SIP WG R. Mahy
Internet-Draft Cisco Systems, Inc.
Expires: November 30, 2003 Jun 2003
Connection Reuse in the Session Initiation Protocol (SIP)
draft-mahy-sip-connect-reuse-00.txt
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Copyright (C) The Internet Society (2003). All Rights Reserved.
Abstract
When SIP entities use a connection oriented protocol to send a
request, they typically originate their connections from an ephemeral
port. The SIP protocol includes mechanisms which insure that
responses to a request, and new requests sent in the original
direction reuse an existing connection. However, new requests sent
in the opposite direction are unlikely to reuse the existing
connection. This frequently causes a pair of SIP entities to use one
connection for requests sent in each direction, and can result in
potential scaling and performance problems. This document proposes
requirements and a mechanism which address this deficiency.
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Table of Contents
1. Conventions . . . . . . . . . . . . . . . . . . . . . . . . . 3
2. Introduction and Problem Statement . . . . . . . . . . . . . . 3
3. Requirements . . . . . . . . . . . . . . . . . . . . . . . . . 5
4. Overview of Proposed Mechanism . . . . . . . . . . . . . . . . 6
4.1 Authorizing an alias request . . . . . . . . . . . . . . . . . 8
4.2 Formal Syntax . . . . . . . . . . . . . . . . . . . . . . . . 9
5. Security Considerations . . . . . . . . . . . . . . . . . . . 9
6. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 9
7. Acknowledgments . . . . . . . . . . . . . . . . . . . . . . . 9
Normative References . . . . . . . . . . . . . . . . . . . . . 9
Informational References . . . . . . . . . . . . . . . . . . . 10
Author's Address . . . . . . . . . . . . . . . . . . . . . . . 10
Intellectual Property and Copyright Statements . . . . . . . . 11
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1. Conventions
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 [2].
2. Introduction and Problem Statement
SIP [1] entities can communicate using either unreliable/
connectionless (ex: UDP) or reliable/connection-oriented (ex: TCP,
SCTP [11]) transport protocols. When SIP entities use a
connection-oriented protocol (such as TCP or SCTP) to send a request,
they typically originate their connections from an ephemeral port.
In the following example, Entity A listens for SIP requests over TLS
[4] on TCP port 5061 (the default port for SIP over TLS over TCP),
but uses an ephemeral port (port 8293) for a new connection to Entity
B. These entities could be SIP User Agents or SIP Proxy Servers.
+-----------+ 8293 (UAC) 5061 (UAS) +-----------+
| |--------------------------->| |
| Entity | | Entity |
| A | | B |
| | 5061 (UAS) | |
+-----------+ +-----------+
The SIP protocol includes mechanisms which insure that responses to a
request reuse the existing connection which is typically still
available, and also includes provisions for reusing existing
connections for other requests sent by the originator of the
connection. However, new requests sent in the opposite direction
(routed from the target of the original connection toward the
originator of the original connection) are unlikely to reuse the
existing connection. This frequently causes a pair of SIP entities
to use one connection for requests sent in each direction, as shown
below.
+-----------+ 8293 5061 +-----------+
| |.......................>| |
| Entity | | Entity |
| A | 5061 9741 | B |
| |<-----------------------| |
+-----------+ +-----------+
This extra pair of connections can result in potential scaling and
performance problems. For example, each new connection using TLS
requires a TCP 3-way handshake, a handful of round-trips to establish
TLS, and (typically) expensive asymetric authentication and key
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generation algorithms, and certificate verification. This
effectively doubles the load on each entity. Setting up a second
connection can also cause excessive delay (especially in networks
with long round-trip times) for subsequent requests, even requests in
the context of an existing dialog (for example a reINVITE or BYE
after an initial INVITE, or a NOTIFY after a SUBSCRIBE [8] or a REFER
[9]).
Consider the call flow shown below where Proxy A and Proxy B use the
Record-Route mechanism to stay involved in a dialog. Proxy B will
establish a new TLS connection just to send a BYE request.
INVITE -> create connection 1
<- 200 response over connection 1
ACK -> reuse connection 1
<- BYE create connection 2
-> 200 response over connection 2
ReINVITEs are expected to be handled automatically and rapidly in
order to avoid media and session state from being out of step. If a
reINVITE requires a new TLS connection, the reINVITE could be delayed
by several extra round-trip times. Depending on the round-trip time,
this combined delay could be perceptible or even annoying to a human
user. This is especially problematic for some common SIP call flows
(for example, the recommended example flow in figure number 4 in 3pcc
[7]) use many reINVITEs.
Consider also a call flow where a handheld organizer sends a REFER
request which establishes a dialog to a SIP phone. Typically this
would require a second connection back to the handheld to be
established.
REFER -> connection 1
<- 202 connection 1
<- NOTIFY connection 2
200 -> connection 2
INVITE ->
<- 200
<- NOTIFY connection 2
200 -> connection 2
Likewise when clusters or farms of cooperating SIP servers (for
example proxy servers) are configured together, SIP entities have no
way to prefer a server with an existing connection. For example,
Proxy server B has no mechanism to choose an existing connection with
Proxy cluster A.
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+-----------+
| |
| Proxy |
| A1 | +-----------+
| | | |
+-----------+ | Proxy |
+-----------+ 8293 5061 | B |
| |----------------------->| |
| Proxy | +-----------+
| A2 |
| |
+-----------+
As a result, Proxy B might open a new connection to another proxy
server for requests sent in the opposite direction.
+-----------+
| |
| Proxy |
| A1 | 5061 9741 +-----------+
| |<.......................| |
+-----------+ | Proxy |
+-----------+ 8293 5061 | B |
| |----------------------->| |
| Proxy | +-----------+
| A2 |
| |
+-----------+
The rules for handling the Transport layer described in Section 18 of
SIP [1] do not associate incoming connections with the listening port
which corresponds to the same SIP entity. If the Tranport layer had
some way to associate these connections, then request and responses
originated from either node could reuse existing connections as shown
below.
+-----------+ +-----------+
| | | |
| Node A | 8293 5061 | Node B |
| |<======================>| |
| | | |
+-----------+ +-----------+
3. Requirements
1. A connection sharing mechanism SHOULD allow SIP entities to reuse
existing connections for requests and repsonses originated from
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either peer in the connection.
2. A connection sharing mechanism SHOULD allow SIP entities to reuse
existing connections with closely coupled nodes which act as a
single SIP entity (for example a cluster of nodes acting as a
proxy server).
3. A connection sharing mechanism MUST NOT require UACs (clients) to
send all traffic from well-know SIP ports.
4. A connection sharing mechanism MUST NOT require configuring
ephemeral port numbers in DNS.
5. A connection sharing mechanism MUST prevent unauthorized
hijacking of other connections.
6. Connection sharing SHOULD persist across SIP transactions and
dialogs.
4. Overview of Proposed Mechanism
The proposed mechanism uses a new Via header field parameter. The
"alias" parameter is included in a Via header field value to indicate
that the originator of the request wants to create a transport layer
alias, so that the sent-by address becomes mapped to the current
connection.
Assuming the Via header field value shown below from the most recent
request arrived over a connection from 60.54.32.1 port 8241:
Via: SIP/2.0/TLS 60.54.32.1:5061;branch=z9hG4bKa7c8dze ;alias
The transport layer creates an alias, such that any requests going to
the "advertised address" (60.54.32.1 port 5061) are instead sent over
the existing connection (to the "target" of the alias) which is
coming from port 8241. This sharing continues as long as the target
connection stays up. The SIP community recommends that servers keep
connections up unless they need to reclaim resources, and that
clients keep connections up as long as they are needed. Connection
reuse works best when the client and the server maintain their
connections for long periods of time. SIP entities therefore SHOULD
NOT drop connections on completion of a transaction or termination of
a dialog.
Likewise when clusters or farms of cooperating SIP servers (for
example proxy servers) are configured together, the proposed
mechanism allows a SIP entity to select a server with an existing
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connection. With the proposed mechanism, Proxy B sends requests for
Proxy cluster A to node A2 with whom it shares an existing
connection.
+-----------+
| |
| Proxy |
| A1 | +-----------+
| | | |
+-----------+ | Proxy |
+-----------+ 8293 5061 | B |
| |<======================>| |
| Proxy | +-----------+
| A2 |
| |
+-----------+
For example, on receipt of a message with the topmost Via header
shown below, the transport layer creates an alias such that requests
going to the advertised address (host-a.atlanta.com) are sent over
the target connection (from 60.54.32.1:8241).
Via: SIP/2.0/TLS host-a.atlanta.com;branch=z9hG4bK7c8dze ;alias
As a result, this has an important interaction with the DNS
resolution mechanisms for SIP described in RFC3263 [6]. When new
requests arrive for host-a.atlanta.com, proxy B still needs to
perform a DNS NAPTR lookup to select the transport. Once the
transport is selected, an SRV lookup would ordinarily occur to find
the appropriate port number. In this case, the transport layer uses a
connection reuse alias instead of performing the SRV query.
Below is a partial DNS zone file for atlanta.com.
; NAPTR queries for the current domain (atlanta.com)
;
; order pref flags service regexp replacement
@ IN NAPTR 50 50 "s" "SIPS+D2T" "" _sips._tcp.
; SRV records for the proxy use 5060/5061
;
;; Priority Weight Port Target
_sips._tcp.host-a IN SRV 0 1 5061 host-a1
_sips._tcp.host-a IN SRV 0 1 5061 host-a2
host-a1 IN A 60.54.32.1
host-a2 IN A 60.54.32.2
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The existence of an alias parameter is treated as a request which
asks the transport layer to create an alias (named by the sent-by
parameter, which could be a hostname) that points to the alias
target (the current connection)
This mechanism is fully backwards compatible with existing
implementations. If the proposed Via parameter is not understood by
the recipient, it will be ignored and the two implementations will
revert to current behavior (two connections).
4.1 Authorizing an alias request
Authorizing connection aliases is essential to prevent connection
hijacking. For example a program run by a malicious user of a
multiuser system could attempt to hijack SIP requests destined for
the well-known SIP port from a large relay proxy.
To correctly authorize an alias, both the active connection and the
alias need to authenticate using the same credentials. This could be
accomplished using one of two mechanisms. The first (and preferred)
mechanism is using TLS mutual authentication, such that the
subjectAltName of the originator certificate corresponds to both the
current connection and the target address of the alias. The Via
sent-by address needs to be within the scope protected by the
certificate presented by the originator during TLS mutual
authentication and the received IP address needs be a valid IP
address for the sent-by host or hosts. In other words, the sent-by
address and port combination MUST be resolvable from the
subjectAltName of the originator certificate, and the received IP
address MUST be resolvable from the sent-by address. This is in
addition to other requirements for TLS authentication and
authorization discussed in SIP [1] and Locating SIP Servers [6].
The second mechanism is to accept an alias if the target address of
the alias is equivalent (using SIP comparison rules) to a valid
Contact already registered by the same user. This user could be
authenticated through any SIP or TLS mechanism (ex: user certificate,
or Kerberos [10]), but would typically use Digest authentication [5].
For example, if Alice registers a Contact of 123.45.67.89:5061, she
could inform Proxy 1 of the existence of a connection to her from
Proxy 2. This would allow her to preemptively originate TLS
connections, as her user agent may not have access to a site
certificate with which to authenticate incoming TLS connections.
+-----------+
| |
| Proxy |
+-----------+ 8672 5061 | 1 |
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| |----------------------->| |
| Alice | +-----------+
| | +-----------+
| |----------------------->| |
+-----------+ 8293 5061 | Proxy |
| 2 |
| |
+-----------+
4.2 Formal Syntax
The following syntax specification uses the augmented Backus-Naur
Form (BNF) as described in RFC-2234 [3]. This document proposes to
extend via-params to include a a new via-alias defined below.
via-params = via-ttl / via-maddr / via-received / via-branch /
via-alias / via-extension
via-alias = "alias"
5. Security Considerations
This document presents requirements and a mechanism for reusing
existing connections easily. Connection reuse presents many
opportunities for abuse and hijacking, but these attacks can be
prevented if the guidelines in the authorization section are
followed.
6. IANA Considerations
This document introduces no additional considerations for IANA.
7. Acknowledgments
Thanks to Jon Peterson for helpful answers about certificate behavior
with SIP, Jonathan Rosenberg for his initial support of this concept,
and Cullen Jennings for providing a sounding board for this idea.
Normative References
[1] 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.
[2] Bradner, S., "Key words for use in RFCs to Indicate Requirement
Levels", BCP 14, RFC 2119, March 1997.
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[3] Crocker, D. and P. Overell, "Augmented BNF for Syntax
Specifications: ABNF", RFC 2234, November 1997.
[4] Dierks, T., Allen, C., Treese, W., Karlton, P., Freier, A. and
P. Kocher, "The TLS Protocol Version 1.0", RFC 2246, January
1999.
[5] Franks, J., Hallam-Baker, P., Hostetler, J., Lawrence, S.,
Leach, P., Luotonen, A. and L. Stewart, "HTTP Authentication:
Basic and Digest Access Authentication", RFC 2617, June 1999.
[6] Rosenberg, J. and H. Schulzrinne, "Session Initiation Protocol
(SIP): Locating SIP Servers", RFC 3263, June 2002.
Informational References
[7] Rosenberg, J., Schulzrinne, H., Camarillo, G. and J. Peterson,
"Best Current Practices for Third Party Call Control in the
Session Initiation Protocol", draft-ietf-sipping-3pcc-03 (work
in progress), March 2003.
[8] Roach, A., "Session Initiation Protocol (SIP)-Specific Event
Notification", RFC 3265, June 2002.
[9] Sparks, R., "The Session Initiation Protocol (SIP) Refer
Method", RFC 3515, April 2003.
[10] Kohl, J. and B. Neuman, "The Kerberos Network Authentication
Service (V5)", RFC 1510, September 1993.
[11] 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.
Author's Address
Rohan Mahy
Cisco Systems, Inc.
101 Cooper Street
Santa Cruz, CA 95060
USA
EMail: rohan@cisco.com
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