Internet Engineering Task Force SIP WG
Internet Draft J.Rosenberg,H.Schulzrinne
draft-ietf-sip-guidelines-02.txt dynamicsoft,Columbia U.
March 5, 2001
Expires: September 2001
Guidelines for Authors of SIP Extensions
STATUS OF THIS MEMO
This document is an Internet-Draft and is in full conformance with
all provisions of Section 10 of RFC2026.
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Abstract
The Session Initiation Protocol (SIP) is a flexible, yet simple tool
for establishing interactive connections across the Internet. Part of
this flexibility is the ease with which it can be extended. In order
to facilitate effective and interoperable extensions to SIP, some
guidelines need to be followed when developing SIP extensions. This
document outlines a set of such guidelines for authors of SIP
extensions.
1 Terminology
In this document, the key words "MUST", "MUST NOT", "REQUIRED",
"SHALL", "SHALL NOT", "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY",
and "OPTIONAL" are to be interpreted as described in RFC 2119 [1] and
indicate requirement levels for compliant SIP guidelines
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implementations.
2 Introduction
The Session Initiation Protocol (SIP) [2] is a flexible, yet simple
tool for establishing interactive connections across the Internet.
Part of this flexibility is the ease with which it can be extended.
SIP can be extended in numerous ways. New methods, headers, body
types and parameters for existing headers can be defined. This
flexibility also means that caution should be exercised when defining
extensions, in order to ensure interoperability.
In order to facilitate interoperability, this document serves as a
set of guidelines for authors of SIP extensions. It points out issues
to consider when deciding whether a SIP extension is the right answer
for a specific problem. It then points out issues which extensions
should deal with from within their specification. Finally, it
discusses common interactions with existing SIP features which often
cause difficulties in extensions.
3 Should I define a SIP Extension?
The first question to be addressed when defining a SIP extension is:
is a SIP extension the best solution to my problem? SIP has been
proposed as a solution for numerous problems, including mobility,
configuration and management, QoS control, call control, caller
preferences, device control, third party call control, and MPLS path
setup, to name a few. Clearly, not every problem can be solved by a
SIP extension. More importantly, some problems that could be solved
by a SIP extension, probably shouldn't.
To assist engineers in determining whether a SIP extension is an
appropriate solution to their problem, we present two broad criteria.
First, the problem SHOULD fit into the general purvey of SIPs
solution space. Secondly, the solution MUST conform to the general
SIP architectural model.
While the first criteria might seem obvious, we have observed that
numerous extensions to SIP have been proposed because some function
is needed in a device which also speaks SIP. The argument is
generally given that "I'd rather implement one protocol than many".
As an example, user agents, like all other IP hosts, need some way to
obtain their IP address. This is generally done through DHCP [3].
SIPs multicast registration mechanisms might supply an alternate way
to obtain an IP address. This would eliminate the need for DHCP in
clients. However, we do not believe such extensions are appropriate.
We believe that protocols should be defined to provide specific,
narrow functions, rather than being defined based on all
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communications requirements between a pair of devices. The latter
approach to protocol design yields modular protocols with broad
application. It also facilitates extensibility and growth; single
protocols can be removed and changed without affecting the entire
system. We observe that this approach to protocol engineering mirrors
object oriented software engineering.
Our second criteria, that the extension must conform to the general
SIP architectural model, ensures that the protocol remains manageable
and broadly applicable.
3.1 SIP's Solution Space
In order to evaluate the first criteria, it is necessary to define
exactly what SIPs solution space is, and what it is not.
SIP is a protocol for initiating, modifying, and terminating
interactive sessions. This process involves the discovery of users,
(or more generally, entities that can be communicated with, including
services, such as voicemail or translation devices) wherever they may
be located, so that a description of the session can be delivered to
the user. It is assumed that these users or communications entities
are mobile, and their point of attachment to the network changes over
time. The primary purpose of SIP is a rendezvous function, to allow a
request initiator to deliver a message to a recipient wherever they
be. Such rendezvous is needed to establish a session, but can be used
for other purposes related to communications, such as querying for
capabilities or delivery of an instant message.
Much of SIP focuses on this discovery and rendezvous component. Its
ability to fork, its registration capabilities, and its routing
capabilities are all present for the singular purpose of finding the
desired user wherever they may be. As such, features and capabilities
such as personal mobility, automatic call distribution, and follow-me
are well within the SIP solution space.
Session initiation also depends on the ability of the called party to
have enough information about the session itself in order to make a
decision on whether to join or not. That information includes data
about the caller, the purpose for the invitation, and parameters of
the session itself. For this reason, SIP includes this kind of
information.
Part of the process of session initiation is the communication of
progress and the final results of establishment of the session. SIP
provides this information as well.
SIP itself is independent of the session, and the session description
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is delivered as an opaque body within SIP messages. Keeping SIP
independent of the sessions it initiates and terminates is
fundamental. As such, there are many functions that SIP explicitly
does not provide. It is not a session management protocol or a
conference control protocol. The particulars of the communications
within the session are outside of SIP. This includes features such as
media transport, voting and polling, virtual microphone passing,
chairman election, floor control, and feedback on session quality.
SIP is not a resource reservation protocol for sessions. This is
fundamentally because (1) SIP is independent of the underlying
session it establishes, and (2) the path of SIP messages is
completely independent from the path that packets for a session may
take. The path independence refers to paths within a providers
network, and the set of providers itself. For example, it is
perfectly reasonable for a SIP message to traverse a completely
different set of autonomous systems than the audio in a session SIP
establishes.
SIP is not a transfer protocol. It is not meant to send large amounts
of data unrelated to SIPs operation. It is not meant as a replacement
for HTTP. This is for numerous reasons, one of which is that SIP's
recommended mode of operation is over UDP. Sending large messages
over UDP can lead to fragmentation at the IP layer and thus poor
performance in even mildly lossy networks. This is not to say that
carrying payloads in SIP messages is never a good thing; in many
cases, the data is very much related to SIPs operation. However, SIP
is not meant to carry large amounts of data unrelated to SIPs general
function.
The only exception to this rule is REGISTER, which is, in many ways,
its own protocol within SIP. REGISTER is ideally suited for
configuration and exchange of application layer data between a user
agent and its proxy. This may entail exchange of modest amounts of
data. Because of the infrequency of such exchanges and their
limitation in extent (i.e., usually not multi-hop), it is appropriate
to transfer larger amounts of content in REGISTER. In such cases, TCP
is preferred.
SIP is not meant to be a general RPC mechanism. None of its user
discovery and registration capabilities are needed for RPC, neither
are most of its proxy functions. As it is not an ideal transfer
protocol, it is not good at carrying serialized objects of any large
size.
SIP is not meant to be used as a strict PSTN signaling replacement.
It is not a superset of ISUP. While it can support gatewaying of PSTN
signaling, and can provide many features present in the PSTN, the
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mere existence of a feature or capability in the PSTN is not a
justification for its inclusion in SIP. Extensions needed to support
telephony MUST meet the other criteria described here.
SIP is a poor control protocol. It is not meant to be used for one
entity to tell another to pick up or answer a phone, send audio using
a particular codec, or change a configuration parameter. Control
protocols have different trust relationships than is assumed in SIP,
and are more centralized in architecture than SIP, which is a very
distributed protocol.
There are many network layer services needed to make SIP function.
These include quality of service, mobility, and security, among
others. Rather than building these capabilities into SIP itself, they
SHOULD be developed outside of SIP, and then used by it.
Specifically, any protocol mechanisms that are needed by SIP, but are
also needed by many other application layer protocols, SHOULD NOT be
addressed within SIP.
3.2 SIP Architectural Model
We describe here some of the primary architectual assumptions which
underly SIP. Extensions which violate these assumptions should be
examined more carefully to determine their appropriateness for SIP.
Session independence: SIP is independent of the session it
establishes. This includes the type of session, be it
audio, video, game, chat session, or virtual reality. SIP
operation SHOULD NOT be dependent on some characteristic of
the session. SIP is not specific to VoIP only. Any
extensions to SIP MUST consider the application of SIP to a
variety of different session types.
SIP and Session Path Independence: We have already touched on
this once, but it is worth noting again. The set of routers
and/or networks and/or autonomous systems traversed by SIP
messages and the packets in the session are unrelated. They
may be the same in some cases, but it is fundamental to
SIPs architecture that they need not be the same.
Extensions which only work under some assumption of overlap
are not generally applicable to SIPs operation and should
be scrutinized carefully.
Multi-provider and Multi-hop: SIP assumes that its messages will
traverse the Internet. That is, SIP works through multiple
networks administered by different providers. It is also
assumed that SIP messages traverse many hops (where each
hop is a proxy). Extensions SHOULD NOT work only under the
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assumption of a single hop or single provider.
Transactional: SIP is a request/response protocol, possibly
enhanced with intermediate responses. Many of the rules of
operation in SIP are based on general processing of
requests and responses. This includes the reliability
mechanisms, routing mechanisms, and state maintenance
rules. Extensions SHOULD NOT add messages that are not
within the request-response model.
Proxies can ignore bodies: In order for proxies to scale well,
they must be able to operate with minimal message
processing. SIP has been engineered so that proxies can
always ignore bodies. Extensions SHOULD NOT require proxies
to examine bodies.
Proxies don't need to understand the method: Processing of
requests in proxies does not depend on the method, except
for the well known methods INVITE, ACK, and CANCEL. This
allows for extensibility. Extensions MUST NOT define new
methods which must be understood by proxies.
INVITE messages carry full state: An initial INVITE message for
a session is nearly identical (the exception is the tag) to
a re-INVITE message to modify some characteristic of the
session. This soft-state property is fundamental to SIP,
and is critical for robustness of SIP systems. Extensions
SHOULD NOT modify INVITE processing such that data spanning
multiple INVITEs must be collected in order to perform some
feature.
Generality over efficiency: Wherever possible, SIP has favored
general purpose components rather than narrow ones. If some
capability is added to support one service, but a slightly
broader capability can support a larger variety of services
(at the cost of complexity or message sizes), the broader
capability SHOULD be preferred.
The Request URI is the primary key for routing: Forwarding logic
at SIP servers depends primarily on the request URI. It is
fundamental to the operation of SIP that the request URI
indicate a resource that, under normal operations, resolves
to the desired recipient. Extensions SHOULD NOT use other
components of the SIP message as the primary routing key,
and SHOULD NOT modify the semantics of the request URI.
Proxies can operate statelessly: SIP allows for great
flexibility in the design of proxies. They can operate in
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fast, stateless modes, or they can maintain complete call
and session state, providing advanced services. SIP
extensions SHOULD insure that such a range of servers can
always be built. Therefore, extensions which SHOULD NOT be
defined which operate only with stateful proxies.
Heterogeneity is the norm: SIP supports hetereogeneous devices.
It has built in mechanisms for determining the set of
overlapping protocol functionalities. Extensions SHOULD NOT
be defined which only function if all devices support the
extension.
4 Issues to be Addressed
Given an extension has met the litmus tests in the previous section,
there are several issues that all extensions should take into
consideration.
4.1 Backwards Compatibility
One of the most important issues to consider is whether the new
extension is backwards compatible with baseline SIP. This is tightly
coupled with how the Require, Proxy-Require, and Supported [4]
headers are used.
If an extension consists of new headers inserted by a user agent in a
request with an existing method, and the request cannot be processed
reasonably by a proxy and/or user agent without understanding the
headers, the extension MUST mandate the usage of the Require and/or
Proxy-Require headers in the request. These extensions are not
backwards compatible with SIP. The result of mandating usage of these
headers means that requests cannot be serviced unless the entities
being communicated with also understand the extension. If some entity
does not understand the extension, the request will be rejected. The
UAC can then handle this in one of two ways. In the first, the
request simply fails, and the service cannot be provided. This is
basically an interoperability failure. In the second case, the UAC
retries the request without the extension. This will preserve
interoperability, at the cost of a "dual stack" implementation in a
UAC (processing rules for operation with and without the extension).
As the number of extensions increases, this leads to an exponential
explosion in the sets of processing rules a UAC may need to
implement. The result is excessive complexity.
Because of the possibility of interoperability and complexity
problems that result from the usage of Require and Proxy-Require, we
believe the following guidelines are appropriate:
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o The usage of these headers in requests for basic SIP services
(in particular, session initiation and termination) is NOT
RECOMMENDED. The less frequently a particular extension is
needed in a request, the more reasonable it is to use these
headers.
o The Proxy-Require header SHOULD be avoided at all costs. The
failure likelihood in an individual proxy stays constant, but
the path failure grows exponentially with the number of hops.
On the other hand, the Require header only mandates that a
single entity, the UAS, support the extension. Usage of
Proxy-Require is thus considered exponentially worse than
usage of the Require header.
Extensions which define new methods do not need to use the Require
header. SIP defines mechanisms which allow a UAC to know whether a
new method is understood by a UAS. This includes both the OPTIONS
request, and the 405 (Method Not Allowed) response with the Allow
header. It is fundamental to SIP that proxies do not need to
understand the semantics of a new method in order to process it. If
an extension defines a new method which must be understood by proxies
in order to be processed, a Proxy-Require header is needed. As
discussed above, these kinds of extensions are frowned upon.
In order to achieve backwards compatibility for extensions that
define new methods, the Allow header is used. There are two types of
new methods - those that are used for established sessions (initiated
by INVITE, for example), and those that are sent as the initial
request to a UA. Since INVITE and its response both SHOULD contain an
Allow header, a UA can readily determine whether the new method can
be supported within the call. For example, if a new method for a
mid-call feature, such as hold, were to be defined, the hold button
on the UI could be "greyed out" once the call is established, if the
new method were not listed in the Allow header.
Another type of extension are those which require a proxy to insert
headers into a request as it traverses the network, or for the UAS to
insert headers into a response. For some extensions, if the UAC or
UAS does not understand these headers, the message can still be
processed correctly. These extensions are completely backwards
compatible.
Most other extensions of this type require that the server only
insert the header if it is sure the client understands it. In this
case, these extensions will need to make use of the Supported request
header mechanism. This mechanism allows a server to determine if the
client can understand some extension, so that it can apply the
extension to the response [4]. By their nature, these extensions may
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not always be able to be applied to every response.
If an extension requires a proxy to insert a header into a request,
and this header needs to be understood by both UAC and UAS to be
executed correctly, a combination of the Require and the Supported
mechanism will need to be used. The proxy can insert a Require header
into the request, given the Supported header is present. An example
of such an extension is the SIP Session Timer [5].
Yet another type of extension is that which defines new body types to
be carried in SIP messages. According to the SIP specification,
bodies must be understood in order to process a request. As such, the
interoperability issues are similar to new methods. However, a new
header, Content-Disposition, has been defined that allows a client or
server to indicate that the message body is optional [6]. Usage of
optional bodies, as opposed to mandatory ones, is RECOMMENDED
wherever possible.
When a body must be understood to properly process a request or
response, it is preferred that the sending entity know ahead of time
whether the new body is understood by the recipient. For requests
that are the first in a sequence of exchanges between user agents
(such as INVITE), inclusion of Accept in the request and its success
responses is RECOMMENDED. This will allow both parties to determine
what body types are supported by their peers. Subsequent messaging
between the peers would then only include body types that were
indicated as being understood.
4.2 Security
Security is an important component of any protocol. SIP extensions
SHOULD consider how (or if) they affect usage of the general SIP
security mechanisms. Most extensions should not require any new
security capabilities beyond general purpose SIP. If they do, it is
likely that the security mechanism has more general purpose
application, and should be considered an extension in its own right.
4.3 Usage Guidelines
All SIP extensions must contain guidelines defining when the
extension is to be used.
For extensions that define new headers, the extension MUST define the
request methods the header can appear in, and what responses it can
be used in. It is recommended that this information be represented as
a new row of Table 4 of RFC 2543 [2]. The extension SHOULD specify
which entities (UAC, UAS, proxy, redirect, registrar) are allowed to
insert the header.
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4.4 Syntactic Issues
Extensions that define new methods SHOULD use all capitals for the
method name. Method names SHOULD be less than 10 characters, and
SHOULD attempt to convey the general meaning of the request.
Extensions that define new headers SHOULD define a compact form
representation if the non-compact header is more than four
characters. Compact headers MUST be a single character. When all 26
characters are exhausted, new compact forms will no longer be
defined. Header names SHOULD use ASCII characters. Header names are
always case insensitive. Header values are generally case sensitive,
with the exception of domain names which MUST be case insensitive.
Case sensitivity of parameters and values is a constant source of
confusion. SIP extensions MUST clearly indicate the case sensitivity
or insensitivity of every parameter, value or field they define. In
general, case sensitivity is preferred because of the reduced
processing requirements.
Extensions which contain freeform text MUST allow that text to be
UTF-8, as per the IETF policies on character set usage [7]. This
ensures that SIP remains an internationalized standard. As a general
guideline, freeform text is never needed by programs in order to
perform protocol processing. It is usually entered by and displayed
to the user. If an extension uses a parameter which can contain UTF-8
encoded characters, and that extension requires a comparison to be
made of this parameter to other parameters, the comparison SHOULD be
case sensitive. Case insensitive comparison rules for UTF-8 text are
extremely complicated and are to be avoided.
Extensions which make use of dates and times MUST use the SIP-Date
BNF defined in RFC 2543. No other date formats are allowed.
Extensions which include network layer addresses SHOULD permit dotted
quad IPv4 addresses, IPv6 addresses in the format described in [8],
and domain names.
Extensions which have headers containing URLs SHOULD allow any URI,
not just SIP URLs.
Headers SHOULD follow the standard formatting for SIP, defined as:
header-name ":" # (value *( ";" parameter-name ["=" token] ) |
";" parameter-name ["=" quoted-string] ))
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Developers of extensions SHOULD allow for extension parameters in
their headers.
Headers that contain a list of URIs SHOULD follow the same syntax as
the Contact header in SIP. Implementors are also encouraged to always
wrap these URI in angle brackets "<" and ">". We have found this to
be a frequently misimplemented feature.
Beyond compact form, there is no need to define compressed versions
of header values. Compression of SIP messages SHOULD be handled at
lower layers, for example, using IP payload compression [9] or link
layer compression.
Syntax for headers is expressed in Augmented Backus-Naur Form.
Extensions MUST make use of the primitive components defined in
RFC2543 [2]. If the construction for a BNF element is defined in
another specification, it is RECOMMENDED that the construction be
referenced rather than copied. The reference SHOULD include both the
document and section number. All BNF elements must be either defined
or referenced.
All tokens and quoted strings are separated by implicit linear white
space. Linear white space, for better or worse, allows for line
folding. Extensions cannot define new headers that use alternate
linear white space rules.
4.5 Semantics, Semantics, Semantics
Developers of protocols often get caught up in syntax issues, without
spending enough time on semantics. The semantics of a protocol are
far more important. SIP extensions MUST clearly define the semantics
of the extensions. Specifically, the extension MUST specify the
behaviors expected of a UAC, UAS and proxy in processing the
extension. This is often best described by having separate sections
for each of these three elements. Each section SHOULD step through
the processing rules in temporal order of the most common messaging
scenario.
Processing rules generally specify actions to take (in terms of
messages to send, variables to store, rules to follow) on receipt of
messages and expiration of timers. If an action requires transmission
of a message, the rule SHOULD outline requirements for insertion of
headers or other information in the message.
The extension SHOULD specify procedures to take in exceptional
conditions. This usually includes receipt of messages that are not
expected, expiration of timers that handle timeouts, and presence of
headers in messages when they are not expected.
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4.6 Examples Section
Presence of sections in the extension giving examples of call flows
and message formatting is RECOMMENDED. Extensions which define
substantial new syntax SHOULD include examples of messages containing
that syntax. Examples of message flows should be given to cover
common cases and at least one failure or unusual case.
For an example of how to construct a good examples section, see the
message flows and message formatting defined in the Call Flows
Example specification [10]. Note that complete messages SHOULD be
used. Be careful to include tags, Via headers, Content-Lengths,
Record-Route and Route headers.
4.7 Overview Section
Too often, extension documents dive into detailed syntax and
semantics without giving a general overview of operation. This makes
understanding of the extension harder. It is RECOMMENDED that
extensions have a protocol overview section which discusses the basic
operation of the extension. Basic operation usually consists of the
message flow, in temporal order, for the most common case covered by
the extension. The most important processing rules for the elements
in the call flow SHOULD be mentioned. Usage of the RFC 2119 [1]
terminology in the overview section is RECOMMENDED.
4.8 Additional Considerations for New Methods
Extensions which define new methods SHOULD take into consideration,
and discuss, the following issues:
o Can it contain bodies? If so, what is the meaning of the
presence of those bodies? What body types are allowed?
o Can a transaction with this request method occur while another
transaction, in the same and/or reverse direction, is in
progress?
o What headers are allowed in requests of this method? It is
RECOMMENDED that this information be presented through a
column of Table 4 in RFC 2543 [2].
o All SIP requests can generally be cancelled. However, an
extension MAY mandate that a new method cannot be cancelled.
In either case, handling of CANCEL SHOULD be described. In
particular, the rules a UAS should follow upon cancellation of
an unanswered request SHOULD be described.
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o Can the request be sent within a call or not? In this context,
within means that the request is sent with the same Call-ID,
To and From field as an INVITE that was sent or received
previously. For, example, the REGISTER method is not
associated with a call, whereas the BYE method is.
Note that the reliability mechanisms for all new methods must be the
same as for BYE. The delayed response feature of INVITE is only
available in INVITE, never for new methods. This means requests with
new SIP methods need to be responded to within short time periods (on
the order of seconds).
4.9 Additional Considerations for New Headers or Header Parameters
The most important issue for extensions that define new headers is
backwards compatibility. See Section 4.1 for a discussion of the
issues. The extension MUST detail how backwards compatibility is
addressed.
It is often tempting to avoid creation of a new method by overloading
an existing method through a header. Headers are not meant to
fundamentally alter the meaning of the method of the request. A new
header cannot change the basic semantic and processing rules of a
method. There is no shortage of method names, so when an extension
changes the basic meaning of a request, a new method SHOULD be
defined.
4.10 Additional Considerations for New Body Types
Because SIP can run over UDP, extensions that specify the inclusion
of large bodies are frowned upon. If at all possible, the content
SHOULD be included indirectly through an http URL.
Note that the presence of a body MUST NOT change the nature of the
message. That is, bodies cannot alter the state machinery associated
with processing a request of a particular method or a response.
Bodies enhance this processing by providing additional data.
5 Interactions with SIP Features
We have observed that certain capabilities of SIP continually
interact with extensions in unusual ways. Writers of extensions
SHOULD consider the interactions of their extensions with these SIP
capabilities, document any unusual interactions if they exist. The
most common causes of problems are:
Forking: Forking by far presents the most troublesome
interactions with extensions. This is generally because it
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can cause (1) a single transmitted request to be received
by an unknown number of UASs, and (2) a single request to
have multiple responses.
Tags: Tags are used to uniquely identify call legs. Their
presence is neccesitated as a result of forking. They are
an unfortunate exception to many SIP processing rules.
Extensions SHOULD carefully consider their effect.
CANCEL and ACK: CANCEL and ACK are "special" SIP requests, in
that they are exceptions to many of the general request
processing rules. The main reason for this special status
is that CANCEL and ACK are always associated with another
request. New methods SHOULD consider the meaning of
cancellation. Extensions which defined new headers in
INVITE requests SHOULD consider whether they also need to
be included in ACK.
Routing: The Route, Record-Route and Via headers are used to
support message routing. New request methods SHOULD
carefully consider how these headers are used.
Stateless Proxies: SIP allows a proxy to be stateless. Stateless
proxies are unable to retransmit messages and cannot
execute certain services. Extensions which depend on some
kind of proxy processing SHOULD consider how stateless
proxies affect that processing.
6 Security Considerations
The nature of this document is such that it does not introduce any
new security considerations.
7 Changes since -01
o Return to rfc2119 strength wording.
8 Authors Addresses
Jonathan Rosenberg
dynamicsoft
72 Eagle Rock Avenue
East Hanover, NJ 07936
email: jdrosen@dynamicsoft.com
Henning Schulzrinne
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Columbia University
M/S 0401
1214 Amsterdam Ave.
New York, NY 10027-7003
email: schulzrinne@cs.columbia.edu
9 Bibliography
[1] S. Bradner, "Key words for use in RFCs to indicate requirement
levels," Request for Comments 2119, Internet Engineering Task Force,
Mar. 1997.
[2] M. Handley, H. Schulzrinne, E. Schooler, and J. Rosenberg, "SIP:
session initiation protocol," Request for Comments 2543, Internet
Engineering Task Force, Mar. 1999.
[3] R. Droms, "Dynamic host configuration protocol," Request for
Comments 2131, Internet Engineering Task Force, Mar. 1997.
[4] J. Rosenberg and H. Schulzrinne, "The SIP supported header,"
Internet Draft, Internet Engineering Task Force, Mar. 2000. Work in
progress.
[5] S. Donovan and J. Rosenberg, "SIP session timer," Internet Draft,
Internet Engineering Task Force, Oct. 2000. Work in progress.
[6] M. Handley, H. Schulzrinne, E. Schooler, and J. Rosenberg, "SIP:
Session initiation protocol," Internet Draft, Internet Engineering
Task Force, Aug. 2000. Work in progress.
[7] H. Alvestrand, "IETF policy on character sets and languages,"
Request for Comments 2277, Internet Engineering Task Force, Jan.
1998.
[8] R. Hinden, B. Carpenter, and L. Masinter, "Format for literal
IPv6 addresses in URL's," Request for Comments 2732, Internet
Engineering Task Force, Dec. 1999.
[9] A. Shacham, R. Monsour, R. Pereira, and M. Thomas, "IP payload
compression protocol (ipcomp)," Request for Comments 2393, Internet
Engineering Task Force, Dec. 1998.
[10] A. Johnston, S. Donovan, R. Sparks, C. Cunningham, D. Willis, J.
Rosenberg, K. Summers, and H. Schulzrinne, "SIP telephony call flow
examples," Internet Draft, Internet Engineering Task Force, July
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Internet Draft guidelines March 5, 2001
2000. Work in progress.
J.Rosenberg,H.Schulzrinne [Page 16]