Internet Engineering Task Force SIP WG Internet Draft J. Rosenberg dynamicsoft H. Schulzrinne Columbia U. G. Camarillo Ericsson
draft-ietf-sip-sctp-03.txt June 28, 2002draft-ietf-sip-sctp-04.txt November 19, 2003 Expires: December, 2002May, 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 Task Force (IETF), its areas, and its working groups. Note that other groups may also distribute working documents as Internet- Drafts. 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 http://www.ietf.org/ietf/1id-abstracts.txt To view the list Internet-Draft Shadow Directories, see http://www.ietf.org/shadow.html. 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. 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 1 Introduction The Stream Control Transmission Protocol (SCTP)  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) , 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 Potential Benefits Coene et. al. present some of the key benefits of SCTP . We summarize some of these benefits here and analyze how they relate to SIP:SIP (a more detailed analysis can be found in .) 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 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 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  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. 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 ,, 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 ).). 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 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  a server that supports SCTP. 6 Security Considerations No extra security risk outside these specified by  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: email@example.com Henning Schulzrinne Columbia University M/S 0401 1214 Amsterdam Ave. New York, NY 10027-7003 email: firstname.lastname@example.org Gonzalo Camarillo Ericsson Advanced Signalling Research Lab. FIN-02420 Jorvas Finland Phone: +358 9 299 3371 Fax: +358 9 299 3052 Email: Gonzalo.Camarillo@ericsson.com 9 Normative References  R. J. Stewart, Q. Xie, K. Morneault, C. Sharp, H. Schwarzbauer, T. Taylor, I. Rytina, and M. Kalla, L. Zhang, and V. Paxson,"Stream control transmission protocol," RFC 2960, Internet Engineering Task Force, Oct. 2000.  J. Rosenberg, H. Schulzrinne, et al. ,G. Camarillo, A. R. Johnston, J. Peterson, R. Sparks, M. Handley, and E. Schooler, "SIP: Sessionsession initiation protocol," Internet Draft,RFC 3261, Internet Engineering Task Force, Feb.June 2002. Work in progress. S. Bradner, "Key words for use in RFCs to indicate requirement levels," RFC 2119, Internet Engineering Task Force, Mar. 1997. 10 Informative References  L. Coene, "Stream control transmission protocol applicability statement," RFC 3257, Internet Engineering Task Force, Apr. 2002.  G. Camarillo, H. Schulzrinne, and R. Kantola, "Evaluation of transport protocols for the session initiation protocol," IEEE Network , vol. 17, no. 5, 2003.  M. Handley, H. Schulzrinne, E. Schooler, and J. Rosenberg, "SIP: session initiation protocol," RFC 2543, Internet Engineering Task Force, Mar. 1999.  T. Dierks and C. Allen, "The TLS protocol version 1.0," RFC 2246, Internet Engineering Task Force, Jan. 1999.  H. Schulzrinne and J. Rosenberg, "SIP: LocatingRosenberg and H. Schulzrinne, "Session initiation protocol (SIP): locating SIP servers," Internet Draft,RFC 3263, Internet Engineering Task Force, Feb.June 2002. Work in progress.