        / other-transport)

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   The following are examples of SIP URIs using "sctp" and "tls-sctp" in
   their transport parameters:;transport=sctp;transport=tls-sctp

   Via header fields also carry a transport protocol identifier.  RFC
   3261 defines the value "SCTP" for SCTP but does not define the value
   for the transport parameter for TLS over SCTP.  Note that the value
   "TLS", defined by RFC 3261, is intended for TLS over TCP.

   Here we define the value "TLS-SCTP" for the transport part of the Via
   header field to be used for requests sent over TLS over SCTP [8].
   The updated augmented BNF (Backus-Naur Form) [2] for this parameter
   is the following (the original BNF for this parameter can be found in
   RFC 3261):

   transport         =  "UDP" / "TCP" / "TLS" / "SCTP" / "TLS-SCTP"
                        / other-transport

   The following are examples of Via header fields using "SCTP" and

     Via: SIP/2.0/SCTP
     Via: SIP/2.0/TLS-SCTP

5.  SCTP Usage

   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 5.1).

   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 for managing
   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 for 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

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   transport layer passes the request (or response) to the core, which
   may decide to construct a new server (or client) transaction.

5.1  Mapping of SIP Transactions into SCTP 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 fulfil 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.

      In the past, it was proposed to use SCTP stream IDs as lightweight
      SIP transaction identifiers.  That proposal was withdrawn because
      SIP now provides (as defined in RFC 3261 [5]) a transaction
      identifier in the branch parameter of the Via entries.  This
      transaction identifier, missing in the previous SIP spec [9],
      makes it unnecessary to use the SCTP stream IDs to demultiplex SIP

   SIP has many circumstances in which the use of TLS [3] is required,
   for instance, for routing a SIPS URI [5].  As defined in RFC 3436
   [8], TLS running over SCTP MUST NOT use the SCTP unordered delivery
   service.  Moreover, any SIP use of an extra layer between the
   transport layer and SIP that requires ordered delivery of messages
   MUST NOT use the SCTP unordered delivery service.

   SIP applications that require ordered delivery of messages from the
   transport layer (e.g., TLS) SHOULD send SIP messages belonging to the
   same SIP transaction over the same SCTP stream.  Additionally, they
   SHOULD send messages belonging to different SIP transactions over
   different SCTP streams, as long as there are enough available

      A common scenario where the above mechanism should be used
      consists of two proxies exchanging SIP traffic over a TLS
      connection using SCTP as the transport protocol.  This works
      because all of the SIP transactions between the two proxies can be
      established within one SCTP association.

   Note that if both sides of the association follow this
   recommendation, when 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.

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6.  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 [6] a server
   that supports SCTP.

   However, in order to be able to use TLS on top of SCTP, an extra
   definition is needed.  RFC 3263 defines the NAPTR (Naming Authority
   Pointer) [7] service value "SIP+D2S" for SCTP but fails to define a
   value for TLS over SCTP.  Here we define the NAPTR service value
   "SIPS+D2S" for servers that support TLS over SCTP [8].

7.  Security Considerations

   The security issues raised in RFC 3261 [5] are not worsened by SCTP,
   provided the advice in Section 5.1 is followed and TLS over SCTP [8]
   is used where TLS would be required in RFC 3261 [5] or in RFC 3263
   [6].  So, the mechanisms described in RFC 3436 [8] MUST be used when
   SIP runs on top of TLS [3] and SCTP.

8.  IANA Considerations

   This document defines a new value (tls-sctp) for the SIP and SIPS URI
   transport parameter.  The IANA is requested to add a reference to
   this document to the entry of the transport parameter in the
   "SIP/SIPS URI Parameters" subregistry, which is located under the
   "Session Initiation Protocol (SIP) Parameters" registry.  The
   reference should appear in double-brackets, as indicated in RFC 3969
   [11].  The resulting entry should be:

   Parameter Name     Predefined Values     Reference
   --------------     -----------------     ---------
   transport          Yes                   [RFC3261] [[RFCXXXX]]

   This document also defines a new NAPTR service field value
   (SIPS+D2S).  The IANA is requested to register this value under the
   "Registry for the SIP SRV Resource Record Services Field".  The
   resulting entry should be:

   Services Field        Protocol  Reference
   --------------------  --------  ---------
   SIPS+D2S              SCTP      [RFCXXXX]

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9.  References

9.1  Normative References

   [1]  Bradner, S., "Key words for use in RFCs to Indicate Requirement
        Levels", BCP 14, RFC 2119, March 1997.

   [2]  Crocker, D. and P. Overell, "Augmented BNF for Syntax
        Specifications: ABNF", RFC 2234, November 1997.

   [3]  Dierks, T. and C. Allen, "The TLS Protocol Version 1.0", RFC
        2246, January 1999.

   [4]  Stewart, R., Xie, Q., Morneault, K., Sharp, C., Schwarzbauer,
        H., Taylor, T., Rytina, I., Kalla, M., Zhang, L. and V. Paxson,
        "Stream Control Transmission Protocol", RFC 2960, October 2000.

   [5]  Rosenberg, J., Schulzrinne, H., Camarillo, G., Johnston, A.,
        Peterson, J., Sparks, R., Handley, M. and E. Schooler, "SIP:
        Session Initiation Protocol", RFC 3261, June 2002.

   [6]  Rosenberg, J. and H. Schulzrinne, "Session Initiation Protocol
        (SIP): Locating SIP Servers", RFC 3263, June 2002.

   [7]  Mealling, M., "Dynamic Delegation Discovery System (DDDS) Part
        Three: The Domain Name System (DNS) Database", RFC 3403, October

   [8]  Jungmaier, A., Rescorla, E. and M. Tuexen, "Transport Layer
        Security over Stream Control Transmission Protocol", RFC 3436,
        December 2002.

9.2  Informative References

   [9]   Handley, M., Schulzrinne, H., Schooler, E. and J. Rosenberg,
         "SIP: Session Initiation Protocol", RFC 2543, March 1999.

   [10]  Coene, L., "Stream Control Transmission Protocol Applicability
         Statement", RFC 3257, April 2002.

   [11]  Camarillo, G., "The Internet Assigned Number Authority (IANA)
         Uniform Resource Identifier (URI) Parameter Registry for the
         Session Initiation Protocol (SIP)", BCP 99, RFC 3969, December

   [12]  Camarillo, G., Schulrinne, H. and R. Kantola, "Evaluation of
         Transport Protocols for the Session Initiation Protocol", IEEE
         Network vol. 17, no. 5, 2003.

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Authors' Addresses

   Jonathan Rosenberg
   Cisco Systems
   600 Lanidex Plaza
   Parsippany, NJ  07054

   Phone: +1 973 952-5000

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


   Gonzalo Camarillo
   Hirsalantie 11
   Jorvas  02420


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