SPEERMINT Working Group J-F. Mule
Internet-Draft CableLabs
Intended status: Informational February 25, 2008
Expires: August 28, 2008
SPEERMINT Requirements for SIP-based VoIP Interconnection
draft-ietf-speermint-requirements-04.txt
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Abstract
A number of use cases have been described for session peering of
voice, presence, instant messaging and other types of multimedia
traffic. This memo captures some of the requirements identified by
these use case scenarios. It is intended to become an informational
document linking the use cases to potential protocol solutions.
Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . . 3
2. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 4
3. General Requirements . . . . . . . . . . . . . . . . . . . . . 5
3.1. Scope . . . . . . . . . . . . . . . . . . . . . . . . . . 5
3.2. Session Peering Points . . . . . . . . . . . . . . . . . . 5
3.3. Session Establishment Data . . . . . . . . . . . . . . . . 8
3.3.1. User Identities and SIP URIs . . . . . . . . . . . . . 8
3.3.2. URI Reachability . . . . . . . . . . . . . . . . . . . 9
3.4. Other Considerations . . . . . . . . . . . . . . . . . . . 10
4. Considerations and Requirements for Session Peering of
Presence and Instant Messaging . . . . . . . . . . . . . . . . 12
5. Security Requirements . . . . . . . . . . . . . . . . . . . . 14
5.1. Security in SIP networks in the context of session
peering . . . . . . . . . . . . . . . . . . . . . . . . . 14
5.2. Security Requirements for the Lookup and Location
Routing Data . . . . . . . . . . . . . . . . . . . . . . . 14
5.3. Hop-by-hop Security for SIP Signaling and TLS
Considerations . . . . . . . . . . . . . . . . . . . . . . 15
5.4. End-to-End Media Security . . . . . . . . . . . . . . . . 16
6. Acknowledgments . . . . . . . . . . . . . . . . . . . . . . . 17
7. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 18
8. Security Considerations . . . . . . . . . . . . . . . . . . . 19
9. References . . . . . . . . . . . . . . . . . . . . . . . . . . 20
9.1. Normative References . . . . . . . . . . . . . . . . . . . 20
9.2. Informative References . . . . . . . . . . . . . . . . . . 20
Appendix A. Policy Parameters for Session Peering . . . . . . . . 23
A.1. Categories of Parameters and Justifications . . . . . . . 23
A.2. Summary of Parameters for Consideration in Session
Peering Policies . . . . . . . . . . . . . . . . . . . . . 26
Author's Address . . . . . . . . . . . . . . . . . . . . . . . . . 27
Intellectual Property and Copyright Statements . . . . . . . . . . 28
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1. Introduction
Peering at the session level represents an agreement between parties
to allow the exchange of multimedia traffic. It is assumed that
these sessions use the Session Initiation Protocol (SIP) protocol to
enable peering between two or more actors. These actors are called
SIP Service Providers (SSPs) and they are typically represented by
users, user groups such as enterprises, real-time collaboration
service communities, or other service providers offering voice or
multimedia services.
Common terminology for SIP session peering is defined
([I-D.ietf-speermint-terminology]) and a reference architecture is
described in [I-D.ietf-speermint-architecture]. A number of use
cases have been exposed by users of SIP services and various other
actors describing how layer-5 peering has been or could be deployed
based on the reference architecture
([I-D.ietf-speermint-voip-consolidated-usecases] and
[I-D.ietf-speermint-consolidated-presence-im-usecases]).
Peering at the session layer can be achieved on a bilateral basis
(direct peering established directly between two SSPs), or on an
indirect basis via an intermediary (indirect peering via a third-
party SSP that has a trust relationship with the SSPs) - see the
terminology document for more details.
This document first describes general requirements that have been
derived from the working group discussions. The use cases are then
analyzed in the spirit of extracting relevant protocol requirements
that must be met to accomplish the use cases. These requirements are
intended to be independent of the type of media exchanged such as
Voice over IP (VoIP), video telephony, and instant messaging. In the
case where some requirements are media-specific, we define them in a
separate section.
It is not the goal of this document to mandate any particular use of
IETF protocols by SIP Service Providers in order to establish session
peering. Instead, the document highlights what requirements should
be met and what protocols may be used to define the solution space.
Finally, we conclude with a list of parameters for the definition of
a session peering policy, provided in an informative appendix. It
should be considered as an example of the information SIP Service
Providers may have to discuss or agree on to exchange SIP traffic.
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2. Terminology
The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL
NOT","SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in
this document are to be interpreted as described in RFC 2119
[RFC2119].
This document also reuses the SIP terminology defined in [I-D.ietf-
speermint-terminology]. It is assumed that the reader is familiar
with the Session Description Protocol (SDP) [RFC4566] and the Session
Initiation Protocol (SIP) [RFC3261].
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3. General Requirements
The following sub-sections contain general requirements applicable to
multiple use cases for multimedia session peering.
3.1. Scope
The primary focus of this document is on the requirements applicable
to the boundaries of Layer 5 SIP networks: SIP entities and Signaling
path Border Elements (SBEs); any requirements touching SIP UA or end-
devices are considered out of scope.
SSPs desiring to establish session peering relationships have to
reach an agreement on numerous aspects.
This document highlights only certain aspects of a session peering
agreement, mostly the requirements relevant to protocols, including
the declaration, advertisement and management of ingress and egress
for session signaling and media, information related to the Session
Establishment Data (SED), and the security mechanisms a peer may use
to accept and secure session exchanges.
Numerous other aspects of session peering arrangement are critical to
reach a successful agreement but they are considered out of scope of
the SPEERMINT working group. They include aspects such as SIP
protocol support (e.g. SIP extensions and field conventions), media
(e.g., type of media traffic to be exchanged, compatible media codecs
and media transport protocols, mechanisms to ensure differentiated
quality of service for media), SIP layer-3 IP connectivity between
the Signaling Path and Data Path Border Elements, traffic capacity
control (e.g. maximum number of SIP sessions at each ingress point,
maximum number of concurrent IM or VoIP sessions), and accounting.
The informative Appendix A lists parameters that SPPs may consider
when discussing the technical aspects of SIP session peering. The
purpose of this list which has evolved through the working group use
case discussions is to capture the parameters that are considered
outside the scope of the protocol requirements.
3.2. Session Peering Points
For session peering to be scalable and operationally manageable,
maximum flexibility should be given for how signaling path and media
path border elements are declared, dynamically advertised and
updated.
Indeed, in any session peering environment, there is a need for a SIP
Service Provider to declare or dynamically advertise the SIP entities
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that will face the peer's network. The media or data path border
elements are typically signaled dynamically in the session
description.
The use cases defined
([I-D.ietf-speermint-voip-consolidated-usecases]) catalog the various
session peering points between SIP Service Providers; they include
Signaling Path Border Elements (SBEs) and SIP proxies (or any SIP
entity at the boundary of the Layer 5 network).
o Requirement #1:
Protocol mechanisms must exist for a SIP Service Provider (SSP) to
communicate the ingress Signaling Path Border Elements of its
service domain.
Notes on solution space:
The SBEs may be advertised to session peers using static
mechanisms or they may be dynamically advertised. There seems to
be general agreement that [RFC3263] provides a solution for
dynamically advertising ingress SBEs in most cases of Direct or
Indirect peering. However, this DNS-based solution may be limited
in cases where the DNS response varies based on who sends the
query (peer-dependent SBEs, see below).
o Requirement #2:
Protocol mechanisms should exist for a SIP Service Provider (SSP)
to communicate the egress SBEs of its service domain.
Notes on motivations for this requirement:
For the purposes of capacity planning, traffic engineering and
call admission control, a SIP Service Provider may be asked where
it will generate SIP calls from. Note that this may not be
applicable to all types of session peering (voice may be a
particular case where this is needed -- at least based on current
practices).
If the SSP also provides media streams to its users as shown in the
use cases for "Originating" and "Terminating" SSPs, a mechanism
should exist to allow SSPs to advertise their media border elements
responsible for egress and ingress data path border elements (DBEs),
if applicable. While some SPPs may have open policies and accept
media traffic from anywhere outside their network to anywhere inside
their network, some SSPs may want to optimize media delivery and
identify media paths between peers prior to traffic being sent (layer
5 to layer 3 QoS mapping).
o Requirement #3:
Protocol mechanisms should be available to allow a SIP Service
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Provider to communicate its DBEs to its peers.
Notes: Some SSPs engaged in SIP interconnects do exchange this
type of DBE information today in a static manner. Some SSPs do
not.
Some SSPs may have some restrictions on the type of media traffic
their SIP entities acting as SBEs are capable of establishing. In
order to avoid a failed attempt to establish a session, a mechanism
may be provided to allow SSPs to indicate if some restrictions exist
on the type of media traffic: ingress and egress SBE points may be
peer-dependent, and/or media-dependent.
o Requirement #4:
The mechanisms recommended for the declaration or advertisement of
SBE and DBE entities must allow for peer and media variability.
Notes on solution space:
For advertising peer-dependent SBEs (peer variability), the
solution space based on [RFC3263] is under specified and there are
no know best current practices. Is DNS the right place for
putting data that varies based on who asks?
For advertising media-dependent SBEs, solutions exist as long as
URIs are protocol-dependent URIs. A protocol-dependent URI like a
SIP URI can be mapped to more than one types of media. It should
be noted that some URIs like the IM URI are abstract ([RFC3428])
and need to be translated to protocol dependent URIs. It is also
not possible to know what media is supported by the SIP SBE before
initiating a query by using mechanisms like [RFC3263].
The following example provides some additional motivations for the
above requirement on advertising media-dependent SBEs to peers.
In large multi-service SIP networks, an SSP chooses to have several
SBEs for receiving incoming SIP session requests (ingress SBEs), and
several SBEs for outgoing SIP session requests (egress SBEs). In
order to facilitate the operations, feature management, and
maintenance of its SBEs, the SSP opts for having distinct SBEs for
voice, real-time collaboration, etc. Some SBEs are therefore
dedicated to exchanging certain types of media traffic due to
specific SIP extensions required for certain media types (e.g.
SIMPLE, the SIP MESSAGE Method for Instant Messaging [RFC3428] or the
Message Sessions Relay Protocol (MSRP)). Note that this example is
applicable to some enterprise networks where IP voice traffic hits
different SIP gateways and voice servers (e.g. IP-PBX) than Instant
Messaging and real-time collaboration servers (e.g. real-time
collaboration and IM server supporting SIMPLE and XMPP).
In the use cases provided as part of direct and indirect scenarios,
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an SSP deals with multiple SIP entities and multiple SBEs in its own
domain. There is often a many-to-many relationship between SIP
Proxies and Signaling path Border Elements.
It should be possible for an SSP to define which egress SBE a SIP
entity must use based on a given peer destination. For example, in
the case of an indirect peering scenario (section 5.1.5 of
[I-D.ietf-speermint-voip-consolidated-usecases], Figure 5), it should
be possible for the O-Proxy to choose the appropriate O-SBE based on
the information the O-Proxy receives from the Lookup Function (LUF)
or Location Routing Function (LRF) - message response labeled (3).
Note that this example also applies to the case of Direct Peering
when a service provider has multiple service areas and each service
area involves multiple SIP Proxies and a few SBEs.
o Requirement #5:
The mechanisms recommended for the lookup and location routing
service must be capable or returning both a target URI destination
and a SIP Route.
Notes: solutions exist if the protocol used between the Proxy and
the LUF/LRF is SIP; if ENUM is used, the author of this document
does not know of any solution today.
It is desirable for an SSP to be able to communicate how
authentication of a peer's SBEs will occur (see the security
requirements for more details).
o Requirement #6:
The mechanisms recommended for locating a peer's SBE must be able
to convey how a peer should initiate secure session establishment.
Notes : certain mechanisms exist; for example, the required
protocol use of SIP over TLS may be discovered via [RFC3263].
3.3. Session Establishment Data
The Session Establishment Data (SED) is defined in
[I-D.ietf-speermint-terminology] as the data used to route a call to
the next hop associated with the called domain's ingress point. The
following paragraphs capture some general requirements on the SED
data.
3.3.1. User Identities and SIP URIs
User identities used between peers can be represented in many
different formats. Session Establishment Data should rely on URIs
(Uniform Resource Identifiers, [RFC3986]) and SIP URIs should be
preferred over tel URIs ([RFC3966]) for session peering of VoIP
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traffic.
The use of DNS domain names and hostnames is recommended in SIP URIs
and they should be resolvable on the public Internet. It is
recommended that the host part of SIP URIs contain a fully-qualified
domain name instead of a numeric IPv4 or IPv6 address. As for the
user part of the SIP URIs, the mechanisms for session peering should
not require an SSP to be aware of which individual user identities
are valid within its peer's domain.
o Requirement #7:
The protocols used for session peering must accommodate the use of
different types of URIs. URIs with the same domain-part should
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 Requirement #8:
The mechanisms for session peering should not require an SSP to be
aware of which individual user identities are valid within its
peer's domain.
o Notes on the solution space for #7 and #8:
This is generally well supported by IETF protocols. 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 [RFC3824]. This means that
the solutions built for session peering must not solely use PSTN
identifiers such as Service Provider IDs (SPIDs) or Trunk Group
IDs (they should not be precluded but solutions should not be
limited to these).
Motivations:
Although SED data may be based on E.164-based SIP URIs for voice
interconnects, a generic peering methodology should not rely on
such E.164 numbers.
3.3.2. URI Reachability
Based on a well-known URI type (for e.g. sip, pres, or im URIs), it
must be possible to determine whether the SSP domain servicing the
URI allows for session peering, and if it does, it should be possible
to locate and retrieve the domain's policy and SBE entities.
For example, an originating service provider must be able to
determine whether a SIP URI is open for direct interconnection
without requiring an SBE 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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o Requirement #9:
The mechanisms for session peering must allow an SBE to locate its
peer SBE given a URI type and the target SSP domain name.
3.4. Other Considerations
The considerations listed below were gathered early on in the
SPEERMINT working group as part of discussions to define the scope of
the working group. They have not been updated in this revision of
the draft.
o It is assumed that session peering is independent of lower layers.
The mechanisms used to establish session peering should
accommodate diverse supporting lower layers. 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:
Mechanisms developed for session peering should be flexible and
extensible to cover existing and future session peering models.
It is also recommended that SSP 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 an SPP 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 an SSP's session-dependent
or session independent policies as defined in
[I-D.ietf-sipping-session-policy].
o Administrative and Technical Policies:
Various types of policy information may need to be discovered or
exchanged in order to establish session peering. At a minimum, a
policy should specify information related to session establishment
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, see examples in Section
Appendix A.
Motivations:
The reasons for declining or accepting incoming calls from a
prospective peering partner can be both administrative
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(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
incompatible policy parameters will cause the session to fail
(this was originally referred to as "no blocked calls").
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4. Considerations and Requirements for Session Peering of Presence and
Instant Messaging
This section describes requirements for presence and instant
messaging session peering. Several use cases for presence and
instant messaging peering are described in
[I-D.ietf-speermint-consolidated-presence-im-usecases], a document
authored by A. Houri, E. Aoki and S. Parameswar. Credits for this
section must go to A. Houri, E. Aoki and S. Parameswar.
The following requirements for presence and instant messaging session
peering are derived from
[I-D.ietf-speermint-consolidated-presence-im-usecases] and
[I-D.houri-speermint-presence-im-requirements]:
o Requirement #10:
The mechanisms recommended for the exchange of presence
information between SSPs MUST allow a user of one SSP's presence
community to subscribe presentities served by another SSP via its
local community, including subscriptions to a single presentity, a
personal, public or ad-hoc group list of presentities.
Notes: see section 2.2 of
[I-D.ietf-speermint-consolidated-presence-im-usecases].
o Requirement #11:
The mechanisms recommended for Instant Messaging message exchanges
between SSPs MUST allow a user of one SSP's community to
communicate with users of the other SSP community via their local
community using various methods. Such methods include sending a
one-time IM message, initiating a SIP session for transporting
sessions of messages, participating in n-way chats using chat
rooms with users from the peer SSPs, sending a file or sharing a
document.
Notes: see section 2.6 of
[I-D.ietf-speermint-consolidated-presence-im-usecases].
o Requirement #12: Privacy Sharing
In order to enable sending less notifications between communities,
there should be a mechanism that will enable sharing privacy
information of users between the communities. This will enable
sending a single notification per presentity that will be sent to
the appropriate watchers on the other community according to the
presentity's privacy information.
The privacy sharing mechanism must be done in a way that will
enable getting the consent of the user whose privacy will be sent
to the other community prior to sending the privacy information.
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if user consent is not give, it should not be possible to this
optimization. In addition to getting the consent of users
regarding privacy sharing, the privacy data must be sent only via
secure channels between communities.
Notes: see section 2.3 of
[I-D.ietf-speermint-consolidated-presence-im-usecases].
o Requirement #13: Multiple Recipients
It should be possible to send a presence document with a list of
watchers on the other community that should receive the presence
document notification. This will enable sending less presence
document notifications between the communities while avoiding the
need to share privacy information of presentities from one
community to the other.
o Requirement #14: Mappings
Early deployments of SIP based presence and IM gateways are done
in front of legacy proprietary systems that use different names
for different properties that exist in PIDF. For example "Do Not
Disturb" may be translated to "Busy" in another system. In order
to make sure that the meaning of the status is preserved, there is
a need that either each system will translate its internal
statuses to standard PIDF based statuses of a translation table of
proprietary statuses to standard based PIDF statuses will be
provided from one system to the other.
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5. Security Requirements
Session peering does bring a new environment in which security
requirements should be analyzed but the fundamental mechanisms for
securing SIP and media exchanges remain applicable (see Section 26.2
of [RFC3261]. The issues are less in the mechanisms that do exist
and can be used to mitigate threats than they are in getting two SSPs
to agree on which ones to use.
This section first provides a broad picture of the various mechanisms
used today in the context of SIP session peering. We then describe
security considerations for the three types of information flows
described in the use cases: the data queried from the Lookup or
Location Routing Functions, data exchanged in the SIP signaling
between SSPs (directly and indirectly), and media.
5.1. Security in SIP networks in the context of session peering
In today's SIP deployments, various approaches exist to secure
exchanges between SIP Service Providers. Lookup, signaling and media
security are the three primary topics for consideration in most
deployments.
A number of transport, network and session-level mechanisms are used
for SIP by some categories of SSPs. TLS is used in the enterprise
networks for applications such as VoIP and secure Instant Messaging
and session-level security is used end-to-end for some instant
messaging systems or in service provider networks for Instant
Messaging and presence applications. At the network-level, IPsec and
L2/L3 VPNs are widely used in some SSP networks where there is a
desire to secure all signaling and media traffic at or below the IP
layer.
Media level security between providers is not widely used today for
media transported using the Real-Time Protocol (RTP), even though it
is in use in few deployments where the privacy of voice and other RTP
media is critical.
A security threat analysis provides guidance for session peering
([I-D.draft-niccolini-speermint-voipthreats]). More discussions
based on this threat analysis and use cases continue to be required
in the working group to define what hop-by-hop or end-to-end security
requirements are necessary in the context of session peering.
5.2. Security Requirements for the Lookup and Location Routing Data
The Look-Up Function (LUF) and Location Routing Function (LRF) are
defined in [I-D.ietf-speermint-terminology]. They provide a
mechanism for determining for a given request the target domain to
which the request should be routed, and SED required to route the
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request to that domain.
Requirement #15:
The protocols used for the LUF and LRF must allow the look-up and SED
data to be exchanged securely (authentication and encryption services
should be provided).
Notes on the solution space: ENUM, SIP and proprietary protocols are
typically used today for accessing these functions.
5.3. Hop-by-hop Security for SIP Signaling and TLS Considerations
Given the direct and indirect peering uses cases referenced in the
previous sections of this document, hop-by-hop security between two
SSPs using Transport Layer Security (TLS) is desirable.
The Transport Layer Security (TLS) is a standard way to secure
signaling between SIP entities. TLS can be used in direct peering to
mutually authenticate SSPs and provide message confidentiality and
integrity protection. The remaining paragraphs explore how TLS could
be deployed and used between 2 SSPs to secure SIP exchanges. The
intent is to capture what two SSPs should discuss and agree on in
order to establish TLS connections for SIP session peering.
1. One or more Certificate Authorities (CAs) should be agreed
between SSPs for securing session peering exchanges.
Alternatively, self-signed certificates may also be used.
Motivations:
An SSP should have control over which root CAs it trusts for SIP
communications. This may imply creating a certificate trust list
and including the peer's CA for each authorized domain. In the
case of a federation, 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 an SSP
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.
There are various use cases of direct peering where there is no
pre-established trust relationship that can rely on self-signed
certificates.
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]. Federations supporting
[RFC4474] and CA(s) must specify the CA(s) permitted to issue
certificates of the authentication service.
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3. SIP entities and SBEs 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 as
defined in [RFC3261].
4. The following SIP and TLS protocol parameters should be agreed
upon as part of session peering policies: the version of TLS
supported by SIP entities and SBEs (TLSv1, TLSv1.1), the SIP TLS
port (default 5061), the server-side session timeout (default 300
seconds), the list of supported or recommended ciphersuites, the
list of trusted root CAs if applicable or whether self-signed
certs are acceptable.
5. SIP entities and SBEs involved in the session establishment
over TLS must verify and validate the client certificates. See
section 9 and 9.3 of [I-D.ietf-sip-certs].
6. A session peering policy should include details on SIP session
establishment over TLS if TLS is supported.
5.4. End-to-End Media Security
Media security is critical to guarantee end-to-end confidentiality of
the communication between the end-users' devices, independently of
how many direct or indirect peers are along the signaling path.
o Requirement #16:
It is recommended that the establishment of media security be
provided along the media path and not over the signaling path
given the indirect peering use cases.
Notes on the solution space:
Media carried over the Real-Time Protocol (RTP) can be secured
using secure RTP or sRTP ([RFC3711]). A framework for
establishing sRTP security using Datagram TLS [RFC4347] is
described in [I-D.ietf-sip-dtls-srtp-framework]: it allows for
end-to-end media security establishment using extensions to DTLS
([I-D.ietf-avt-dtls-srtp]). This DTLS-SRTP framework meets the
above requirement.
Note that media can also be carried in numerous protocols other than
RTP such as SIP (SIP MESSAGE method), MSRP, XMPP, etc. In these
cases, the above requirement is also met given the security features
of these protocols.
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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: Edwin Aoki, Scott Brim, John Elwell, Mike
Hammer, Avshalom Houri, Richard Shocky, Henry Sinnreich, Richard
Stastny, Patrik Faltstrom, Otmar Lendl, Daryl Malas, Dave Meyer,
Sriram Parameswar, Jon Peterson, Jason Livingood, Bob Natale, Benny
Rodrig, Brian Rosen, Eric Rosenfeld, Adam Uzelac and Dan Wing.
Specials thanks go to Rohan Mahy, Brian Rosen, John Elwell for their
initial drafts describing guidelines or best current practices in
various environments, and to Avshalom Houri, Edwin Aoki and Sriram
Parameswar for authoring the presence and instant messaging
requirements.
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7. IANA Considerations
None.
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8. 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 [RFC3261],
and [RFC4474] are applicable to the SIP protocol exchanges. A number
of security considerations are also described in Section Section 5.
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9. References
9.1. Normative References
[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
Requirement Levels", BCP 14, RFC 2119, March 1997.
9.2. Informative References
[I-D.draft-malas-performance-metrics]
Malas, D., "SIP End-to-End Performance Metrics",
December 2007.
[I-D.draft-niccolini-speermint-voipthreats]
Niccolini, S., Chen, E., and J. Seedorf, "VoIP Security
Threats relevant to SPEERMINT",
draft-niccolini-speermint-voipthreats-03.txt (work in
progress), February 2008.
[I-D.houri-speermint-presence-im-requirements]
Houri, A., Aoki, E., and S. Parameswar, "Presence and IM
Requirements", May 2007.
[I-D.ietf-avt-dtls-srtp]
McGrew, D. and E. Rescorla, "DTLS Extensions to Establish
Keys for SRTP", draft-ietf-avt-dtls-srtp-01.txt (work in
progress), November 2007.
[I-D.ietf-sip-certs]
Jennings, C., Peterson, J., and J. Fischl, "Certificate
Management Service for The Session Initiation Protocol
(SIP)", draft-ietf-sip-certs-05.txt (work in progress),
January 2008.
[I-D.ietf-sip-dtls-srtp-framework]
Fischl, J., Tschofenig, H., and E. Rescorla, "DTLS-SRTP
Framework", draft-ietf-sip-dtls-srtp-framework-01 (work in
progress), February 2008.
[I-D.ietf-sip-hitchhikers-guide]
Rosenberg, J., "A Hitchhikers Guide to the Session
Initiation Protocol (SIP)", July 2007.
[I-D.ietf-sipping-session-policy]
Hilt, V. and G. Camarillo, "A Session Initiation Protocol
(SIP) Event Package for Session-Specific Session
Policies", draft-ietf-sipping-policy-package-04.txt (work
in progress), August 2007.
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[I-D.ietf-speermint-architecture]
Penno et al., R., "SPEERMINT Peering Architecture",
draft-ietf-speermint-architecture-04.txt (work in
progress), August 2007.
[I-D.ietf-speermint-consolidated-presence-im-usecases]
Houri, A., Aoki, E., and S. Parameswar, "Presence &
Instant Messaging Peering Use Cases",
draft-ietf-speermint-consolidated-presence-im-usecases-04
(work in progress), February 2008.
[I-D.ietf-speermint-terminology]
Meyer, R. and D. Malas, "SPEERMINT Terminology",
draft-ietf-speermint-terminology-16.txt (work in
progress), February 2008.
[I-D.ietf-speermint-voip-consolidated-usecases]
Uzelac et al., A., "VoIP SIP Peering Use Cases",
draft-ietf-speermint-voip-consolidated-usecases-05.txt
(work in progress), February 2008.
[RFC2198] Perkins, C., Kouvelas, I., Hodson, O., Hardman, V.,
Handley, M., Bolot, J., Vega-Garcia, A., and S. Fosse-
Parisis, "RTP Payload for Redundant Audio Data", RFC 2198,
September 1997.
[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.
[RFC3263] Rosenberg, J. and H. Schulzrinne, "Session Initiation
Protocol (SIP): Locating SIP Servers", RFC 3263,
June 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.
[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.
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[RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V.
Jacobson, "RTP: A Transport Protocol for Real-Time
Applications", STD 64, RFC 3550, July 2003.
[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.
[RFC3711] Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K.
Norrman, "The Secure Real-time Transport Protocol (SRTP)",
RFC 3711, March 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.
[RFC4347] Rescorla, E. and N. Modadugu, "Datagram Transport Layer
Security", RFC 4347, April 2006.
[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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Appendix A. Policy Parameters for Session Peering
This informative section lists various types of parameters that
should be first considered by implementers when deciding what
configuration parameters to expose to system admins or management
stations, and second, by SSPs or federations of SSPs when discussing
the technical aspects of a session peering policy.
Some aspects of session peering policies must be agreed to and
manually implemented; they are static and are typically documented as
part of a business contract, technical document or agreement between
parties. For some parameters linked to protocol support and
capabilities, standard ways of expressing those policy parameters may
be defined among SSP and exchanged dynamically. For e.g., templates
could be created in various document formats so that it could be
possible to dynamically discover some of the domain policy. Such
templates could be initiated by implementers (for each software/
hardware release, a list of supported RFCs, RFC parameters is
provided in a standard format) and then adapted by each SSP based on
its service description, server or device configurations and variable
based on peer relationships.
A.1. Categories of Parameters and Justifications
The following list should be considered as an initial list of
"discussion topics" to be addressed by peers when initiating a VoIP
peering relationship.
o IP Network Connectivity:
Session peers should define how the IP network connectivity
between their respective SBEs and DBEs. While this is out of
scope of session peering, SSPs must agree on a common mechanism
for IP transport of 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.
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* 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 private 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].
o Presence and Instant Messaging: TBD
o Accounting:
Methods used for call or session accounting should be specified.
An SSP may require a peer to track session usage. 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 SSP 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.draft-malas-performance-metrics]; a subset of these may be
applicable to session peering and interconnects.
Some media-related metrics for monitoring VoIP calls have been
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defined in the VoIP Metrics Report Block, in Section 4.7 of
[RFC3611].
o Security:
An SSP 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 SSP 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
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 an SSP. Indeed,
some SSPs 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: an SSP 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 SSP 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
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Section 5.
A.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
* Modes of communications for audio (voice, fax, DTMF), IM (page
mode, MSRP)
* Media transport and means to establish secure media sessions
* List of ingress and egress DBEs where applicable, including
STUN Relay servers if present
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 Capacity Control and Performance Management: any limits on, or,
means to measure and limit the maximum number of active calls to a
peer or federation, maximum number of sessions and messages per
specified unit time, maximum number of active users or subscribers
per specified unit time, the aggregate media bandwidth per peer or
for the federation, specified SIP signaling performance metrics to
measure and report; media-level VoIP metrics if applicable.
o Security: Call admission control, call authorization, network and
transport layer security parameters, media security parameters
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Author's Address
Jean-Francois Mule
CableLabs
858 Coal Creek Circle
Louisville, CO 80027
USA
Email: jf.mule@cablelabs.com
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