| Rfc | 5411 |
|---|
| Title | A Hitchhiker's Guide to the Session Initiation Protocol (SIP) |
|---|
| Author | J.
Rosenberg |
|---|
| Date | February 2009 |
|---|
| Format: | TXT, HTML |
|---|
| Status: | INFORMATIONAL |
|---|
|
Network Working Group J. Rosenberg
Request for Comments: 5411 Cisco
Category: Informational January 2009
A Hitchhiker's Guide to the Session Initiation Protocol (SIP)
Status of This Memo
This memo provides information for the Internet community. It does
not specify an Internet standard of any kind. Distribution of this
memo is unlimited.
Abstract
The Session Initiation Protocol (SIP) is the subject of numerous
specifications that have been produced by the IETF. It can be
difficult to locate the right document, or even to determine the set
of Request for Comments (RFC) about SIP. This specification serves
as a guide to the SIP RFC series. It lists a current snapshot of the
specifications under the SIP umbrella, briefly summarizes each, and
groups them into categories.
Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . . 2
2. Scope of This Document . . . . . . . . . . . . . . . . . . . . 4
3. Core SIP Specifications . . . . . . . . . . . . . . . . . . . 5
4. Public Switched Telephone Network (PSTN) Interworking . . . . 8
5. General Purpose Infrastructure Extensions . . . . . . . . . . 10
6. NAT Traversal . . . . . . . . . . . . . . . . . . . . . . . . 12
7. Call Control Primitives . . . . . . . . . . . . . . . . . . . 13
8. Event Framework . . . . . . . . . . . . . . . . . . . . . . . 14
9. Event Packages . . . . . . . . . . . . . . . . . . . . . . . . 15
10. Quality of Service . . . . . . . . . . . . . . . . . . . . . . 16
11. Operations and Management . . . . . . . . . . . . . . . . . . 17
12. SIP Compression . . . . . . . . . . . . . . . . . . . . . . . 17
13. SIP Service URIs . . . . . . . . . . . . . . . . . . . . . . . 17
14. Minor Extensions . . . . . . . . . . . . . . . . . . . . . . . 19
15. Security Mechanisms . . . . . . . . . . . . . . . . . . . . . 20
16. Conferencing . . . . . . . . . . . . . . . . . . . . . . . . . 23
17. Instant Messaging, Presence, and Multimedia . . . . . . . . . 24
18. Emergency Services . . . . . . . . . . . . . . . . . . . . . . 25
19. Security Considerations . . . . . . . . . . . . . . . . . . . 25
20. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 25
21. Informative References . . . . . . . . . . . . . . . . . . . . 26
1. Introduction
The Session Initiation Protocol (SIP) [RFC3261] is the subject of
numerous specifications that have been produced by the IETF. It can
be difficult to locate the right document, or even to determine the
set of Request for Comments (RFC) about SIP. "Don't Panic!" [HGTTG]
This specification serves as a guide to the SIP RFC series. It is a
current snapshot of the specifications under the SIP umbrella at the
time of publication. It is anticipated that this document itself
will be regularly updated as SIP specifications mature. Furthermore,
it references many specifications, which, at the time of publication
of this document, were not yet finalized, and may eventually be
completed or abandoned. Therefore, the enumeration of specifications
here is a work-in-progress and subject to change.
For each specification, a paragraph or so description is included
that summarizes the purpose of the specification. Each specification
also includes a letter that designates its category in the Standards
Track [RFC2026]. These values are:
S: Standards Track (Proposed Standard, Draft Standard, or Standard)
E: Experimental
B: Best Current Practice
I: Informational
The specifications are grouped together by topic. The topics are:
Core: The SIP specifications that are expected to be utilized for
each session or registration an endpoint participates in.
Public Switched Telephone Network (PSTN) Interop: Specifications
related to interworking with the telephone network.
General Purpose Infrastructure: General purpose extensions to SIP,
SDP (Session Description Protocol), and MIME, but ones that are
not expected to always be used.
NAT Traversal: Specifications to deal with firewall and NAT
traversal.
Call Control Primitives: Specifications for manipulating SIP dialogs
and calls.
Event Framework: Definitions of the core specifications for the SIP
event framework, providing for pub/sub capability.
Event Packages: Packages that utilize the SIP event framework.
Quality of Service: Specifications related to multimedia quality of
service (QoS).
Operations and Management: Specifications related to configuration
and monitoring of SIP deployments.
SIP Compression: Specifications to facilitate usage of SIP with the
Signaling Compression (Sigcomp) framework.
SIP Service URIs: Specifications on how to use SIP URIs to address
multimedia services.
Minor Extensions: Specifications that solve a narrow problem space
or provide an optimization.
Security Mechanisms: Specifications providing security functionality
for SIP.
Conferencing: Specifications for multimedia conferencing.
Instant Messaging, Presence, and Multimedia: SIP extensions related
to IM, presence, and multimedia. This covers only the SIP
extensions related to these topics. See [SIMPLE] for a full
treatment of SIP for IM and Presence (SIMPLE).
Emergency Services: SIP extensions related to emergency services.
See [ECRIT-FRAME] for a more complete treatment of additional
functionality related to emergency services.
Typically, SIP extensions fit naturally into topic areas, and
implementors interested in a particular topic often implement many or
all of the specifications in that area. There are some
specifications that fall into multiple topic areas, in which case
they are listed more than once.
Do not print all the specs cited here at once, as they might share
the fate of the rules of Brockian Ultracricket when bound together:
collapse under their own gravity and form a black hole [HGTTG].
This document itself is not an update to RFC 3261 or an extension to
SIP. It is an informational document, meant to guide newcomers,
implementors, and deployers to the many specifications associated
with SIP.
2. Scope of This Document
It is very difficult to enumerate the set of SIP specifications.
This is because there are many protocols that are intimately related
to SIP and used by nearly all SIP implementations, but are not
formally SIP extensions. As such, this document formally defines a
"SIP specification" as:
o RFC 3261 and any specification that defines an extension to it,
where an extension is a mechanism that changes or updates in some
way a behavior specified there.
o The basic SDP specification [RFC4566] and any specification that
defines an extension to SDP whose primary purpose is to support
SIP.
o Any specification that defines a MIME object whose primary purpose
is to support SIP.
Excluded from this list are requirements, architectures, registry
definitions, non-normative frameworks, and processes. Best Current
Practices are included when they normatively define mechanisms for
accomplishing a task, or provide significant description of the usage
of the normative specifications, such as call flows.
The SIP change process [RFC3427] defines two types of extensions to
SIP: normal extensions and the so-called P-headers (where P stands
for "preliminary", "private", or "proprietary", and the "P-" prefix
is included in the header field name), which are meant to be used in
areas of limited applicability. P-headers cannot be defined in the
Standards Track. For the most part, P-headers are not included in
the listing here, with the exception of those that have seen general
usage despite their P-header status.
This document includes specifications, which have already been
approved by the IETF and granted an RFC number, in addition to
Internet Drafts, which are still under development within the IETF
and will eventually finish and get an RFC number. Inclusion of
Internet Drafts here helps encourage early implementation and
demonstrations of interoperability of the protocol, and thus aids in
the standards-setting process. Inclusion of these also identifes
where the IETF is targetting a solution at a particular problem
space. Note that final IANA assignment of codepoints (such as option
tags and header field names) does not take place until shortly before
publication as an RFC, and thus codepoint assignments may change.
3. Core SIP Specifications
The core SIP specifications represent the set of specifications whose
functionality is broadly applicable. An extension is broadly
applicable if it fits into one of the following categories:
o For specifications that impact SIP session management, the
extension would be used for almost every session initiated by a
user agent.
o For specifications that impact SIP registrations, the extension
would be used for almost every registration initiated by a user
agent.
o For specifications that impact SIP subscriptions, the extension
would be used for almost every subscription initiated by a user
agent.
In other words, these are not specifications that are used just for
some requests and not others; they are specifications that would
apply to each and every request for which the extension is relevant.
In the galaxy of SIP, these specifications are like towels [HGTTG].
RFC 3261, The Session Initiation Protocol (S): [RFC3261] is the core
SIP protocol itself. RFC 3261 obsoletes [RFC2543]. It is the
president of the galaxy [HGTTG] as far as the suite of SIP
specifications is concerned.
RFC 3263, Locating SIP Servers (S): [RFC3263] provides DNS
procedures for taking a SIP URI and determining a SIP server that
is associated with that SIP URI. RFC 3263 is essential for any
implementation using SIP with DNS. RFC 3263 makes use of both DNS
SRV records [RFC2782] and NAPTR records [RFC3401].
RFC 3264, An Offer/Answer Model with the Session Description Protocol
(S): [RFC3264] defines how the Session Description Protocol (SDP)
[RFC4566] is used with SIP to negotiate the parameters of a media
session. It is in widespread usage and an integral part of the
behavior of RFC 3261.
RFC 3265, SIP-Specific Event Notification (S): [RFC3265] defines the
SUBSCRIBE and NOTIFY methods. These two methods provide a general
event notification framework for SIP. To actually use the
framework, extensions need to be defined for specific event
packages. An event package defines a schema for the event data
and describes other aspects of event processing specific to that
schema. An RFC 3265 implementation is required when any event
package is used.
RFC 3325, Private Extensions to SIP for Asserted Identity within
Trusted Networks (I): Though its P-header status implies that it has
limited applicability, [RFC3325], which defines the P-Asserted-
Identity header field, has been widely deployed. It is used as
the basic mechanism for providing network-asserted caller ID
services. Its intended update, [UPDATE-PAI], clarifies its usage
for connected party identification as well.
RFC 3327, SIP Extension Header Field for Registering Non-Adjacent
Contacts (S): [RFC3327] defines the Path header field. This field
is inserted by proxies between a client and their registrar. It
allows inbound requests towards that client to traverse these
proxies prior to being delivered to the user agent. It is
essential in any SIP deployment that has edge proxies, which are
proxies between the client and the home proxy or SIP registrar.
RFC 3581, An Extension to SIP for Symmetric Response Routing (S):
[RFC3581] defines the rport parameter of the Via header. It
allows SIP responses to traverse NAT. It is one of several
specifications that are utilized for NAT traversal (see
Section 6).
RFC 3840, Indicating User Agent Capabilities in SIP (S): [RFC3840]
defines a mechanism for carrying capability information about a
user agent in REGISTER requests and in dialog-forming requests
like INVITE. It has found use with conferencing (the isfocus
parameter declares that a user agent is a conference server) and
with applications like push-to-talk.
RFC 4320, Actions Addressing Issues Identified with the Non-INVITE
Transaction in SIP (S): [RFC4320] formally updates RFC 3261 and
modifies some of the behaviors associated with non-INVITE
transactions. This addresses some problems found in timeout and
failure cases.
RFC 4474, Enhancements for Authenticated Identity Management in SIP
(S): [RFC4474] defines a mechanism for providing a cryptographically
verifiable identity of the calling party in a SIP request. Known
as "SIP Identity", this mechanism provides an alternative to RFC
3325. It has seen little deployment so far, but its importance as
a key construct for anti-spam techniques and new security
mechanisms makes it a core part of the SIP specifications.
GRUU, Obtaining and Using Globally Routable User Agent Identifiers
(GRUU) in SIP (S): [GRUU] defines a mechanism for directing requests
towards a specific UA instance. GRUU is essential for features
like transfer and provides another piece of the SIP NAT traversal
story.
OUTBOUND, Managing Client Initiated Connections through SIP (S):
[OUTBOUND], also known as SIP outbound, defines important changes
to the SIP registration mechanism that enable delivery of SIP
messages towards a UA when it is behind a NAT. This specification
is the cornerstone of the SIP NAT traversal strategy.
RFC 4566, Session Description Protocol (S): [RFC4566] defines a
format for representing multimedia sessions. SDP objects are
carried in the body of SIP messages and, based on the offer/answer
model, are used to negotiate the media characteristics of a
session between users.
SDP-CAP, SDP Capability Negotiation (S): [SDP-CAP] defines a set of
extensions to SDP that allows for capability negotiation within
SDP. Capability negotiation can be used to select between
different profiles of RTP (secure vs. unsecure) or to negotiate
codecs such that an agent has to select one amongst a set of
supported codecs.
ICE, Interactive Connectivity Establishment (ICE) (S): [ICE] defines
a technique for NAT traversal of media sessions for protocols that
make use of the offer/answer model. This specification is the
IETF-recommended mechanism for NAT traversal for SIP media
streams, and is meant to be used even by endpoints that are
themselves never behind a NAT. A SIP option tag and media feature
tag [OPTION-TAG] (also a core specification) have been defined for
use with ICE.
RFC 3605, Real Time Control Protocol (RTCP) Attribute in the Session
Description Protocol (SDP) (S): [RFC3605] defines a way to
explicitly signal, within an SDP message, the IP address and port
for RTCP, rather than using the port+1 rule in the Real Time
Transport Protocol (RTP) [RFC3550]. It is needed for devices
behind NAT, and the specification is required by ICE.
RFC 4916, Connected Identity in the Session Initiation Protocol (SIP)
(S): [RFC4916] formally updates RFC 3261. It defines an extension
to SIP that allows a calling user to determine the identity of the
final called user (connected party). Due to forwarding and
retargeting services, this may not be the same as the user that
the caller was originally trying to reach. The mechanism works in
tandem with the SIP identity specification [RFC4474] to provide
signatures over the connected party identity. It can also be used
if a party identity changes mid-call due to third-party call
control actions or PSTN behavior.
RFC 3311, The SIP UPDATE Method (S): [RFC3311] defines the UPDATE
method for SIP. This method is meant as a means for updating
session information prior to the completion of the initial INVITE
transaction. It can also be used to update other information,
such as the identity of the participant [RFC4916], without
involving an updated offer/answer exchange. It was developed
initially to support [RFC3312], but has found other uses. In
particular, its usage with RFC 4916 means it will typically be
used as part of every session, to convey a secure, connected
identity.
SIPS-URI, The Use of the SIPS URI Scheme in the Session Initiation
Protocol (SIP) (S): [SIPS-URI] is intended to update RFC 3261. It
revises the processing of the SIPS URI, originally defined in RFC
3261, to fix many errors and problems that have been encountered
with that mechanism.
RFC 3665, Session Initiation Protocol (SIP) Basic Call Flow Examples
(B): [RFC3665] contains best-practice call flow examples for basic
SIP interactions -- call establishment, termination, and
registration.
Essential Corrections to SIP: A collection of fixes to SIP that
address important bugs and vulnerabilities. These include a fix
requiring loop detection in any proxy that forks [LOOP-FIX], a
clarification on how record-routing works [RECORD-ROUTE], and a
correction to the IPv6 BNF [ABNF-FIX].
4. Public Switched Telephone Network (PSTN) Interworking
Numerous extensions and usages of SIP are related to interoperability
and communications with or through the PSTN.
RFC 2848, The PINT Service Protocol (S): [RFC2848] is one of the
earliest extensions to SIP. It defines procedures for using SIP
to invoke services that actually execute on the PSTN. Its main
application is for third-party call control, allowing an IP host
to set up a call between two PSTN endpoints. PINT (PSTN/Internet
Interworking) has a relatively narrow focus and has not seen
widespread deployment.
RFC 3910, The SPIRITS Protocol (S): Continuing the trend of naming
PSTN-related extensions with alcohol references, SPIRITS (Services
in PSTN Requesting Internet Services) [RFC3910] defines the
inverse of PINT. It allows a switch in the PSTN to ask an IP
element how to proceed with call waiting. It was developed
primarily to support Internet Call Waiting (ICW). Perhaps the
next specification will be called the Pan Galactic Gargle Blaster
[HGTTG].
RFC 3372, SIP for Telephones (SIP-T): Context and Architectures (I):
SIP-T [RFC3372] defines a mechanism for using SIP between pairs of
PSTN gateways. Its essential idea is to tunnel ISDN User Part
(ISUP) signaling between the gateways in the body of SIP messages.
SIP-T motivated the development of INFO [RFC2976]. SIP-T has seen
widespread implementation for the limited deployment model that it
addresses. As ISUP endpoints disappear from the network, the need
for this mechanism will decrease.
RFC 3398, ISUP to SIP Mapping (S): [RFC3398] defines how to do
protocol mapping from the SS7 ISDN User Part (ISUP) signaling to
SIP. It is widely used in SS7 to SIP gateways and is part of the
SIP-T framework.
RFC 4497, Interworking between the Session Initiation Protocol (SIP)
and QSIG (B): [RFC4497] defines how to do protocol mapping from
Q.SIG, used for Private Branch Exchange (PBX) signaling, to SIP.
RFC 3578, Mapping of ISUP Overlap Signaling to SIP (S): [RFC3578]
defines a mechanism to map overlap dialing into SIP. This
specification is widely regarded as the ugliest SIP specification,
as the introduction to the specification itself advises that it
has many problems. Overlap signaling (the practice of sending
digits into the network as dialed instead of waiting for complete
collection of the called party number) is largely incompatible
with SIP at some fairly fundamental levels. That said, RFC 3578
is mostly harmless and has seen some usage.
RFC 3960, Early Media and Ringtone Generation in SIP (I): [RFC3960]
defines some guidelines for handling early media -- the practice
of sending media from the called party or an application server
towards the caller prior to acceptance of the call. Early media
is often generated from the PSTN. Early media is a complex topic,
and this specification does not fully address the problems
associated with it.
RFC 3959, Early Session Disposition Type for the Session Initiation
Protocol (SIP) (S): [RFC3959] defines a new session disposition type
for use with early media. It indicates that the SDP in the body
is for a special early media session. This has seen little usage.
RFC 3204, MIME Media Types for ISUP and QSIG Objects (S): [RFC3204]
defines MIME objects for representing SS7 and QSIG signaling
messages. SS7 signaling messages are carried in the body of SIP
messages when SIP-T is used. QSIG signaling messages can be
carried in a similar way.
RFC3666, Session Initiation Protocol (SIP) Public Switched Telephone
Network (PSTN) Call Flows (B): [RFC3666] provides best practice call
flows around interworking with the PSTN.
5. General Purpose Infrastructure Extensions
These extensions are general purpose enhancements to SIP, SDP, and
MIME that can serve a wide variety of uses. However, they are not
used for every session or registration, as the core specifications
are.
RFC 3262, Reliability of Provisional Responses in SIP (S): SIP
defines two types of responses to a request: final and
provisional. Provisional responses are numbered from 100 to 199.
In SIP, these responses are not sent reliably. This choice was
made in RFC 2543 since the messages were meant to just be truly
informational and rendered to the user. However, subsequent work
on PSTN interworking demonstrated a need to map provisional
responses to PSTN messages that needed to be sent reliably.
[RFC3262] was developed to allow reliability of provisional
responses. The specification defines the PRACK method, used for
indicating that a provisional response was received. Though it
provides a generic capability for SIP, RFC 3262 implementations
have been most common in PSTN interworking devices. However,
PRACK brings a great deal of complication for relatively small
benefit. As such, it has seen only moderate levels of deployment.
RFC 3323, A Privacy Mechanism for the Session Initiation Protocol
(SIP) (S): [RFC3323] defines the Privacy header field, used by
clients to request anonymity for their requests. Though it
defines several privacy services, the only one broadly used is the
one that supports privacy of the P-Asserted-Identity header field
[RFC3325].
UA-PRIVACY, UA-Driven Privacy Mechanism for SIP (S): [UA-PRIVACY]
defines a mechanism for achieving anonymous calls in SIP. It is
an alternative to [RFC3323], and instead places more intelligence
in the endpoint to craft anonymous messages by directly accessing
network services.
RFC 2976, The INFO Method (S): [RFC2976] was defined as an extension
to RFC 2543. It defines a method, INFO, used to transport mid-
dialog information that has no impact on SIP itself. Its driving
application was the transport of PSTN-related information when
using SIP between a pair of gateways. Though originally conceived
for broader use, it only found standardized usage with SIP-T
[RFC3372]. It has been used to support numerous proprietary and
non-interoperable extensions due to its poorly defined scope.
RFC 3326, The Reason Header Field for SIP (S): [RFC3326] defines the
Reason header field. It is used in requests, such as BYE, to
indicate the reason that the request is being sent.
RFC 3388, Grouping of Media Lines in the Session Description Protocol
(S): RFC 3388 [RFC3388] defines a framework for grouping together
media streams in an SDP message. Such a grouping allows
relationships between these streams, such as which stream is the
audio for a particular video feed, to be expressed.
RFC 3420, Internet Media Type message/sipfrag (S): [RFC3420] defines
a MIME object that contains a SIP message fragment. Only certain
header fields and parts of the SIP message are present. For
example, it is used to report back on the responses received to a
request sent as a consequence of a REFER.
RFC 3608, SIP Extension Header Field for Service Route Discovery
During Registration (S): [RFC3608] allows a client to determine,
from a REGISTER response, a path of proxies to use in requests it
sends outside of a dialog. It can also be used by proxies to
verify the Route header in client-initiated requests. In many
respects, it is the inverse of the Path header field, but has seen
less usage since default outbound proxies have been sufficient in
many deployments.
RFC 3841, Caller Preferences for SIP (S): [RFC3841] defines a set of
headers that a client can include in a request to control the way
in which the request is routed downstream. It allows a client to
direct a request towards a UA with specific capabilities, which a
UA indicates using [RFC3840].
RFC 4028, Session Timers in SIP (S): [RFC4028] defines a keepalive
mechanism for SIP signaling. It is primarily meant to provide a
way to clean up old state in proxies that are holding call state
for calls from failed endpoints that were never terminated
normally. Despite its name, the session timer is not a mechanism
for detecting a network failure mid-call. Session timers
introduce a fair bit of complexity for relatively little gain, and
have seen moderate deployment.
RFC 4168, SCTP as a Transport for SIP (S): [RFC4168] defines how to
carry SIP messages over the Stream Control Transmission Protocol
(SCTP) [RFC4960]. SCTP has seen very limited usage for SIP
transport.
RFC 4244, An Extension to SIP for Request History Information (S):
[RFC4244] defines the History-Info header field, which indicates
information on how and why a call came to be routed to a
particular destination.
RFC 4145, TCP-Based Media Transport in the Session Description
Protocol (SDP) (S): [RFC4145] defines an extension to SDP for
setting up TCP-based sessions between user agents. It defines who
sets up the connection and how its lifecycle is managed. It has
seen relatively little usage due to the small number of media
types to date that use TCP.
RFC 4091, The Alternative Network Address Types (ANAT) Semantics for
the Session Description Protocol (SDP) Grouping Framework (S):
[RFC4091] defines a mechanism for including both IPv4 and IPv6
addresses for a media session as alternates. This mechanism has
been deprecated in favor of ICE [ICE].
SDP-MEDIA, SDP Media Capabilities Negotiation (S): [SDP-MEDIA]
defines an extension to the SDP capability negotiation framework
[SDP-CAP] for negotiating codecs, codec parameters, and media
streams.
BODY-HANDLING, Message Body Handling in the Session Initiation
Protocol (SIP): [BODY-HANDLING] clarifies handling of bodies in SIP,
focusing primarily on multi-part behavior, which was under-
specified in SIP.
6. NAT Traversal
These SIP extensions are primarily aimed at addressing NAT traversal
for SIP.
ICE, Interactive Connectivity Establishment (ICE) (S): [ICE] defines
a technique for NAT traversal of media sessions for protocols that
make use of the offer/answer model. This specification is the
IETF-recommended mechanism for NAT traversal for SIP media
streams, and is meant to be used even by endpoints that are
themselves never behind a NAT. A SIP option tag and media feature
tag [OPTION-TAG] have been defined for use with ICE.
ICE-TCP, TCP Candidates with Interactive Connectivity Establishment
(ICE) (S): [ICE-TCP] specifies the usage of ICE for TCP streams.
This allows for selection of RTP-based voice on top of TCP only
when NAT or firewalls would prevent UDP-based voice from working.
RFC 3605, Real Time Control Protocol (RTCP) Attribute in the Session
Description Protocol (SDP) (S): [RFC3605] defines a way to
explicitly signal, within an SDP message, the IP address and port
for RTCP, rather than using the port+1 rule in the Real Time
Transport Protocol (RTP) [RFC3550]. It is needed for devices
behind NAT, and the specification is required by ICE.
OUTBOUND, Managing Client Initiated Connections through SIP (S):
[OUTBOUND], also known as SIP outbound, defines important changes
to the SIP registration mechanism that enable delivery of SIP
messages towards a UA when it is behind a NAT.
RFC 3581, An Extension to SIP for Symmetric Response Routing (S):
[RFC3581] defines the rport parameter of the Via header. It
allows SIP responses to traverse NAT.
GRUU, Obtaining and Using Globally Routable User Agent Identifiers
(GRUU) in SIP (S): [GRUU] defines a mechanism for directing requests
towards a specific UA instance. GRUU is essential for features
like transfer and provides another piece of the SIP NAT traversal
story.
7. Call Control Primitives
Numerous SIP extensions provide a toolkit of dialog- and call-
management techniques. These techniques have been combined together
to build many SIP-based services.
RFC 3515, The REFER Method (S): REFER [RFC3515] defines a mechanism
for asking a user agent to send a SIP request. It's a form of SIP
remote control, and is the primary tool used for call transfer in
SIP. Beware that not all potential uses of REFER (neither for all
methods nor for all URI schemes) are well defined. Implementors
should only use the well-defined ones, and should not second guess
or freely assume behavior for the others to avoid unexpected
behavior of remote UAs, interoperability issues, and other bad
surprises.
RFC 3725, Best Current Practices for Third Party Call Control (3pcc)
(B): [RFC3725] defines a number of different call flows that allow
one SIP entity, called the controller, to create SIP sessions
amongst other SIP user agents.
RFC 3911, The SIP Join Header Field (S): [RFC3911] defines the Join
header field. When sent in an INVITE, it causes the recipient to
join the resulting dialog into a conference with another dialog in
progress.
RFC 3891, The SIP Replaces Header (S): [RFC3891] defines a mechanism
that allows a new dialog to replace an existing dialog. It is
useful for certain advanced transfer services.
RFC 3892, The SIP Referred-By Mechanism (S): [RFC3892] defines the
Referred-By header field. It is used in requests triggered by
REFER, and provides the identity of the referring party to the
referred-to party.
RFC 4117, Transcoding Services Invocation in SIP Using Third Party
Call Control (I): [RFC4117] defines how to use 3pcc for the purposes
of invoking transcoding services for a call.
8. Event Framework
RFC 3265, SIP-Specific Event Notification (S): [RFC3265] defines the
SUBSCRIBE and NOTIFY methods. These two methods provide a general
event notification framework for SIP. To actually use the
framework, extensions need to be defined for specific event
packages. An event package defines a schema for the event data
and describes other aspects of event processing specific to that
schema. An RFC 3265 implementation is required when any event
package is used.
RFC 3903, SIP Extension for Event State Publication (S): [RFC3903]
defines the PUBLISH method. It is not an event package, but is
used by all event packages as a mechanism for pushing an event
into the system.
RFC 4662, A Session Initiation Protocol (SIP) Event Notification
Extension for Resource Lists (S): [RFC4662] defines an extension to
RFC 3265 that allows a client to subscribe to a list of resources
using a single subscription. The server, called a Resource List
Server (RLS), will "expand" the subscription and subscribe to each
individual member of the list. It has found applicability
primarily in the area of presence, but can be used with any event
package.
SUBNOT-ETAGS, An Extension to Session Initiation Protocol (SIP)
Events for Conditional Event Notification (S): [SUBNOT-ETAGS]
defines an extension to RFC 3265 to optimize the performance of
notifications. When a client subscribes, it can indicate what
version of a document it has so that the server can skip sending a
notification if the client is up-to-date. It is applicable to any
event package.
9. Event Packages
These are event packages defined to utilize the SIP events framework.
Many of these are also listed elsewhere in their respective areas.
RFC 3680, A SIP Event Package for Registrations (S): [RFC3680]
defines an event package for finding out about changes in
registration state.
GRUU-REG (S): [GRUU-REG] is an extension to the registration event
package [RFC3680] that allows user agents to learn about their
GRUUs. It is particularly useful in helping to synchronize a
client and its registrar with their currently valid temporary
GRUU.
RFC 3842, A Message Summary and Message Waiting Indication Event
Package for SIP (S): [RFC3842] defines a way for a user agent to
find out about voicemails and other messages that are waiting for
it. Its primary purpose is to enable the voicemail waiting lamp
on most business telephones.
RFC 3856, A Presence Event Package for SIP (S): [RFC3856] defines an
event package for indicating user presence through SIP.
RFC 3857, A Watcher Information Event Template Package for SIP (S):
[RFC3857], also known as winfo, provides a mechanism for a user
agent to find out what subscriptions are in place for a particular
event package. Its primary usage is with presence, but it can be
used with any event package.
RFC 4235, An INVITE-Initiated Dialog Event Package for SIP (S):
[RFC4235] defines an event package for learning the state of the
dialogs in progress at a user agent, and is one of several RFCs
starting with the important number 42 [HGTTG].
RFC 4575, A SIP Event Package for Conference State (S): [RFC4575]
defines a mechanism for learning about changes in conference
state, including conference membership.
RFC 4730, A SIP Event Package for Key Press Stimulus (KPML) (S):
[RFC4730] defines a way for an application in the network to
subscribe to the set of key presses made on the keypad of a
traditional telephone. It, along with RFC 4733 [RFC4733], are the
two mechanisms defined for handling DTMF. RFC 4730 is a
signaling-path solution, and RFC 4733 is a media-path solution.
RTCP-SUM, SIP Event Package for Voice Quality Reporting (S):
[RTCP-SUM] defines a SIP event package that enables the collection
and reporting of metrics that measure the quality for Voice over
Internet Protocol (VoIP) sessions.
SESSION-POLICY, A Framework for Session Initiation Protocol (SIP)
Session Policies (S): [SESSION-POLICY] defines a framework for
session policies. In this framework, policy servers are used to
tell user agents about the media characteristics required for a
particular session. The session policy framework has not been
widely implemented.
POLICY-PACK, A Session Initiation Protocol (SIP) Event Package for
Session-Specific Session Policies (S): [POLICY-PACK] defines a SIP
event package used in conjunction with the session policy
framework [SESSION-POLICY].
RFC 5362, The Session Initiation Protocol (SIP) Pending Additions
Event Package (S): [RFC5362] defines a SIP event package that allows
a UA to learn whether consent has been given for the addition of
an address to a SIP "mailing list". It is used in conjunction
with the SIP framework for consent [RFC5360].
10. Quality of Service
Several specifications concern themselves with the interactions of
SIP with network Quality of Service (QoS) mechanisms.
RFC 3312, Integration of Resource Management and SIP (S): [RFC3312],
updated by [RFC4032], defines a way to make sure that the phone of
the called party doesn't ring until a QoS reservation has been
installed in the network. It does so by defining a general
preconditions framework, which defines conditions that must be
true in order for a SIP session to proceed.
QoS-ID, Quality of Service (QoS) Mechanism Selection in the Session
Description Protocol (SDP) (S): [QoS-ID] defines a way for user
agents to negotiate what type of end-to-end QoS mechanism to use
for a session. At this time, there are two that can be used: the
Resource Reservation Protocol (RSVP) and Next Steps in Signaling
(NSIS). This negotiation is done through an SDP extension. Due
to limited deployment of RSVP and even more limited deployment of
NSIS, this extension has not been widely used.
RFC 3313, Private SIP Extensions for Media Authorization (I):
[RFC3313] defines a P-header that provides a mechanism for passing
an authorization token between SIP and a network QoS reservation
protocol like RSVP. Its purpose is to make sure network QoS is
only granted if a client has made a SIP call through the same
provider's network. This specification is sometimes referred to
as the SIP walled-garden specification by the truly paranoid
androids in the SIP community. This is because it requires
coupling of signaling and the underlying IP network.
RFC 3524, Mapping of Media Streams to Resource Reservation Flows
(S): [RFC3524] defines a usage of the SDP grouping framework for
indicating that a set of media streams should be handled by a
single resource reservation.
11. Operations and Management
Several specifications have been defined to support operations and
management of SIP systems. These include mechanisms for
configuration and network diagnostics.
CONFIG-FRAME, A Framework for SIP User Agent Profile Delivery (S):
[CONFIG-FRAME] defines a mechanism that allows a SIP user agent to
bootstrap its configuration from the network and receive updates
to its configuration, should it change. This is considered an
essential piece of deploying a usable SIP network.
RTCP-SUM, SIP Event Package for Voice Quality Reporting (S):
[RTCP-SUM] defines a SIP event package that enables the collection
and reporting of metrics that measure the quality for Voice over
Internet Protocol (VoIP) sessions.
12. SIP Compression
Sigcomp [RFC3320] [RFC4896] was defined to allow compression of SIP
messages over low bandwidth links. Sigcomp is not formally part of
SIP. However, usage of Sigcomp with SIP has required extensions to
SIP.
RFC 3486, Compressing SIP (S): [RFC3486] defines a SIP URI parameter
that can be used to indicate that a SIP server supports Sigcomp.
RFC 5049, Applying Signaling Compression (SigComp) to the Session
Initiation Protocol (SIP) (S): [RFC5049] defines how to apply
Sigcomp to SIP.
13. SIP Service URIs
Several extensions define well-known services that can be invoked by
constructing requests with specific structures for the Request URI,
resulting in specific behaviors at the User Agent Server (UAS).
RFC 3087, Control of Service Context using Request URI (I):
[RFC3087] introduced the context of using Request URIs, encoded
appropriately, to invoke services.
RFC 4662, A SIP Event Notification Extension for Resource Lists (S):
[RFC4662] defines a resource called a Resource List Server (RLS).
A client can send a subscribe to this server. The server will
generate a series of subscriptions, compile the resulting
information, and send it back to the subscriber. The set of
resources that the RLS will subscribe to is a property of the
request URI in the SUBSCRIBE request.
RFC 5363, Framework and Security Considerations for Session
Initiation Protocol (SIP) Uniform Resource Identifier (URI)-List
Services (S): [RFC5363] defines the framework for list services in
SIP. In this framework, a UA can include an XML list object in
the body of various requests and the server will provide list-
oriented services as a consequence. For example, a SUBSCRIBE with
a list subscribes to the URI in the list.
RFC 5367, Subscriptions To Request-Contained Resource Lists in SIP
(S): [RFC5367] uses the URI-list framework [RFC5363] and allows a
client to subscribe to a resource called a Resource List Server.
This server will generate subscriptions to the URI in the list,
compile the resulting information, and send it back to the
subscriber.
RFC 5365, Multiple-Recipient MESSAGE Requests in SIP (S): [RFC5365]
uses the URI-list framework [RFC5363] and allows a client to send
a MESSAGE to a number of recipients.
RFC 5366, Conference Establishment Using Request-Contained Lists in
SIP (S): [RFC5366] uses the URI-list framework [RFC5363]. It allows
a client to ask the server to act as a conference focus and send
an invitation to each recipient in the list.
RFC 4240, Basic Network Media Services with SIP (I): [RFC4240]
defines a way for SIP application servers to invoke announcement
and conferencing services from a media server. This is
accomplished through a set of defined URI parameters that tell the
media server what to do, such as what file to play and what
language to render it in.
RFC 4458, Session Initiation Protocol (SIP) URIs for Applications
such as Voicemail and Interactive Voice Response (IVR) (I):
[RFC4458] defines a way to invoke voicemail and IVR services by
using a SIP URI constructed in a particular way.
14. Minor Extensions
These SIP extensions don't fit easily into a single specific use
case. They have somewhat general applicability, but they solve a
relatively small problem or provide an optimization.
RFC 4488, Suppression of the SIP REFER Implicit Subscription (S):
[RFC4488] defines an enhancement to REFER. REFER normally creates
an implicit subscription to the target of the REFER. This
subscription is used to pass back updates on the progress of the
referral. This extension allows that implicit subscription to be
bypassed as an optimization.
RFC 4538, Request Authorization through Dialog Identification in SIP
(S): [RFC4538] provides a mechanism that allows a UAS to authorize a
request because the requestor proves it knows a dialog that is in
progress with the UAS. The specification is useful in conjunction
with the SIP application interaction framework [INTERACT-FRAME].
RFC 4508, Conveying Feature Tags with the REFER Method in SIP (S):
[RFC4508] defines a mechanism for carrying RFC 3840 feature tags
in REFER. It is useful for informing the target of the REFER
about the characteristics of the intended target of the referred
request.
RFC 5373, Requesting Answer Modes for SIP (S): [RFC5373] defines an
extension for indicating to the called party whether or not the
phone should ring and/or be answered immediately. This is useful
for push-to-talk and for diagnostic applications.
RFC 5079, Rejecting Anonymous Requests in SIP (S): [RFC5079] defines
a mechanism for a called party to indicate to the calling party
that a call was rejected since the caller was anonymous. This is
needed for implementation of the Anonymous Call Rejection (ACR)
feature in SIP.
RFC 5368, Referring to Multiple Resources in SIP (S): [RFC5368]
allows a UA sending a REFER to ask the recipient of the REFER to
generate multiple SIP requests, not just one. This is useful for
conferencing, where a client would like to ask a conference server
to eject multiple users.
RFC 4483, A Mechanism for Content Indirection in Session Initiation
Protocol (SIP) Messages (S): [RFC4483] defines a mechanism for
content indirection. Instead of carrying an object within a SIP
body, a URL reference is carried instead, and the recipient
dereferences the URL to obtain the object. The specification has
potential applicability for sending large instant messages, but
has yet to find much actual use.
RFC 3890, A Transport Independent Bandwidth Modifier for the Session
Description Protocol (SDP) (S): [RFC3890] specifies an SDP extension
that allows for the description of the bandwidth for a media
session that is independent of the underlying transport mechanism.
RFC 4583, Session Description Protocol (SDP) Format for Binary Floor
Control Protocol (BFCP) Streams (S): [RFC4583] defines a mechanism
in SDP to signal floor control streams that use BFCP. It is used
for push-to-talk and conference floor control.
CONNECT-PRECON, Connectivity Preconditions for Session Description
Protocol Media Streams (S): [CONNECT-PRECON] defines a usage of the
precondition framework [RFC3312]. The connectivity precondition
makes sure that the session doesn't get established until actual
packet connectivity is checked.
RFC 4796, The SDP (Session Description Protocol) Content Attribute
(S): [RFC4796] defines an SDP attribute for describing the purpose
of a media stream. Examples include a slide view, the speaker, a
sign language feed, and so on.
IPv6-TRANS, IPv6 Transition in the Session Initiation Protocol (SIP)
(S): [IPv6-TRANS] defines practices for interworking between IPv6
and IPv6 user agents. This is done through multi-homed proxies
that interwork IPv4 and IPv6, along with ICE [ICE] for media
traversal. The specification includes some minor extensions and
clarifications to SDP in order to cover some additional cases.
CONNECT-REUSE, Connection Reuse in the Session Initiation Protocol
(SIP) (S): [CONNECT-REUSE] defines an extension to SIP that allows a
Transport Layer Security (TLS) connection between servers to be
reused for requests in both directions. Normally, two connections
are set up between a pair of servers, one for requests in each
direction.
15. Security Mechanisms
Several extensions provide additional security features to SIP.
RFC 4474, Enhancements for Authenticated Identity Management in SIP
(S): [RFC4474] defines a mechanism for providing a cryptographically
verifiable identity of the calling party in a SIP request. Known
as "SIP Identity", this mechanism provides an alternative to RFC
3325. It has seen little deployment so far, but its importance as
a key construct for anti-spam techniques and new security
mechanisms makes it a core part of the SIP specifications.
RFC 4916, Connected Identity in the Session Initiation Protocol (SIP)
(S): [RFC4916] formally updates RFC 3261. It defines an extension
to SIP that allows a calling user to determine the identity of the
final called user (connected party). Due to forwarding and
retargeting services, this may not be the same as the user that
the caller was originally trying to reach. The mechanism works in
tandem with the SIP identity specification [RFC4474] to provide
signatures over the connected party identity. It can also be used
if a party identity changes mid call due to third party call
control actions or PSTN behavior.
SIPS-URI, The Use of the SIPS URI Scheme in the Session Initiation
Protocol (SIP) (S): [SIPS-URI] is intended to update RFC 3261. It
revises the processing of the SIPS URI, originally defined in RFC
3261, to fix many errors and problems that have been encountered
with that mechanism.
DOMAIN-CERTS, Domain Certificates in the Session Initiation Protocol
(SIP) (B): [DOMAIN-CERTS] clarifies the usage of SIP over TLS with
regards to certificate handling, and defines additional procedures
needed for interoperability.
RFC 3323, A Privacy Mechanism for the Session Initiation Protocol
(SIP) (S): [RFC3323] defines the Privacy header field, used by
clients to request anonymity for their requests. Though it
defines several privacy services, the only one broadly used is the
one that supports privacy of the P-Asserted-Identity header field
[RFC3325].
RFC 4567, Key Management Extensions for Session Description Protocol
(SDP) and Real Time Streaming Protocol (RTSP) (S): [RFC4567] defines
extensions to SDP that allow tunneling of a key management
protocol, namely MIKEY [RFC3830], through offer/answer exchanges.
This mechanism is one of three Secure Realtime Transport Protocol
(SRTP) keying techniques specified for SIP, with Datagram
Transport Layer Security (DTLS)-SRTP [SRTP-FRAME] having been
selected as the final solution.
RFC 4568, Session Description Protocol (SDP) Security Descriptions
for Media Streams (S): [RFC4568] defines extensions to SDP that
allow for the negotiation of keying material directly through
offer/answer, without a separate key management protocol. This
mechanism, sometimes called sdescriptions, has the drawback that
the media keys are available to any entity that has visibility to
the SDP. It is one of three SRTP keying techniques specified for
SIP, with DTLS-SRTP [SRTP-FRAME] having been selected as the final
solution.
SRTP-FRAME, Framework for Establishing an SRTP Security Context using
DTLS (S): [SRTP-FRAME] defines the overall framework and SDP and SIP
processing required to perform key management for RTP using
Datagram TLS (DTLS) [RFC4347] directly between endpoints, over the
media path. It is one of three SRTP keying techniques specified
for SIP, with DTLS-SRTP [SRTP-FRAME] having been selected as the
final solution.
RFC 3853, S/MIME Advanced Encryption Standard (AES) Requirement for
SIP (S): [RFC3853] formally updates RFC 3261. It is a brief
specification that updates the cryptography mechanisms used in SIP
S/MIME. However, SIP S/MIME has seen very little deployment.
CERTS, Certificate Management Service for the Session Initiation
Protocol (SIP) (S): [CERTS] defines a certificate service for SIP
whose purpose is to facilitate the deployment of S/MIME. The
certificate service allows clients to store and retrieve their own
certificates, in addition to obtaining the certificates for other
users.
RFC 3893, Session Initiation Protocol (SIP) Authenticated Identity
Body (AIB) Format (S): [RFC3893] defines a SIP message fragment that
can be signed in order to provide an authenticated identity over a
request. It was an early predecessor to [RFC4474], and
consequently AIB has seen no deployment.
SAML, SIP SAML Profile and Binding (S): [SAML] defines the usage of
the Security Assertion Markup Language (SAML) within SIP, and
describes how to use it in conjunction with SIP identity [RFC4474]
to provide authenticated assertions about a user's role or
attributes.
RFC 5360, A Framework for Consent-Based Communications in the Session
Initiation Protocol (SIP) (S): [RFC5360] defines several extensions
to SIP, including the Trigger-Consent and Permission-Missing
header fields. These header fields, in addition to the other
procedures defined in the document, define a way to manage
membership on "SIP mailing lists" used for instant messaging or
conferencing. In particular, it helps avoid the problem of using
such amplification services for the purposes of an attack on the
network by making sure a user authorizes the addition of their
address onto such a service.
RFC 5361, A Document Format for Requesting Consent (S): [RFC5361]
defines an XML object used by the consent framework. Consent
documents are sent from SIP "mailing list servers" to users to
allow them to manage their membership on lists.
RFC 5362, The Session Initiation Protocol (SIP) Pending Additions
Event Package (S): [RFC5362] defines a SIP event package that allows
a UA to learn whether consent has been given for the addition of
an address to a SIP "mailing list". It is used in conjunction
with the SIP framework for consent [RFC5360].
RFC 3329, Security Mechanism Agreement for SIP (S): [RFC3329]
defines a mechanism to prevent bid-down attacks in conjunction
with SIP authentication. The mechanism has seen very limited
deployment. It was defined as part of the 3GPP IP Multimedia
Subsystem (IMS) specification suite [3GPP.24.229], and is needed
only when there is a multiplicity of security mechanisms deployed
at a particular server. In practice, this has not been the case.
RFC 4572, Connection-Oriented Media Transport over the Transport
Layer Security (TLS) Protocol in the Session Description Protocol
(SDP) (S): [RFC4572] specifies a mechanism for signaling TLS-based
media streams between endpoints. It expands the TCP-based media
signaling parameters defined in [RFC4145] to include fingerprint
information for TLS streams so that TLS can operate between end
hosts using self-signed certificates.
RFC 5027, Security Preconditions for Session Description Protocol
Media Streams (S): [RFC5027] defines a precondition for use with the
preconditions framework [RFC3312]. The security precondition
prevents a session from being established until a security media
stream is set up.
RFC 3310, Hypertext Transfer Protocol (HTTP) Digest Authentication
Using Authentication and Key Agreement (S): [RFC3310] defines an
extension to digest authentication to allow it to work with the
credentials stored in cell phones. Though technically it is an
extension to HTTP digest, its primary application is SIP. This
extension is useful primarily to implementors of IMS.
RFC 4169, Hypertext Transfer Protocol (HTTP) Digest Authentication
Using Authentication and Key Agreement (AKA) Version-2 (S):
[RFC4169] is an enhancement to [RFC3310] that further improves
security of the authentication.
16. Conferencing
Numerous SIP and SDP extensions are aimed at conferencing as their
primary application.
RFC 4574, The SDP (Session Description Protocol) Label Attribute
(S): [RFC4574] defines an SDP attribute for providing an opaque
label for media streams. These labels can be referred to by
external documents, and in particular, by conference policy
documents. This allows a UA to tie together documents it may
obtain through conferencing mechanisms to media streams to which
they refer.
RFC 3911, The SIP Join Header Field (S): [RFC3911] defines the Join
header field. When sent in an INVITE, it causes the recipient to
join the resulting dialog into a conference with another dialog in
progress.
RFC 4575, A SIP Event Package for Conference State (S): [RFC4575]
defines a mechanism for learning about changes in conference
state, including conference membership.
RFC 5368, Referring to Multiple Resources in SIP (S): [RFC5368]
allows a UA sending a REFER to ask the recipient of the REFER to
generate multiple SIP requests, not just one. This is useful for
conferencing, where a client would like to ask a conference server
to eject multiple users.
RFC 5366, Conference Establishment Using Request-Contained Lists in
SIP (S): [RFC5366] is similar to [RFC5367]. However, instead of
subscribing to the resource, an INVITE request is sent to the
resource, and it will act as a conference focus and generate an
invitation to each recipient in the list.
RFC4579, Session Initiation Protocol (SIP) Call Control -
Conferencing for User Agents (B): [RFC4579] defines best practice
procedures and call flows for conferencing. This includes
conference creation, joining, and dial out, amongst other
capabilities.
RFC 4583, Session Description Protocol (SDP) Format for Binary Floor
Control Protocol (BFCP) Streams (S): [RFC4583] defines a mechanism
in SDP to signal floor control streams that use BFCP. It is used
for push-to-talk and conference floor control.
17. Instant Messaging, Presence, and Multimedia
SIP provides extensions for instant messaging, presence, and
multimedia.
RFC 3428, SIP Extension for Instant Messaging (S): [RFC3428] defines
the MESSAGE method, used for sending an instant message without
setting up a session (sometimes called "page mode").
RFC 3856, A Presence Event Package for SIP (S): [RFC3856] defines an
event package for indicating user presence through SIP.
RFC 3857, A Watcher Information Event Template Package for SIP (S):
[RFC3857], also known as winfo, provides a mechanism for a user
agent to find out what subscriptions are in place for a particular
event package. Its primary usage is with presence, but it can be
used with any event package.
TRANSFER-MECH, A Session Description Protocol (SDP) Offer/Answer
Mechanism to Enable File Transfer (S): [TRANSFER-MECH] defines a
mechanism for signaling a file transfer session with SIP.
18. Emergency Services
Emergency services include preemption features, which allow
authorized individuals to gain access to network resources in time of
emergency, along with traditional emergency calling.
RFC 4411, Extending the SIP Reason Header for Preemption Events (S):
[RFC4411] defines an extension to the Reason header, allowing a UA
to know that its dialog was torn down because a higher priority
session came through.
RFC 4412, Communications Resource Priority for SIP (S): [RFC4412]
defines a new header field, Resource-Priority, that allows a
session to get priority treatment from the network.
LOCATION, Location Conveyance for the Session Initiation Protocol
(S): [LOCATION] defines a mechanism for carrying location objects in
SIP messages. This is used to convey location from a UA to an
emergency call taker.
19. Security Considerations
This specification is an overview of existing specifications and does
not introduce any security considerations on its own. Of course, the
world would be far more secure if everyone would follow one simple
rule: "Don't Panic!" [HGTTG].
20. Acknowledgements
The author would like to thank Spencer Dawkins, Brian Stucker, Keith
Drage, John Elwell, and Avshalom Houri for their comments on this
document.
21. Informative References
[3GPP.24.229] 3GPP, "Internet Protocol (IP) multimedia call
control protocol based on Session Initiation
Protocol (SIP) and Session Description Protocol
(SDP); Stage 3", 3GPP TS 24.229 5.22.0,
September 2008.
[ABNF-FIX] Gurbani, V. and B. Carpenter, "Essential correction
for IPv6 ABNF in RFC3261", Work in Progress,
November 2007.
[BODY-HANDLING] Camarillo, G., "Message Body Handling in the
Session Initiation Protocol (SIP)", Work
in Progress, November 2008.
[CERTS] Jennings, C. and J. Fischl, "Certificate Management
Service for The Session Initiation Protocol (SIP)",
Work in Progress, November 2008.
[CONFIG-FRAME] Channabasappa, S., "A Framework for Session
Initiation Protocol User Agent Profile Delivery",
Work in Progress, February 2008.
[CONNECT-PRECON] Andreasen, F., Camarillo, G., Oran, D., and D.
Wing, "Connectivity Preconditions for Session
Description Protocol Media Streams", Work
in Progress, October 2008.
[CONNECT-REUSE] Gurbani, V., Mahy, R., and B. Tate, "Connection
Reuse in the Session Initiation Protocol (SIP)",
Work in Progress, October 2008.
[DOMAIN-CERTS] Gurbani, V., Lawrence, S., and B. Laboratories,
"Domain Certificates in the Session Initiation
Protocol (SIP)", Work in Progress, October 2008.
[ECRIT-FRAME] Rosen, B., Schulzrinne, H., Polk, J., and A.
Newton, "Framework for Emergency Calling using
Internet Multimedia", Work in Progress, July 2008.
[GRUU] Rosenberg, J., "Obtaining and Using Globally
Routable User Agent (UA) URIs (GRUU) in the Session
Initiation Protocol (SIP)", Work in Progress,
October 2007.
[GRUU-REG] Kyzivat, P., "Registration Event Package Extension
for Session Initiation Protocol (SIP) Globally
Routable User Agent URIs (GRUUs)", Work
in Progress, July 2007.
[HGTTG] Adams, D., "The Hitchhiker's Guide to the Galaxy",
September 1979.
[ICE] Rosenberg, J., "Interactive Connectivity
Establishment (ICE): A Protocol for Network Address
Translator (NAT) Traversal for Offer/Answer
Protocols", Work in Progress, October 2007.
[ICE-TCP] Rosenberg, J., "TCP Candidates with Interactive
Connectivity Establishment (ICE)", Work
in Progress, July 2008.
[INTERACT-FRAME] Rosenberg, J., "A Framework for Application
Interaction in the Session Initiation Protocol
(SIP)", Work in Progress, July 2005.
[IPv6-TRANS] Camarillo, G., "IPv6 Transition in the Session
Initiation Protocol (SIP)", Work in Progress,
August 2007.
[LOCATION] Polk, J. and B. Rosen, "Location Conveyance for the
Session Initiation Protocol", Work in Progress,
November 2008.
[LOOP-FIX] Sparks, R., Lawrence, S., Hawrylyshen, A., and B.
Campen, "Addressing an Amplification Vulnerability
in Session Initiation Protocol (SIP) Forking
Proxies", Work in Progress, October 2008.
[OPTION-TAG] Rosenberg, J., "Indicating Support for Interactive
Connectivity Establishment (ICE) in the Session
Initiation Protocol (SIP)", Work in Progress,
June 2007.
[OUTBOUND] Jennings, C. and R. Mahy, "Managing Client
Initiated Connections in the Session Initiation
Protocol (SIP)", Work in Progress, October 2008.
[POLICY-PACK] Hilt, V. and G. Camarillo, "A Session Initiation
Protocol (SIP) Event Package for Session-Specific
Session Policies.", Work in Progress, July 2008.
[QoS-ID] Polk, J., Dhesikan, S., and G. Camarillo, "Quality
of Service (QoS) Mechanism Selection in the Session
Description Protocol (SDP)", Work in Progress,
November 2008.
[RECORD-ROUTE] Froment, T., Lebel, C., and B. Bonnaerens,
"Addressing Record-Route issues in the Session
Initiation Protocol (SIP)", Work in Progress,
October 2008.
[RFC2026] Bradner, S., "The Internet Standards Process --
Revision 3", BCP 9, RFC 2026, October 1996.
[RFC2543] Handley, M., Schulzrinne, H., Schooler, E., and J.
Rosenberg, "SIP: Session Initiation Protocol",
RFC 2543, March 1999.
[RFC2782] Gulbrandsen, A., Vixie, P., and L. Esibov, "A DNS
RR for specifying the location of services (DNS
SRV)", RFC 2782, February 2000.
[RFC2848] Petrack, S. and L. Conroy, "The PINT Service
Protocol: Extensions to SIP and SDP for IP Access
to Telephone Call Services", RFC 2848, June 2000.
[RFC2976] Donovan, S., "The SIP INFO Method", RFC 2976,
October 2000.
[RFC3087] Campbell, B. and R. Sparks, "Control of Service
Context using SIP Request-URI", RFC 3087,
April 2001.
[RFC3204] Zimmerer, E., Peterson, J., Vemuri, A., Ong, L.,
Audet, F., Watson, M., and M. Zonoun, "MIME media
types for ISUP and QSIG Objects", RFC 3204,
December 2001.
[RFC3261] 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.
[RFC3262] Rosenberg, J. and H. Schulzrinne, "Reliability of
Provisional Responses in Session Initiation
Protocol (SIP)", RFC 3262, June 2002.
[RFC3263] Rosenberg, J. and H. Schulzrinne, "Session
Initiation Protocol (SIP): Locating SIP Servers",
RFC 3263, June 2002.
[RFC3264] Rosenberg, J. and H. Schulzrinne, "An Offer/Answer
Model with Session Description Protocol (SDP)",
RFC 3264, June 2002.
[RFC3265] Roach, A., "Session Initiation Protocol (SIP)-
Specific Event Notification", RFC 3265, June 2002.
[RFC3310] Niemi, A., Arkko, J., and V. Torvinen, "Hypertext
Transfer Protocol (HTTP) Digest Authentication
Using Authentication and Key Agreement (AKA)",
RFC 3310, September 2002.
[RFC3311] Rosenberg, J., "The Session Initiation Protocol
(SIP) UPDATE Method", RFC 3311, October 2002.
[RFC3312] Camarillo, G., Marshall, W., and J. Rosenberg,
"Integration of Resource Management and Session
Initiation Protocol (SIP)", RFC 3312, October 2002.
[RFC3313] Marshall, W., "Private Session Initiation Protocol
(SIP) Extensions for Media Authorization",
RFC 3313, January 2003.
[RFC3320] Price, R., Bormann, C., Christoffersson, J., Hannu,
H., Liu, Z., and J. Rosenberg, "Signaling
Compression (SigComp)", RFC 3320, January 2003.
[RFC3323] Peterson, J., "A Privacy Mechanism for the Session
Initiation Protocol (SIP)", RFC 3323,
November 2002.
[RFC3325] Jennings, C., Peterson, J., and M. Watson, "Private
Extensions to the Session Initiation Protocol (SIP)
for Asserted Identity within Trusted Networks",
RFC 3325, November 2002.
[RFC3326] Schulzrinne, H., Oran, D., and G. Camarillo, "The
Reason Header Field for the Session Initiation
Protocol (SIP)", RFC 3326, December 2002.
[RFC3327] Willis, D. and B. Hoeneisen, "Session Initiation
Protocol (SIP) Extension Header Field for
Registering Non-Adjacent Contacts", RFC 3327,
December 2002.
[RFC3329] Arkko, J., Torvinen, V., Camarillo, G., Niemi, A.,
and T. Haukka, "Security Mechanism Agreement for
the Session Initiation Protocol (SIP)", RFC 3329,
January 2003.
[RFC3372] Vemuri, A. and J. Peterson, "Session Initiation
Protocol for Telephones (SIP-T): Context and
Architectures", BCP 63, RFC 3372, September 2002.
[RFC3388] Camarillo, G., Eriksson, G., Holler, J., and H.
Schulzrinne, "Grouping of Media Lines in the
Session Description Protocol (SDP)", RFC 3388,
December 2002.
[RFC3398] Camarillo, G., Roach, A., Peterson, J., and L. Ong,
"Integrated Services Digital Network (ISDN) User
Part (ISUP) to Session Initiation Protocol (SIP)
Mapping", RFC 3398, December 2002.
[RFC3401] Mealling, M., "Dynamic Delegation Discovery System
(DDDS) Part One: The Comprehensive DDDS", RFC 3401,
October 2002.
[RFC3420] Sparks, R., "Internet Media Type message/sipfrag",
RFC 3420, November 2002.
[RFC3427] Mankin, A., Bradner, S., Mahy, R., Willis, D., Ott,
J., and B. Rosen, "Change Process for the Session
Initiation Protocol (SIP)", BCP 67, RFC 3427,
December 2002.
[RFC3428] Campbell, B., Rosenberg, J., Schulzrinne, H.,
Huitema, C., and D. Gurle, "Session Initiation
Protocol (SIP) Extension for Instant Messaging",
RFC 3428, December 2002.
[RFC3482] Foster, M., McGarry, T., and J. Yu, "Number
Portability in the Global Switched Telephone
Network (GSTN): An Overview", RFC 3482,
February 2003.
[RFC3486] Camarillo, G., "Compressing the Session Initiation
Protocol (SIP)", RFC 3486, February 2003.
[RFC3515] Sparks, R., "The Session Initiation Protocol (SIP)
Refer Method", RFC 3515, April 2003.
[RFC3524] Camarillo, G. and A. Monrad, "Mapping of Media
Streams to Resource Reservation Flows", RFC 3524,
April 2003.
[RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V.
Jacobson, "RTP: A Transport Protocol for Real-Time
Applications", STD 64, RFC 3550, July 2003.
[RFC3578] Camarillo, G., Roach, A., Peterson, J., and L. Ong,
"Mapping of Integrated Services Digital Network
(ISDN) User Part (ISUP) Overlap Signalling to the
Session Initiation Protocol (SIP)", RFC 3578,
August 2003.
[RFC3581] Rosenberg, J. and H. Schulzrinne, "An Extension to
the Session Initiation Protocol (SIP) for Symmetric
Response Routing", RFC 3581, August 2003.
[RFC3605] Huitema, C., "Real Time Control Protocol (RTCP)
attribute in Session Description Protocol (SDP)",
RFC 3605, October 2003.
[RFC3608] Willis, D. and B. Hoeneisen, "Session Initiation
Protocol (SIP) Extension Header Field for Service
Route Discovery During Registration", RFC 3608,
October 2003.
[RFC3665] Johnston, A., Donovan, S., Sparks, R., Cunningham,
C., and K. Summers, "Session Initiation Protocol
(SIP) Basic Call Flow Examples", BCP 75, RFC 3665,
December 2003.
[RFC3666] Johnston, A., Donovan, S., Sparks, R., Cunningham,
C., and K. Summers, "Session Initiation Protocol
(SIP) Public Switched Telephone Network (PSTN) Call
Flows", BCP 76, RFC 3666, December 2003.
[RFC3680] Rosenberg, J., "A Session Initiation Protocol (SIP)
Event Package for Registrations", RFC 3680,
March 2004.
[RFC3725] Rosenberg, J., Peterson, J., Schulzrinne, H., and
G. Camarillo, "Best Current Practices for Third
Party Call Control (3pcc) in the Session Initiation
Protocol (SIP)", BCP 85, RFC 3725, April 2004.
[RFC3830] Arkko, J., Carrara, E., Lindholm, F., Naslund, M.,
and K. Norrman, "MIKEY: Multimedia Internet
KEYing", RFC 3830, August 2004.
[RFC3840] Rosenberg, J., Schulzrinne, H., and P. Kyzivat,
"Indicating User Agent Capabilities in the Session
Initiation Protocol (SIP)", RFC 3840, August 2004.
[RFC3841] Rosenberg, J., Schulzrinne, H., and P. Kyzivat,
"Caller Preferences for the Session Initiation
Protocol (SIP)", RFC 3841, August 2004.
[RFC3842] Mahy, R., "A Message Summary and Message Waiting
Indication Event Package for the Session Initiation
Protocol (SIP)", RFC 3842, August 2004.
[RFC3853] Peterson, J., "S/MIME Advanced Encryption Standard
(AES) Requirement for the Session Initiation
Protocol (SIP)", RFC 3853, July 2004.
[RFC3856] Rosenberg, J., "A Presence Event Package for the
Session Initiation Protocol (SIP)", RFC 3856,
August 2004.
[RFC3857] Rosenberg, J., "A Watcher Information Event
Template-Package for the Session Initiation
Protocol (SIP)", RFC 3857, August 2004.
[RFC3890] Westerlund, M., "A Transport Independent Bandwidth
Modifier for the Session Description Protocol
(SDP)", RFC 3890, September 2004.
[RFC3891] Mahy, R., Biggs, B., and R. Dean, "The Session
Initiation Protocol (SIP) "Replaces" Header",
RFC 3891, September 2004.
[RFC3892] Sparks, R., "The Session Initiation Protocol (SIP)
Referred-By Mechanism", RFC 3892, September 2004.
[RFC3893] Peterson, J., "Session Initiation Protocol (SIP)
Authenticated Identity Body (AIB) Format",
RFC 3893, September 2004.
[RFC3903] Niemi, A., "Session Initiation Protocol (SIP)
Extension for Event State Publication", RFC 3903,
October 2004.
[RFC3910] Gurbani, V., Brusilovsky, A., Faynberg, I., Gato,
J., Lu, H., and M. Unmehopa, "The SPIRITS (Services
in PSTN requesting Internet Services) Protocol",
RFC 3910, October 2004.
[RFC3911] Mahy, R. and D. Petrie, "The Session Initiation
Protocol (SIP) "Join" Header", RFC 3911,
October 2004.
[RFC3959] Camarillo, G., "The Early Session Disposition Type
for the Session Initiation Protocol (SIP)",
RFC 3959, December 2004.
[RFC3960] Camarillo, G. and H. Schulzrinne, "Early Media and
Ringing Tone Generation in the Session Initiation
Protocol (SIP)", RFC 3960, December 2004.
[RFC4028] Donovan, S. and J. Rosenberg, "Session Timers in
the Session Initiation Protocol (SIP)", RFC 4028,
April 2005.
[RFC4032] Camarillo, G. and P. Kyzivat, "Update to the
Session Initiation Protocol (SIP) Preconditions
Framework", RFC 4032, March 2005.
[RFC4091] Camarillo, G. and J. Rosenberg, "The Alternative
Network Address Types (ANAT) Semantics for the
Session Description Protocol (SDP) Grouping
Framework", RFC 4091, June 2005.
[RFC4117] Camarillo, G., Burger, E., Schulzrinne, H., and A.
van Wijk, "Transcoding Services Invocation in the
Session Initiation Protocol (SIP) Using Third Party
Call Control (3pcc)", RFC 4117, June 2005.
[RFC4145] Yon, D. and G. Camarillo, "TCP-Based Media
Transport in the Session Description Protocol
(SDP)", RFC 4145, September 2005.
[RFC4168] Rosenberg, J., Schulzrinne, H., and G. Camarillo,
"The Stream Control Transmission Protocol (SCTP) as
a Transport for the Session Initiation Protocol
(SIP)", RFC 4168, October 2005.
[RFC4169] Torvinen, V., Arkko, J., and M. Naslund, "Hypertext
Transfer Protocol (HTTP) Digest Authentication
Using Authentication and Key Agreement (AKA)
Version-2", RFC 4169, November 2005.
[RFC4235] Rosenberg, J., Schulzrinne, H., and R. Mahy, "An
INVITE-Initiated Dialog Event Package for the
Session Initiation Protocol (SIP)", RFC 4235,
November 2005.
[RFC4240] Burger, E., Van Dyke, J., and A. Spitzer, "Basic
Network Media Services with SIP", RFC 4240,
December 2005.
[RFC4244] Barnes, M., "An Extension to the Session Initiation
Protocol (SIP) for Request History Information",
RFC 4244, November 2005.
[RFC4320] Sparks, R., "Actions Addressing Identified Issues
with the Session Initiation Protocol's (SIP) Non-
INVITE Transaction", RFC 4320, January 2006.
[RFC4347] Rescorla, E. and N. Modadugu, "Datagram Transport
Layer Security", RFC 4347, April 2006.
[RFC4411] Polk, J., "Extending the Session Initiation
Protocol (SIP) Reason Header for Preemption
Events", RFC 4411, February 2006.
[RFC4412] Schulzrinne, H. and J. Polk, "Communications
Resource Priority for the Session Initiation
Protocol (SIP)", RFC 4412, February 2006.
[RFC4458] Jennings, C., Audet, F., and J. Elwell, "Session
Initiation Protocol (SIP) URIs for Applications
such as Voicemail and Interactive Voice Response
(IVR)", RFC 4458, April 2006.
[RFC4474] Peterson, J. and C. Jennings, "Enhancements for
Authenticated Identity Management in the Session
Initiation Protocol (SIP)", RFC 4474, August 2006.
[RFC4483] Burger, E., "A Mechanism for Content Indirection in
Session Initiation Protocol (SIP) Messages",
RFC 4483, May 2006.
[RFC4488] Levin, O., "Suppression of Session Initiation
Protocol (SIP) REFER Method Implicit Subscription",
RFC 4488, May 2006.
[RFC4497] Elwell, J., Derks, F., Mourot, P., and O. Rousseau,
"Interworking between the Session Initiation
Protocol (SIP) and QSIG", BCP 117, RFC 4497,
May 2006.
[RFC4508] Levin, O. and A. Johnston, "Conveying Feature Tags
with the Session Initiation Protocol (SIP) REFER
Method", RFC 4508, May 2006.
[RFC4538] Rosenberg, J., "Request Authorization through
Dialog Identification in the Session Initiation
Protocol (SIP)", RFC 4538, June 2006.
[RFC4566] Handley, M., Jacobson, V., and C. Perkins, "SDP:
Session Description Protocol", RFC 4566, July 2006.
[RFC4567] Arkko, J., Lindholm, F., Naslund, M., Norrman, K.,
and E. Carrara, "Key Management Extensions for
Session Description Protocol (SDP) and Real Time
Streaming Protocol (RTSP)", RFC 4567, July 2006.
[RFC4568] Andreasen, F., Baugher, M., and D. Wing, "Session
Description Protocol (SDP) Security Descriptions
for Media Streams", RFC 4568, July 2006.
[RFC4572] Lennox, J., "Connection-Oriented Media Transport
over the Transport Layer Security (TLS) Protocol in
the Session Description Protocol (SDP)", RFC 4572,
July 2006.
[RFC4574] Levin, O. and G. Camarillo, "The Session
Description Protocol (SDP) Label Attribute",
RFC 4574, August 2006.
[RFC4575] Rosenberg, J., Schulzrinne, H., and O. Levin, "A
Session Initiation Protocol (SIP) Event Package for
Conference State", RFC 4575, August 2006.
[RFC4579] Johnston, A. and O. Levin, "Session Initiation
Protocol (SIP) Call Control - Conferencing for User
Agents", BCP 119, RFC 4579, August 2006.
[RFC4583] Camarillo, G., "Session Description Protocol (SDP)
Format for Binary Floor Control Protocol (BFCP)
Streams", RFC 4583, November 2006.
[RFC4662] Roach, A., Campbell, B., and J. Rosenberg, "A
Session Initiation Protocol (SIP) Event
Notification Extension for Resource Lists",
RFC 4662, August 2006.
[RFC4730] Burger, E. and M. Dolly, "A Session Initiation
Protocol (SIP) Event Package for Key Press Stimulus
(KPML)", RFC 4730, November 2006.
[RFC4733] Schulzrinne, H. and T. Taylor, "RTP Payload for
DTMF Digits, Telephony Tones, and Telephony
Signals", RFC 4733, December 2006.
[RFC4796] Hautakorpi, J. and G. Camarillo, "The Session
Description Protocol (SDP) Content Attribute",
RFC 4796, February 2007.
[RFC4896] Surtees, A., West, M., and A. Roach, "Signaling
Compression (SigComp) Corrections and
Clarifications", RFC 4896, June 2007.
[RFC4916] Elwell, J., "Connected Identity in the Session
Initiation Protocol (SIP)", RFC 4916, June 2007.
[RFC4960] Stewart, R., "Stream Control Transmission
Protocol", RFC 4960, September 2007.
[RFC5027] Andreasen, F. and D. Wing, "Security Preconditions
for Session Description Protocol (SDP) Media
Streams", RFC 5027, October 2007.
[RFC5049] Bormann, C., Liu, Z., Price, R., and G. Camarillo,
"Applying Signaling Compression (SigComp) to the
Session Initiation Protocol (SIP)", RFC 5049,
December 2007.
[RFC5079] Rosenberg, J., "Rejecting Anonymous Requests in the
Session Initiation Protocol (SIP)", RFC 5079,
December 2007.
[RFC5360] Rosenberg, J., Camarillo, G., and D. Willis, "A
Framework for Consent-Based Communications in the
Session Initiation Protocol (SIP)", RFC 5360,
October 2008.
[RFC5361] Camarillo, G., "A Document Format for Requesting
Consent", RFC 5361, October 2008.
[RFC5362] Camarillo, G., "The Session Initiation Protocol
(SIP) Pending Additions Event Package", RFC 5362,
October 2008.
[RFC5363] Camarillo, G. and A. Roach, "Framework and Security
Considerations for Session Initiation Protocol
(SIP) URI-List Services", RFC 5363, October 2008.
[RFC5365] Garcia-Martin, M. and G. Camarillo, "Multiple-
Recipient MESSAGE Requests in the Session
Initiation Protocol (SIP)", RFC 5365, October 2008.
[RFC5366] Camarillo, G. and A. Johnston, "Conference
Establishment Using Request-Contained Lists in the
Session Initiation Protocol (SIP)", RFC 5366,
October 2008.
[RFC5367] Camarillo, G., Roach, A., and O. Levin,
"Subscriptions to Request-Contained Resource Lists
in the Session Initiation Protocol (SIP)",
RFC 5367, October 2008.
[RFC5368] Camarillo, G., Niemi, A., Isomaki, M., Garcia-
Martin, M., and H. Khartabil, "Referring to
Multiple Resources in the Session Initiation
Protocol (SIP)", RFC 5368, October 2008.
[RFC5373] Willis, D. and A. Allen, "Requesting Answering
Modes for the Session Initiation Protocol (SIP)",
RFC 5373, November 2008.
[RTCP-SUM] Clark, A., Pendleton, A., Johnston, A., and H.
Sinnreich, "Session Initiation Protocol Package for
Voice Quality Reporting Event", Work in Progress,
October 2008.
[SAML] Tschofenig, H., Hodges, J., Peterson, J., Polk, J.,
and D. Sicker, "SIP SAML Profile and Binding", Work
in Progress, November 2008.
[SDP-CAP] Andreasen, F., "SDP Capability Negotiation", Work
in Progress, July 2008.
[SDP-MEDIA] Gilman, R., Even, R., and F. Andreasen, "SDP media
capabilities Negotiation", Work in Progress,
July 2008.
[SESSION-POLICY] Hilt, V., Camarillo, G., and J. Rosenberg, "A
Framework for Session Initiation Protocol (SIP)
Session Policies", Work in Progress, November 2008.
[SIMPLE] Rosenberg, J., "SIMPLE made Simple: An Overview of
the IETF Specifications for Instant Messaging and
Presence using the Session Initiation Protocol
(SIP)", Work in Progress, October 2008.
[SIPS-URI] Audet, F., "The Use of the SIPS URI Scheme in the
Session Initiation Protocol (SIP)", Work
in Progress, November 2008.
[SRTP-FRAME] Fischl, J., Tschofenig, H., and E. Rescorla,
"Framework for Establishing an SRTP Security
Context using DTLS", Work in Progress,
October 2008.
[SUBNOT-ETAGS] Niemi, A., "An Extension to Session Initiation
Protocol (SIP) Events for Conditional Event
Notification", Work in Progress, July 2008.
[TRANSFER-MECH] Garcia, M., Isomaki, M., Camarillo, G., Loreto, S.,
and P. Kyzivat, "A Session Description Protocol
(SDP) Offer/Answer Mechanism to Enable File
Transfer", Work in Progress, November 2008.
[UA-PRIVACY] Munakata, M., Schubert, S., and T. Ohba, "UA-Driven
Privacy Mechanism for SIP", Work in Progress,
October 2008.
[UPDATE-PAI] Elwell, J., "Updates to Asserted Identity in the
Session Initiation Protocol (SIP)", Work
in Progress, October 2008.
Author's Address
Jonathan Rosenberg
Cisco
Iselin, NJ
US
EMail: jdrosen@cisco.com
URI: http://www.jdrosen.net
Full Copyright Statement
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