SPEERMINT Working Group J-F. Mule
Internet-Draft CableLabs
Expires: April 26, 2007 October 23, 2006
SPEERMINT Requirements for SIP-based VoIP Interconnection
draft-ietf-speermint-requirements-01.txt
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Copyright Notice
Copyright (C) The Internet Society (2006).
Abstract
This document describes high-level guidelines and general
requirements for Session PEERing for Multimedia INTerconnect. It
also defines a minimum set of requirements applicable to session
peering for Voice over IP interconnects. It is intended to become
best current practices based on the use cases discussed in the
speermint working group.
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Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . . 3
2. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 4
3. General Requirements . . . . . . . . . . . . . . . . . . . . . 5
4. Requirements for SIP-based VoIP Interconnection . . . . . . . 8
4.1. DNS, Call Addressing Data (CAD) and ENUM . . . . . . . . . 8
4.2. Minimum set of SIP-SDP-related requirements . . . . . . . 8
4.3. Media-related Requirements . . . . . . . . . . . . . . . . 9
4.4. Security Requirements . . . . . . . . . . . . . . . . . . 9
4.4.1. Security in today's VoIP networks . . . . . . . . . . 9
4.4.2. TLS Considerations for session peering . . . . . . . . 10
5. Annex A - List of Policy Parameters for VoIP
Interconnections . . . . . . . . . . . . . . . . . . . . . . . 12
5.1. Categories of parameters and Justifications . . . . . . . 12
5.2. Summary of Parameters for Consideration in Session
Peering Policies . . . . . . . . . . . . . . . . . . . . . 14
6. Acknowledgments . . . . . . . . . . . . . . . . . . . . . . . 16
7. Security Considerations . . . . . . . . . . . . . . . . . . . 17
8. References . . . . . . . . . . . . . . . . . . . . . . . . . . 18
8.1. Normative References . . . . . . . . . . . . . . . . . . . 18
8.2. Informative References . . . . . . . . . . . . . . . . . . 18
Author's Address . . . . . . . . . . . . . . . . . . . . . . . . . 21
Intellectual Property and Copyright Statements . . . . . . . . . . 22
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1. Introduction
The Session PEERing for Multimedia INTerconnect (SPEERMINT) Working
Group is chartered to focus on architectures to identify, signal, and
route delay-sensitive communication sessions. These sessions use the
Session Initiation Protocol (SIP) protocol to enable peering between
two or more administrative domains over IP networks.
This document describes high-level guidelines and general
requirements for session peering; these requirements are applicable
to any type of multimedia session peering such as Voice over IP
(VoIP), video telephony, and instant messaging. The document also
defines a minimum set of requirements for a sub-set of the session
peering use cases: VoIP interconnects.
The intent of this version of this document is to describe what
mechanisms are used for establishing SIP session peering with a
special look at VoIP interconnects, and in doing so, it defines some
of requirements associated with the secure establishment of VoIP
interconnects between a large number of peers.
The primary focus is on the requirements applicable to the boundaries
of layer-5 SIP networks: SIP UA or end-device requirements are
considered out of scope.
It is also not the goal of this document to mandate any particular
use of any IETF protocols to establish session peering by users or
service providers. However, when protocol mechanisms are used, the
document aims at providing guidelines or best current practices on
how they should be implemented, or configured and enabled in order to
facilitate session peering.
Finally, a list of parameters for the definition of a session peering
policy is provided in an informative annex. It should be considered
as an example of the information a Voice Service Provider, or
Application Service Provider may require in order to connect to
another using SIP.
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2. 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
[RFC2119].
This specification makes use of terms defined in
[I-D.ietf-speermint-terminology], the Session Description Protocol
(SDP) [RFC4566] and the Session Initiation Protocol (SIP) [RFC3261].
We also use the terms Voice Service Provider (VSP) and Application
Service Provider (ASP) as defined in [I-D.ietf-ecrit-requirements].
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3. General Requirements
The following section defines general guidelines and requirements
applicable to session peering for multimedia sessions.
o Session peering should be independent of lower layers. The
mechanisms used to establish session peering SHOULD accommodate
diverse supporting lower layers.
Motivations:
Session peering is about layer 5 mechanisms. It should not matter
whether lower layers rely on the public Internet or are
implemented by private L3 connectivity, using firewalls or L2/L3
Virtual Private Networks (VPNs), IPSec tunnels or Transport Layer
Security (TLS) connections [RFC3546]...
o Session Peering Policies and Extensibility:
Policies developed for session peering SHOULD be flexible and
extensible to cover existing and future session peering models.
It is also RECOMMENDED that policies be published via local
configuration choices in a distributed system like DNS rather than
in a centralized system like a 'peering registry'.
In the context of session peering, a policy is defined as the set
of parameters and other information needed by one VSP/ASP to
connect to another. Some of the session policy parameters may be
statically exchanged and set throughout the lifetime of the
peering relationship. Others parameters may be discovered and
updated dynamically using by some explicit protocol mechanisms.
These dynamic parameters may also relate to a VSP/ASP's session-
dependent or session independent policies as defined in
[I-D.ietf-sipping-session-policy-framework].
Motivations:
It is critical that the solutions be flexible and extensible given
the various emerging models: layer 5 peering may involve open
federations of SIP proxies, or closed environments with systems
that only accept incoming calls from selected peers based on a set
of bilateral trust relationships. Federations may also be based
on memberships in peering fabrics or voice service provider clubs,
etc. Session peering may be direct or indirect.
The maintenance of the "system" should scale beyond simple lists
of peering partners. In particular, it must incorporate
aggregation mechanisms which avoid O(n^2) scaling (where n is the
number of participating peers). The distributed management of the
DNS is a good example for the scalability of this approach.
o Administrative and Technical Policies:
Various types of policy information may need to be discovered or
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exchanged in order to establish session peering. At a minimum, a
policy SHOULD specify information related to call addressing data
in order to avoid session establishment failures. A policy MAY
also include information related to QoS, billing and accounting,
layer-3 related interconnect requirements which are out of the
scope of this document.
Motivations:
The reasons for declining or accepting incoming calls from a
prospective peering partner can be both administrative
(contractual, legal, commercial, or business decisions) and
technical (certain QoS parameters, TLS keys, domain keys, ...).
The objectives are to provide a baseline framework to define,
publish and optionally retrieve policy information so that a
session establishment does not need to be attempted to know that
imcompatible policy parameters will cause the session to fail
(this was originally referred to as "no blocked calls").
o URIs and Domain-Based Peering Context:
Call Addressing Data SHOULD rely on URIs (Uniform Resource
Identifiers, RFC 3986 [RFC3986]) for call routing and SIP URIs
SHOULD be preferred over tel URIs (RFC 3966 [RFC3966]). Although
the initial call addressing data may be based on E.164 numbers for
voice interconnects, a generic peering methodology SHOULD NOT rely
on such E.164 numbers.
Motivations:
Telephone numbers commonly appear in the username portion of a SIP
URI. When telephone numbers are in tel URIs, SIP requests cannot
be routed in accordance with the traditional DNS resolution
procedures standardized for SIP as indicated in RFC 3824
[RFC3824]. Furthermore, we assume that all SIP URIs with the same
domain-part share the same set of peering policies, thus the
domain of the SIP URI may be used as the primary key to any
information regarding the reachability of that SIP URI.
o URI Reachability and Minimal additional cost on call initiation:
Based on a well-known URI (for e.g. sip, pres, or im URIs), it
MUST be possible to determine whether the domain servicing the URI
(VSP/ASP) allows for session peering, and if it does, it SHOULD be
possible to locate and retrieve the domain's policy and signaling
functions. For example, an originating service provider must be
able to determine whether a SIP URI is open for direct
interconnection without requiring to initiate a SIP request.
Furthermore, since each call setup implies the execution of any
proposed algorithm, the establishment of a SIP session via peering
SHOULD incur minimal overhead and delay, and employ caching
wherever possible to avoid extra protocol round trips.
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Motivations:
This requirement is important as unsuccessful call attempts are
highly undesirable since they can introduce high delays due to
timeouts and can act as an unintended denial of service attack
(e.g., by repeated TLS handshakes). There should be a high
probability of successful call completion for policy-conforming
peers.
o Variability of the Call Address Data:
A terminating VSP/ASP or user SHOULD be able to indicate its
domain ingress points (Signaling Path Border Element(s)) based on
the identity of the originating VSP/ASP or user.
The mechanisms recommended for the use and resolution of the call
addressing data SHOULD allow for variability or customization of
the response(s) depending on various elements, such as the
identity of the originating or terminating user or user domain.
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4. Requirements for SIP-based VoIP Interconnection
This section defines some requirements for SIP-based VoIP
Interconnection. It should be considered as the minimal set of
requirements to be implemented to perform SIP VoIP interconnects.
4.1. DNS, Call Addressing Data (CAD) and ENUM
Call Addressing Data can be derived from various mechanisms available
to the user, such as ENUM when the input is a telephone number, or
other DNS queries using SRV and NAPTR resource records when the entry
is a SIP URI for example. The SPEERMINT Working Group is focused on
the use of CAD.
The following requirements are best current practices for VoIP
session peering:
o SIP URIs SHOULD be preferred over tel URIs when establishing a SIP
session for voice interconnects.
o The recommendations defined in [RFC3824] SHOULD be followed by
implementers when using E.164 numbers with SIP, and by authors of
NAPTR records for ENUM for records with an 'E2U+sip' service
field. Other ENUM implementation issues and experiences are
described in [I-D.ietf-enum-experiences] that may be relevant for
VoIP interconnects using ENUM.
o The use of DNS domain names and hostnames is RECOMMENDED in SIP
URIs and they MUST be resolvable on the public Internet.
o The DNS procedures specified in [RFC3263] SHOULD be followed to
resolve a SIP URI into a reachable host (IP address and port), and
transport protocol. Note that RFC 3263 relies on DNS SRV
[RFC2782] and NAPTR Resource Records [RFC2915].
4.2. Minimum set of SIP-SDP-related requirements
The main objective of VoIP interconnects being the establishment of
successful SIP calls between peer VSPs/ASPs, this section provides a
minimum set of SIP-related requirements.
o The Core SIP Specifications as defined in [RFC3261] and
[I-D.ietf-sip-hitchhikers-guide] MUST be supported by Signaling
Path Border Elements and any other SIP implementations involved in
session peering.
Justifications:
The specifications contained in the Core SIP group provide the
fundamental and basic mechanisms required to enable VoIP
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interconnects. This includes: the SIP protocol for session
establishment and its updates such as RFC 3853 and RFC 4320, SDP
[RFC4566] and its Offer/Answer model [RFC3264] for VoIP media
session descriptions and codec negotiations, SIP Asserted Identity
for caller ID services, and various other extensions to support
NAT traversal, etc.
o The following RFCs SHOULD be supported: Reliability of Provisional
Responses in SIP - PRACK [RFC3262], the SIP UPDATE method (for
e.g. for codec changes during a session) [RFC3311], the Reason
header field [RFC3326].
In the context of session peering where peers desire to maximize the
chances of successful call establishment, the recommendations
contained in RFC 3261 regarding the use of the Supported and Require
headers MUST be followed. Signaling Path Border Elements SHOULD
include the supported SIP extensions in the Supported header and the
use of the Require header must be configurable on a per target domain
basis in order to match a network peer policy and to maximize
interoperability.
4.3. Media-related Requirements
VSPs engaged in session peering SHOULD support of compatible codecs
and include media-related parameters in their domain's policy.
Transcoding SHOULD be avoided by proposing commonly agreed codecs.
Motivations: The media capabilities of a VSP's network are either a
property of the SIP end-devices, or, a combination of the property of
end-devices and Data Path Border Elements that may provide media
transcoding. The choice of one or more common codecs for VoIP
sessions between VSPs is therefore outside the scope of speermint.
Indeed, as stated in introduction, requirements applicable to end-
devices of a VSP are considered out of scope. A list of media-
related policy parameters are provided in the informative Section 5.
4.4. Security Requirements
4.4.1. Security in today's VoIP networks
In today's VoIP deployments, various approaches exist to secure
exchanges between VSPs/ASPs. Signaling and media security are the
two primary topics for consideration in most deployments. A number
of transport-layer and network-layer mechanisms are widely used by
some categories of VSPs: TLS in the enterprise networks for
applications such as VoIP and secure Instant Messaging, IPSec and
L2/L3 VPNs in some VSP networks where there is a desire to secure all
signaling and media traffic at or below the IP layer. Media level
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security is not widely deployed for RTP, even though it is in use in
few deployments where the privacy of voice communications is
critical.
A detailed security threat analysis of session peering exchanges
should provide more guidance on what scalable and efficient methods
should be used to help mitigate the the main security risks in large-
scale session peering.
A recent IETF BoF at IETF 66 (rtpsec) was organized to analyze SIP
requirements for SRTP keying; a number of security requirements for
VoIP were discussed. A few Internet-Drafts have since been released
and focus on media security requirements for SIP sessions
([I-D.ietf-wing-media-security-requirements]). Some of these
scenarios may be applicable to interdomain VSP/ASP session peering or
they may be augmented in the future by interdomain scenarios.
4.4.2. TLS Considerations for session peering
The remaining of Section 4 covers some details on how TLS could be
deployed and used between 2 VSPs/ASPs to secure SIP exchanges. The
intent is to capture what two VSPs/ASPs should discuss and agree on
in order to establish TLS connections for SIP session peering.
1. Peers SHOULD agree on one or more Certificate Authorities
(CAs) to trust for securing session peering exchanges.
Motivations:
A VSP/ASP should have control over which root CA it trusts for SIP
communications. This may imply creating a certificate trust list
and including the peer's CA for each authorized domain. This
requirement allows for the initiating side to verify that the
server certificate chains up to a trusted root CA. This also
means that SIP servers SHOULD allow the configuration of a
certificate trust list in order to allow a VSP/ASP to control
which peer's CAs are trusted for TLS connections. Note that these
considerations seem to be around two themes: one is trusting a
root, the other is trusting intermediate CAs.
2. Peers SHOULD indicate whether their domain policies require
proxy servers to inspect and verify the identity provided in SIP
requests as defined in [RFC4474].
3. SIP servers involved in the secure session establishment over
TLS MUST have valid X.509 certificates and MUST be able to receive
a TLS connection on a well-known port.
4. The following TLS/SIP Protocol parameters SHOULD be agreed
upon as part of session peering policies: the version of TLS
supported by Signaling Border Elements (TLSv1, TLSv1.1), the SIP
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TLS port (default 5061), the server-side session timeout (default
300 seconds), the list of supported or recommended ciphersuites,
and the list of trusted root CAs.
5. SIP servers involved in the session establishment over TLS
MUST verify and validate the client certificates: the client
certificate MUST contain a DNS or URI choice type in the
subjectAltName which corresponds to the domain asserted in the
host portion of the URI contained in the From header. It is also
recommended that VSPs/ASPs convey the domain identity in the
certificates using both a canonical name of the SIP server(s) and
the SIP URI for the domain as described in section 4 of
[I-D.gurbani-sip-domain-certs]. On the client side, it is also
critical for the TLS client to authenticate the server as defined
in [RFC3261] and in section 9 of draft-ietf-sip-certs-01.txt.
6. A session peering policy SHOULD include details on SIP session
establishment over TLS if TLS is supported.
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5. Annex A - List of Policy Parameters for VoIP Interconnections
This informative annex lists the various types of parameters that
should be considered when discussing the technical aspects of a VoIP
Peering policy .
5.1. Categories of parameters and Justifications
It is intended as an initial list of topics that should be addressed
by peers when establishing a VoIP peering relationship.
o IP Network Connectivity:
It is assumed that IP network connectivity exists between peers.
While this is out of scope of session peering, VSPs must agree
upon a common mechanism for IP transport of Layer 5 session
signaling and media. This may be accomplish via private (e.g.
IPVPN, IPSEC, etc.) or public IP networks.
o Media-related Parameters:
* Media Codecs: list of supported media codecs for audio, real-
time fax (version of T.38, if applicable), real-time text (RFC
4103), DTMF transport, voice band data communications (as
applicable) along with the supported or recommended codec
packetization rates, level of RTP paylod redundancy, audio
volume levels, etc.
* Media Transport: level of support for RTP-RTCP [RFC3550], RTP
Redundancy (RTP Payload for Redundant Audio Data - [RFC2198]) ,
T.38 transport over RTP, etc.
* Other: support of the VoIP metric block as defined in RTP
Control Protocol Extended Reports [RFC3611] , etc.
o SIP:
* A session peering policy SHOULD include the list of supported
and required SIP RFCs, supported and required SIP methods
(including p headers if applicable), error response codes,
supported or recommended format of some header field values ,
etc.
* It should also be possible to describe the list of supported
SIP RFCs by various functional groupings. A group of SIP RFCs
may represent how a call feature is implemented (call hold,
transfer, conferencing, etc.), or it may indicate a functional
grouping as in [I-D.ietf-sip-hitchhikers-guide].
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o Accounting:
Call accounting may be required for tracking session usage on a
peer's network. It is critical for peers to determine whether the
support of any SIP extensions for accounting is a pre-requisite
for SIP interoperability. In some cases, call accounting may feed
data for billing purposes but not always: some operators may
decide to use accounting as a 'bill and keep' model to track
session usage and monitor usage against service level agreements.
[RFC3702] defines the terminology and basic requirements for
accounting of SIP sessions. A few private SIP extensions have
also been defined and used over the years to enable call
accounting between VSP domains such as the P-Charging* headers in
[RFC3455], the P-DCS-Billing-Info header in [RFC3603], etc.
o Performance Metrics:
Layer-5 performance metrics should be defined and shared between
peers. The performance metrics apply directly to signaling or
media; they may be used pro-actively to help avoid congestion,
call quality issues or call signaling failures, and as part of
monitoring techniques, they can be used to evaluate the
performance of peering exchanges.
Examples of SIP performance metrics include the maximum number of
SIP transactions per second on per domain basis, Session
Completion Rate (SCR), Session Establishment Rate (SER), etc.
Some SIP end-to-end performance metrics are defined in
[I-D.Malas-sip-performance]; a subset of these may be applicable
to session peering and interconnects.
Some media-related metrics for monitoring VoIP calls have been
defined in the VoIP Metrics Report Block, in Section 4.7 of
[RFC3611].
o Security:
A VSP/ASP SHOULD describe the security requirements that other
peers must meet in order to terminate calls to its network. While
such a list of security-related policy parameters often depends on
the security models pre-agreed to by peers, it is expected that
these parameters will be discoverable or signaled in the future to
allow session peering outside VSP clubs. The list of security
parameters may be long and composed of high-level requirements
(e.g. authentication, privacy, secure transport) and low level
protocol configuration elements like TLS parameters.
The following list is not intended to be complete, it provides a
preliminary list in the form of examples:
* Call admission requirements: for some providers, sessions can
only be admitted if certain criteria are met. For example, for
some providers' networks, only incoming SIP sessions signaled
over established IPSec tunnels or presented to the well-known
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TLS ports are admitted. Other call admission requirements may
be related to some performance metrics as descrived above.
Finally, it is possible that some requiremetns be imposed on
lower layers, but these are considered out of scope of session
peering.
* Call authorization requirements and validation: the presence of
a caller or user identity MAY be required by a VSP/ASP.
Indeed, some VSPs/ASPs may further authorize an incoming
session request by validating the caller's identity against
white/black lists maintained by the service provider or users
(traditional caller ID screening applications or IM white
list).
* Privacy requirements: a VSP/ASP MAY demand that its SIP
messages be securely transported by its peers for privacy
reasons so that the calling/called party information be
protected. Media sessions may also require privacy and some
ASP/VSP policies may include requirements on the use of secure
media transport protocols such as sRTP, along with some
contraints on the minimum authentication/encryption options for
use in sRTP.
* Network-layer security parameters: this covers how IPSec
security associated may be established, the IPSec key exchange
mechanisms to be used and any keying materials, the lifetime of
timed Security Associated if applicable, etc.
* Transport-layer security parameters: this covers how TLS
connections should be established as described in Section 4.4.2
5.2. Summary of Parameters for Consideration in Session Peering
Policies
The following is a summary of the parameters mentioned in the
previous section. They may be part of a session peering policy and
appear with a level of requirement (mandatory, recommended,
supported, ...).
o IP Network Connectivity (assumed, requirements out of scope of
this document)
o Media session parameters:
* Codecs for audio, video, real time text, instant messaging
media sessions
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* Modes of communications for audio (voice, fax, DTMF), IM (page
mode, MSRP)
* Media transport and means to establish secure media sessions
o SIP
* SIP RFCs, methods and error responses
* headers and header values
* possibly, list of SIP RFCs supported by groups (e.g. by call
feature)
o Accounting
o Performance Metrics: SIP signaling performance metrics; media-
level VoIP metrics.
o Security: Call admission control, call authorization, network and
transport layer security parameters, media security parameters
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6. Acknowledgments
This document is a work-in-progress and it is based on the input and
contributions made by a large number of people in the SPEERMINT
working group, including: Scott Brim, Mike Hammer, Richard Shocky,
Henry Sinnreich, Richard Stastny, Patrik Faltstrom, Otmar Lendl,
Daryl Malas, Dave Meyer, Jason Livingood, Bob Natale, Brian Rosen,
Eric Rosenfeld and Adam Uzelac.
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7. Security Considerations
Securing session peering communications involves numerous protocol
exchanges, first and foremost, the securing of SIP signaling and
media sessions. The security considerations contained in RF 3261,
RFC 4474 are applicable to the SIP protocol exchanges. A number of
security considerations are also described in Section 4.4 for VoIP
Interconnects.
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8. References
8.1. Normative References
[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
Requirement Levels", BCP 14, RFC 2119, March 1997.
8.2. Informative References
[I-D.Malas-sip-performance]
Malas, D., "SIP End-to-End Performance Metrics",
September 2006.
[I-D.gurbani-sip-domain-certs]
Gurbani, V., Jeffrey, A., and S. Lawrence, "Domain
Certificates in the Session Initiation Protocol (SIP)",
draft-gurbani-sip-domain-certs-03 (work in progress),
August 2006.
[I-D.ietf-ecrit-requirements]
Schulzrinne, H. and R. Marshall, "Requirements for
Emergency Context Resolution with Internet Technologies",
August 2006.
[I-D.ietf-enum-experiences]
Conroy, L. and K. Fujiwara, "ENUM Implementation Issues
and Experiences", June 2006.
[I-D.ietf-sip-hitchhikers-guide]
Rosenberg, J., "A Hitchhikers Guide to the Session
Initiation Protocol (SIP)", October 2006.
[I-D.ietf-sipping-session-policy-framework]
Hilt, V., "A Framework for Session Initiation Protocol
(SIP) Session Policies",
draft-ietf-sipping-session-policy-framework-01 (work in
progress), June 2006.
[I-D.ietf-speermint-terminology]
Meyer, R., "SPEERMINT Terminology", September 2006.
[I-D.ietf-wing-media-security-requirements]
Wing, D., Fries, S., and H. Tschofenig, "A Framework for
Session Initiation Protocol (SIP) Session Policies",
draft-wing-media-security-requirements-00 (work in
progress), October 2006.
[RFC2198] Perkins, C., Kouvelas, I., Hodson, O., Hardman, V.,
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Handley, M., Bolot, J., Vega-Garcia, A., and S. Fosse-
Parisis, "RTP Payload for Redundant Audio Data", RFC 2198,
September 1997.
[RFC2782] Gulbrandsen, A., Vixie, P., and L. Esibov, "A DNS RR for
specifying the location of services (DNS SRV)", RFC 2782,
February 2000.
[RFC2915] Mealling, M. and R. Daniel, "The Naming Authority Pointer
(NAPTR) DNS Resource Record", RFC 2915, September 2000.
[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.
[RFC3311] Rosenberg, J., "The Session Initiation Protocol (SIP)
UPDATE Method", RFC 3311, October 2002.
[RFC3326] Schulzrinne, H., Oran, D., and G. Camarillo, "The Reason
Header Field for the Session Initiation Protocol (SIP)",
RFC 3326, December 2002.
[RFC3455] Garcia-Martin, M., Henrikson, E., and D. Mills, "Private
Header (P-Header) Extensions to the Session Initiation
Protocol (SIP) for the 3rd-Generation Partnership Project
(3GPP)", RFC 3455, January 2003.
[RFC3546] Blake-Wilson, S., Nystrom, M., Hopwood, D., Mikkelsen, J.,
and T. Wright, "Transport Layer Security (TLS)
Extensions", RFC 3546, June 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.
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[RFC3603] Marshall, W. and F. Andreasen, "Private Session Initiation
Protocol (SIP) Proxy-to-Proxy Extensions for Supporting
the PacketCable Distributed Call Signaling Architecture",
RFC 3603, October 2003.
[RFC3611] Friedman, T., Caceres, R., and A. Clark, "RTP Control
Protocol Extended Reports (RTCP XR)", RFC 3611,
November 2003.
[RFC3702] Loughney, J. and G. Camarillo, "Authentication,
Authorization, and Accounting Requirements for the Session
Initiation Protocol (SIP)", RFC 3702, February 2004.
[RFC3824] Peterson, J., Liu, H., Yu, J., and B. Campbell, "Using
E.164 numbers with the Session Initiation Protocol (SIP)",
RFC 3824, June 2004.
[RFC3966] Schulzrinne, H., "The tel URI for Telephone Numbers",
RFC 3966, December 2004.
[RFC3986] Berners-Lee, T., Fielding, R., and L. Masinter, "Uniform
Resource Identifier (URI): Generic Syntax", STD 66,
RFC 3986, January 2005.
[RFC4474] Peterson, J. and C. Jennings, "Enhancements for
Authenticated Identity Management in the Session
Initiation Protocol (SIP)", RFC 4474, August 2006.
[RFC4566] Handley, M., Jacobson, V., and C. Perkins, "SDP: Session
Description Protocol", RFC 4566, July 2006.
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Author's Address
Jean-Francois Mule
CableLabs
858 Coal Creek Circle
Louisville, CO 80027
USA
Email: jf.mule@cablelabs.com
Mule Expires April 26, 2007 [Page 21]
Internet-Draft SPEERMINT Requirements October 2006
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Mule Expires April 26, 2007 [Page 22]
