INTERNET-DRAFT L. Coene(Ed) Internet Engineering Task Force Siemens Issued: January 2002 Expires: July 2002 Telephony Signalling Transport over SCTP applicability statement 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 The list of Internet-Draft Shadow Directories can be accessed at http://www.ietf.org/shadow.html 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. Coene, et al [Page 1] Draft SCTP & Telephony signalling January 2002 Table of contents Telephony signalling over SCTP Applicability statement ......... ii Chapter 1: Introduction ........................................ 2 Chapter 1.1: Terminology ....................................... 2 Chapter 1.2: Contributors ...................................... 3 Chapter 1.3: Overview ......................................... 3 Chapter 2: Applicability of telephony signalling transport using SCTP ..................................................... 4 Chapter 3: Issues for transporting Telephony signalling information over SCTP .......................................... 4 Chapter 3.1: Congestion control ................................ 4 Chapter 3.2: Detection of failures ............................. 5 Chapter 3.2.1: Retransmission TimeOut (RTO) calculation ........ 5 Chapter 3.2.2: Heartbeat ....................................... 5 Chapter 3.2.3: Maximum Number of retransmissions ............... 5 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 ............................. 6 Chapter 5: References and related work ......................... 7 Chapter 6: Acknowledgments ..................................... 7 Chapter 7: Author's address .................................... 7 1 INTRODUCTION Transport of telephony signalling requires special considerations. In order to use SCTP, special care must be taken to meet the performance, 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 association, within which all user messages are delivered in sequence except for those submitted to the unordered delivery service. Coene, et al [Page 2] Draft SCTP & Telephony signalling January 2002 1.2 Contributors The following people contributed to the document: L. Coene(Editor), M. Tuexen, G. Verwimp, J. Loughney, R.R. Stewart, Qiaobing Xie, M. Holdrege, M.C. Belinchon, A. Jungmaier, and L. Ong. 1.3 Overview SCTP provides a general purpose, reliable transport between two endpoints. 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 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 Coene, et al [Page 3] Draft SCTP & Telephony signalling January 2002 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 indications, data operation primitives, ... for a particular user/application protocol). (2) SCTP, specially configured to meet the telephony application performance 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 transport 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 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 potential congestion collapse of the network. Some telephony signalling applications may have their own congestion Coene, et al [Page 4] Draft SCTP & Telephony signalling January 2002 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 destination 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 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 Coene, et al [Page 5] Draft SCTP & Telephony signalling January 2002 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. 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 message 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 multihoming 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. Coene, et al [Page 6] Draft SCTP & Telephony signalling January 2002 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", , 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 Architecture 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. 6 Acknowledgments This document was initially developed by a design team consisting of Lode Coene, John Loughney, Michel Tuexen, Randall R. Stewart, Qiaobing Xie, Matt Holdrege, Maria-Carmen Belinchon, Andreas Jungmaier, Gery Verwimp and Lyndon Ong. 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 Coene, et al [Page 7] Draft SCTP & Telephony signalling January 2002 Expires: July 2002 Full Copyright Statement Copyright (C) The Internet Society (2002). All Rights Reserved. This document and translations of it may be copied and furnished to others, and derivative works that comment on or otherwise explain it or assist in its implementation may be prepared, copied, published and distributed, in whole or in part, without restriction of any kind, provided that the above copyright notice and this paragraph are included on all such copies and derivative works. However, this document itself may not be modified in any way, such as by removing the copyright notice or references to the Internet Society or other Internet organizations, except as needed for the purpose of developing Internet standards in which case the procedures for copyrights defined in the Internet Standards process must be followed, or as required to translate it into languages other than English. The limited permissions granted above are perpetual and will not be revoked by the Internet Society or its successors or assigns. This document and the information contained herein is provided on an "AS IS" basis and THE INTERNET SOCIETY AND THE INTERNET ENGINEERING TASK FORCE DISCLAIMS ALL WARRANTIES, EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO ANY WARRANTY THAT THE USE OF THE INFORMATION HEREIN WILL NOT INFRINGE ANY RIGHTS OR ANY IMPLIED WARRANTIES OF MERCHANTABILITY OR FITNESS FOR A PARTICULAR PURPOSE. Coene, et al [Page 8]