Internet Engineering Task Force SIP WG
Internet Draft G. Camarillo
Ericsson
draft-ietf-sip-compression-02.txt
October 29, 2002
Expires: April 2003
Compressing 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 describes a mechanism to signal that compression is
desired for one or more SIP messages. It also states when it is
appropriate to send compressed SIP messages to a SIP entity.
G. Camarillo [Page 1]
Internet Draft SIP October 29, 2002
Table of Contents
1 Introduction ........................................ 3
2 Overview of operation ............................... 4
3 SigComp implementations for SIP ..................... 4
4 Sending a Request to a Server ....................... 4
4.1 Obtaining a SIP or SIPS URI with comp=sigcomp ....... 5
5 Sending a Response to a Client ...................... 6
6 Double Record-Routing ............................... 7
7 Error Situations .................................... 7
8 Augmented BNF ....................................... 7
9 Example ............................................. 8
10 Security Considerations ............................. 11
11 Acknowledges ........................................ 11
12 Authors' Addresses .................................. 11
13 Normative References ................................ 11
14 Informative References .............................. 11
G. Camarillo [Page 2]
Internet Draft SIP October 29, 2002
1 Introduction
A SIP [1] client sending a request to a SIP server typically performs
a DNS lookup for the domain name of the server. When NAPTR [3] or SRV
[4] records are available for the server, the client can specify the
type of service it wants. The service in this context is the
transport protocol to be used by SIP (e.g., UDP, TCP or SCTP). A SIP
server that supports, for instance, three different transport
protocols, will have three different DNS entries.
Since it is foreseen that the number of transport protocols supported
by a particular application layer protocol is not going to grow
dramatically, having a DNS entry per transport seems like a scalable
enough solution.
However, sometimes it is necessary to include new layers between the
transport protocol and the application layer protocol. Examples of
these layers are transport layer security and compression. If DNS was
used to discover the availability of these layers for a particular
server, the number of DNS entries needed for that server would grow
dramatically.
A server that, for example, supported TCP and SCTP as transports, TLS
for transport security and SigComp for signaling compression, would
need the 8 DNS entries listed below:
1. TCP, no security, no compression
2. TCP, no security, SigComp
3. TCP, TLS, no compression
4. TCP, TLS, SigComp
5. SCTP, no security, no compression
6. SCTP, no security, SigComp
7. SCTP, TLS, no compression
8. SCTP, TLS, SigComp
It is clear that this way of using DNS is not scalable. Therefore, an
application layer mechanism to express support of signalling
compression is needed.
Note that for historical reasons both HTTP and SIP use a
different port for TLS on top of TCP than for TCP alone,
G. Camarillo [Page 3]
Internet Draft SIP October 29, 2002
although at present, this solution is not considered
scalable any longer.
A SIP element that supports compression will need to be prepared to
receive compressed and uncompressed messages on the same port. It
will perform demultiplexing based on the cookie in the topmost bits
of every compressed message.
2 Overview of operation
There are two types of SIP messages; SIP requests and SIP responses.
Clients send SIP requests to the host part of a URI and servers send
responses to the host in the sent-by parameter of the Via header
field.
We define two parameters, one for SIP URIs and the other for the Via
header field. The format of both parameters is the same, as shown in
the examples below:
sip:alice@atlanta.com;comp=sigcomp
Via: SIP/2.0/UDP server1.foo.com:5060;branch=z9hG4bK87a7;comp=sigcomp
The presence of this parameter (comp=sigcomp) in a URI indicates that
the request has to be compressed using SigComp, as defined in [2].
The presence of comp=sigcomp in a Via header field indicates that the
response has to be compressed using SigComp.
Therefore, the presence of comp=sigcomp indicates that the SIP entity
identified by the URI or by the Via header field supports SigComp and
is willing to receive compressed messages. Having comp=sigcomp mean
"willingness" as well as "support" allows the receiver of a SIP
message to influence the decision of whether or not to use SigComp at
a given time.
3 SigComp implementations for SIP
Every SIP implementation that supports SigComp MUST implement the
procedures described in this document.
4 Sending a Request to a Server
A request is sent to the host part of a URI. This URI, referred to as
the next-hop URI, is the Request-URI of the request or an entry in
the Route header field.
If the next-hop URI contains the parameter comp=sigcomp, the client
SHOULD compress the request using SigComp as defined in [2].
G. Camarillo [Page 4]
Internet Draft SIP October 29, 2002
If the next-hop URI is a SIPS URI, the request SHOULD be compressed
before it is passed to the TLS layer.
A client MUST NOT send a compressed request to a server if it does
not know whether or not the server supports SigComp.
Regardless of whether the request is sent compressed or not, if a
client would like to receive subsequent requests within the same
dialog in the UAS->UAC direction compressed, this client SHOULD add
the parameter comp=sigcomp to the URI in the Contact header field if
it is a user agent client. If the client is a proxy, it SHOULD add
the parameter comp=sigcomp to its URI in the Record-Route header
field.
If a user agent client sends a compressed request, it SHOULD add the
parameter comp=sigcomp to the URI in the Contact header field. If a
proxy that Record-Routes sends a compressed request, it SHOULD add
comp=sigcomp to its URI in the Record-Route header field.
If a client sends a compressed request, it SHOULD add the parameter
comp=sigcomp to the topmost entry of the Via header field.
If a client does not know whether or not the server supports SigComp,
but in case the server supported it, it would like to receive
compressed responses, this client SHOULD add the parameter
comp=sigcomp to the topmost entry of the Via header field. The
request, however, as stated above, will not be compressed.
4.1 Obtaining a SIP or SIPS URI with comp=sigcomp
For requests within a dialog, a next-hop URI with the comp=sigcomp
parameter is obtained from a Record-Route header field when the
dialog is established. A client sending a request outside a dialog
can also obtain SIP URIs with comp=sigcomp in a Contact header field
in a 3xx or 485 response to the request.
However, clients establishing a session will not typically be willing
to wait until the dialog is established in order to begin compressing
messages. One of the biggest gains that SigComp can bring to SIP is
the ability to compress the initial INVITE of a dialog, when the user
is waiting for the session to be established. Therefore, clients need
a means to obtain a comp=sigcomp URI from their outbound proxy before
the user decides to establish a session.
One solution to this problem is manual configuration. However,
sometimes it is necessary to have clients configured in an automatic
fashion. Unfortunately, current mechanisms for SIP client
configuration (e.g., using DHCP [5]) do not allow to provide the
G. Camarillo [Page 5]
Internet Draft SIP October 29, 2002
client with URI parameters. In this case, the client SHOULD send an
uncompressed OPTIONS request to its outbound proxy. The outbound
proxy can provide an alternative SIP URI with the comp=sigcomp
parameter in a Contact header field in a 200 OK response to the
OPTIONS. The client can use this URI for subsequent requests that are
sent through the same outbound proxy using compression.
RFC3261 [1] does not define how a proxy should respond to an OPTIONS
request addressed to itself. It only describes how servers respond to
OPTIONS addressed to a particular user. Section 11.2 of RFC3261 says:
Contact header fields MAY be present in a 200 (OK) response
and have the same semantics as in a 3xx response. That is,
they may list a set of alternative names and methods of
reaching the user.
We extend this behavior to proxy servers responding to OPTIONS
addressed to them. They MAY list a set of alternative URIs to contact
the proxy.
Note that receiving incoming requests (even initial INVITEs)
compressed is not a problem, since user agents can REGISTER a SIP URI
with comp=sigcomp in their registrar. All incoming requests for the
user will be sent to this SIP URI using compression.
5 Sending a Response to a Client
A response is sent to the host in the sent-by parameter of the Via
header field. If the topmost Via header field contains the parameter
comp=sigcomp, the response SHOULD be compressed. Otherwise, the
response MUST NOT be compressed.
In order to avoid asymmetric compression (i.e., two SIP entities
exchanging compressed requests in one direction and uncompressed
requests in the other direction) proxies need to rewrite their
Record-Route entries in the responses. A proxy performing Record-
Route inspects the Record-Route header field in the response and the
Contact header field in the request that triggered this response (see
example in Section 9). It looks for the URI of the next upstream
(closer to the user agent client) hop in the route set. If this URI
contains the parameter comp=sigcomp, the proxy SHOULD add
comp=sigcomp to its entry in the Record-Route header field. If this
URI does not contain the parameter comp=sigcomp, the proxy SHOULD
remove comp=sigcomp (if it is present) from its entry in the Record-
Route header field.
The same way, a user agent server SHOULD add comp=sigcomp to the
Contact header field of the response if the URI of the next upstream
G. Camarillo [Page 6]
Internet Draft SIP October 29, 2002
hop in the route set contained the parameter comp=sigcomp.
6 Double Record-Routing
Although proxies usually add zero or one Record-Route entries to a
particular request, some proxies add two of them to avoid Record-
Route rewriting. A typical example of double Record-Routing is a SIP
proxy that acts as a firewall between two networks. Depending on
which network a request comes from, it will be received on a
different interface by the proxy. The proxy adds one Record-Route
entry for one interface and a second one for the other interface.
This way, the proxy does not need to rewrite the Record-Route header
field on the response.
Proxies that receive compressed messages from one side of the dialog
(e.g., upstream) and uncompressed messages from the other side (e.g.,
downstream) MAY use the mechanism described above.
If a proxy detects that the next-hop proxy for a request is the proxy
itself and that the request will not be sent through the network, the
proxy MAY choose not to compress the request even if the URI contains
the comp=sigcomp parameter.
7 Error Situations
If a compressed SIP request arrives to a SIP server that does not
understand SigComp, the server will not have any means to indicate
the error to the client. The message will be impossible to parse, and
there will be no Via header field indicating an address to send the
error response.
If a SIP client sends a compressed request and the client transaction
times out without having received any response, the client SHOULD
retry the same request without using compression. If the compressed
request was sent over a TCP connection, the client SHOULD close that
connection and open a new one to send the uncompressed request.
Otherwise the server would not be able to detect the beginning of the
new message.
8 Augmented BNF
This section provides the augmented Backus-Naur Form (BNF) of both
parameters described above.
The compression URI parameter is a "uri-parameter", as defined by the
SIP ABNF (Section 25.1 of [1]):
G. Camarillo [Page 7]
Internet Draft SIP October 29, 2002
compression-param = "comp=" ("sigcomp" / other-compression)
other-compression = token
The Via compression parameter is a "via-extension", as defined by the
SIP ABNF (Section 25.1 of [1]):
via-compression = "comp" EQUAL ("sigcomp" / other-compression)
other-compression = token
9 Example
The following example illustrates the use of the parameters defined
above. The call flow of Figure 1 shows an INVITE-200 OK-ACK handshake
between a UAC and a UAS through two proxies. Proxy P1 does not
Record-Route but proxy P2 does. Both proxies support compression, but
they do not use it by default.
UAC P1 P2 UAS
|(1)INVITE(c) | | |
|------------>| (2) INVITE | |
| |------------>| (3) INVITE |
| | |------------>|
| | | (4) 200 OK |
| | (5) 200 OK |<------------|
|(6)200 OK(c) |<------------| |
|<------------| | |
| | (7)ACK(c) | |
|-------------------------->| (8) ACK |
| | |------------>|
| | | |
| | | |
Figure 1: INVITE transaction through two proxies
Messages (1), (6) and (7) are compressed (c).
We provide a partial description of the messages involved in this
call flow below. Only some parts of each message are shown, namely
the Method name, the Request-URI and the Via, Route, Record-Route and
Contact header fields. We have not used a correct format for these
header fields. We have rather focus on the contents of the header
fields and on the presence (or absence) of the "comp=sigcomp"
G. Camarillo [Page 8]
Internet Draft SIP October 29, 2002
parameter.
(1) INVITE UAS
Via: UAC;comp=sigcomp
Route:P1;comp=sigcomp
Contact: UAC;comp=sigcomp
P1 is the outbound proxy of the UAC, and it supports SigComp. The UAC
is configured to send compressed traffic to P1, and therefore, it
compresses the INVITE (1). In addition, the UAC wants to receive
future requests and responses for this dialog compressed. Therefore,
it adds the comp=Sigcomp parameter to the Via and to the Contact
header fields.
(2) INVITE UAS
Via: P1
Via: UAC;comp=sigcomp
Route:P2
Contact: UAC;comp=sigcomp
P1 forwards the INVITE (2) to P2. P1 does not use compression by
default, so it sends the INVITE uncompressed to P2.
(3) INVITE UAS
Via: P2
Via: P1
Via: UAC;comp=sigcomp
Record-Route:P2
Contact: UAC;comp=sigcomp
P2 forwards the INVITE (3) to the UAS. P2 supports compression, but
it does not use it by default. Therefore, it sends the INVITE
uncompressed. P2 wishes to remain in the signalling path and
therefore it Record-Routes.
(4) 200 OK
Via: P2
Via: P1
Via: UAC;comp=sigcomp
Record-Route:P2
Contact: UAS
The UAS generates a 200 OK response and sends it to the host in the
topmost Via, which is P2.
G. Camarillo [Page 9]
Internet Draft SIP October 29, 2002
(5) 200 OK
Via: P1
Via: UAC;comp=sigcomp
Record-Route:P2;comp=sigcomp
Contact: UAS
P2 receives the 200 OK response. P2 Record-Routed, so it inspects the
Route set for this dialog. For requests from the UAS towards the UAC
(the opposite direction than the first INVITE), the next hop will be
the Contact header field of the INVITE, because P1 did not Record-
Route. That Contact identified the UAC:
Contact: UAC;comp=sigcomp
Since the UAC wants to receive compressed requests (Contact of the
INVITE), P2 assumes that the UAC would also like to send compressed
requests (Record-Route of the 200 OK). Therefore, P2 modifies its
entry in the Record-Route header field of the 200 OK (5). In the
INVITE (3), P2 did not used the comp=sigcomp parameter. Now it adds
it in the 200 OK (5). This will allow the UAC sending compressed
requests within this dialog.
(6) 200 OK
Via: UAC;comp=sigcomp
Record-Route:P2;comp=sigcomp
Contact: UAS
P1 sends the 200 OK (6) compressed to the UAC because the Via header
field contained the comp=sigcomp parameter.
(7) ACK UAS
Via: UAC;comp=sigcomp
Route:P2;comp=sigcomp
Contact: UAC;comp=sigcomp
The UAC sends the ACK (7) compressed directly to P2 (P1 did not
Record-Route).
(8) ACK UAS
Via: P2
Via: UAC;comp=sigcomp
Contact: UAC;comp=sigcomp
G. Camarillo [Page 10]
Internet Draft SIP October 29, 2002
P2 sends the ACK (8) uncompressed to the UAS.
10 Security Considerations
A SIP entity receiving a compressed message has to decompress it and
to parse it. This requires slightly more processing power than only
parsing a message. This implies that a denial of service attack using
compressed messages would be slightly worse than an attack with
uncompressed messages.
An attacker inserting the parameter comp=sigcomp in a SIP message
could make a SIP entity send compressed messages to another SIP
entity that did not support SigComp. Appropriate integrity mechanisms
should be used to avoid this attack.
11 Acknowledges
Allison Mankin, Jonathan Rosenberg and Miguel Angel Garcia provided
valuable comments on this memo.
12 Authors' Addresses
Gonzalo Camarillo
Ericsson
Advanced Signalling Research Lab.
FIN-02420 Jorvas
Finland
electronic mail: Gonzalo.Camarillo@ericsson.com
13 Normative References
[1] J. Rosenberg, H. Schulzrinne, G. Camarillo, A. Johnston, J.
Peterson, R. Sparks, M. Handley, and E. Schooler, "SIP: session
initiation protocol," RFC 3261, Internet Engineering Task Force, June
2002.
[2] R. Price et al. , "Signaling compression," Internet Draft,
Internet Engineering Task Force, June 2002. Work in progress.
14 Informative References
[3] M. Mealling and R. Daniel, "The naming authority pointer (NAPTR)
DNS resource record," RFC 2915, Internet Engineering Task Force,
Sept. 2000.
[4] A. Gulbrandsen, P. Vixie, and L. Esibov, "A DNS RR for specifying
the location of services (DNS SRV)," RFC 2782, Internet Engineering
Task Force, Feb. 2000.
G. Camarillo [Page 11]
Internet Draft SIP October 29, 2002
[5] H. Schulzrinne, "DHCP option for SIP servers," Internet Draft,
Internet Engineering Task Force, Mar. 2002. Work in progress.
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.
G. Camarillo [Page 12]
