SPEERMINT Working Group J-F. Mule
Internet-Draft CableLabs
Intended status: Informational October 26, 2009
Expires: April 29, 2010
SPEERMINT Requirements for SIP-based Session Peering
draft-ietf-speermint-requirements-08.txt
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Abstract
This memo captures protocol requirements to enable session peering of
voice, presence, instant messaging and other types of multimedia
traffic. It is based on the use cases that have been described in
the SPEERMINT working group. This informational document is intended
to link the session peering use cases to protocol solutions.
Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . . 3
2. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 4
3. General Requirements . . . . . . . . . . . . . . . . . . . . . 5
3.1. Scope . . . . . . . . . . . . . . . . . . . . . . . . . . 5
3.2. Border Elements . . . . . . . . . . . . . . . . . . . . . 5
3.3. Session Establishment Data . . . . . . . . . . . . . . . . 9
3.3.1. User Identities and SIP URIs . . . . . . . . . . . . . 9
3.3.2. URI Reachability . . . . . . . . . . . . . . . . . . . 10
4. Requirements for Session Peering of Presence and Instant
Messaging . . . . . . . . . . . . . . . . . . . . . . . . . . 11
5. Security Considerations . . . . . . . . . . . . . . . . . . . 13
5.1. Security Properties for the Acquisition of Session
Establishment Data . . . . . . . . . . . . . . . . . . . . 13
5.2. Security Properties for the SIP signaling exchanges . . . 14
5.3. End-to-End Media Security . . . . . . . . . . . . . . . . 15
6. Acknowledgments . . . . . . . . . . . . . . . . . . . . . . . 16
7. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 17
8. References . . . . . . . . . . . . . . . . . . . . . . . . . . 18
8.1. Normative References . . . . . . . . . . . . . . . . . . . 18
8.2. Informative References . . . . . . . . . . . . . . . . . . 18
Appendix A. Policy Parameters for Session Peering . . . . . . . . 22
A.1. Categories of Parameters for VoIP Session Peering and
Justifications . . . . . . . . . . . . . . . . . . . . . . 22
A.2. Summary of Parameters for Consideration in Session
Peering Policies . . . . . . . . . . . . . . . . . . . . . 25
Author's Address . . . . . . . . . . . . . . . . . . . . . . . . . 27
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1. Introduction
Peering at the session level represents an agreement between parties
to exchange multimedia traffic. In this document, we assume that the
Session Initiation Protocol (SIP) is used to establish sessions
between SIP Service Providers (SSPs). SIP Service Providers are
referred to as peers and they are typically represented by users,
user groups, enterprises, real-time collaboration service
communities, or other service providers offering voice or multimedia
services using SIP.
A number of documents have been developed to provide background
information about SIP session peering. It is expected that the
reader is familiar with the reference architecture described in
[I-D.ietf-speermint-architecture], use cases for voice
([I-D.ietf-speermint-voip-consolidated-usecases]) and instant
messaging and presence ([RFC5344]).
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 a session 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. 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. Requirements specific to presence and instant messaging
are defined in Section 4.
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 [RFC2119].
This document also reuses the terminology defined in [RFC5486]. It
is assumed that the reader is familiar with the Session Description
Protocol (SDP) [RFC4566] and the Session Initiation Protocol (SIP)
[RFC3261]. Finally, when used with capital letters, the terms
'Authentication Service' are to be understood as defined by SIP
Identity [RFC4474].
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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, Signaling
path Border Elements (SBEs), and the associated protocol requirements
for the look-up and location routing of the session establishment
data. The requirements applicable to SIP User Agents or related to
the provisioning of the session data are considered out of scope.
SIP Service Providers have to reach an agreement on numerous points
when establishing session peering relationships .
This document highlights only certain aspects of a session peering
agreement, mostly the requirements relevant to protocols: the
declaration, advertisement and management of ingress and egress
border elements for session signaling and media, information related
to the Session Establishment Data (SED), and the security properties
that may be desirable for secure session exchanges.
Numerous other considerations of session peering arrangements are
critical to reach a successful agreement but they are considered out
of scope of this document. They include information about SIP
protocol support (e.g. SIP extensions and field conventions), media
(e.g., type of media traffic to be exchanged, compatible media codecs
and transport protocols, mechanisms to ensure differentiated quality
of service for media), layer-3 IP connectivity between the Signaling
and Data path Border Elements, accounting and traffic capacity
control (e.g. the maximum number of SIP sessions at each ingress
point, or the maximum number of concurrent IM or VoIP sessions).
The informative Appendix A lists parameters that may be considered
when discussing the technical parameters of SIP session peering. The
purpose of this list is to capture the parameters that are considered
outside the scope of the protocol requirements.
3.2. Border Elements
For border elements to be operationally manageable, maximum
flexibility should be given for how they are declared or dynamically
advertised. Indeed, in any session peering environment, there is a
need for a SIP Service Provider to declare or dynamically advertise
the SIP entities that will face the peer's network. The data path
border elements are typically signaled dynamically in the session
description.
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The use cases defined in
[I-D.ietf-speermint-voip-consolidated-usecases] catalog the various
border elements 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 be provided to enable a SIP Service
Provider 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 is
general agreement that [RFC3263] provides a solution for
dynamically advertising ingress SBEs in most cases of Direct or
Indirect peering. We discussed the DNS-based solution space
further in Requirement #4 below, especially in cases where the DNS
response varies based on who sends the query (peer-dependent
SBEs).
o Requirement #2:
Protocol mechanisms MUST be provided to enable a SIP Service
Provider 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. The SSP accepting calls from a
peer may wish to know where SIP calls will originate from (this
information is typically used by the terminating SSP).
While provisioning requirements are out-of-scope, some SSPs may
find use for a mechanism to dynamically advertise or discover the
egress SBEs of a peer.
If the SSP also provides media streams to its users as shown in the
use cases for "Originating" and "Terminating" SSPs, a mechanism must
exist to allow SSPs to advertise their egress and ingress data path
border elements (DBEs), if applicable. While some SSPs 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 MUST be provided to allow a SIP Service
Provider to communicate its DBEs to its peers.
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Notes: Some SSPs engaged in SIP interconnects do exchange this
type of DBE information in a static manner. Some SSPs do not.
In some SIP networks, SSPs may expose the same border elements to all
peers. In other environments, it is common for SSPs to advertise
specific SBEs and DBEs to certain peers. This is done by SSPs to
meet specific objectives for a given peer: routing optimization of
the signaling and media exchanges, optimization of the latency or
throughput based on the 'best' SBE and DBE combination, and other
service provider policy parameters. These are some of the reasons
why advertisement of SBEs and DBEs may be peer-dependent.
o Requirement #4:
The mechanisms recommended for the declaration or advertisement of
SBE and DBE entities MUST allow for peer variability.
Notes on solution space:
A simple solution is to advertise SBE entities using DNS and
[RFC3263] by providing different DNS names to different peers.
This approach has some practical limitations because the SIP URIs
containing the DNS names used to resolve the SBEs may be
propagated by users, for example in the form of sip:user@domain.
It is impractical to ask users to use different target URIs based
upon their SIP service provider's desire to receive incoming
session signaling at different ingress SBEs based upon the
originator. The solution described in [RFC3263] and based on DNS
to advertise SBEs is therefore under-specified for this
requirement.
Other DNS mechanisms have been used extensively in other areas of
the Internet, in particular in Content Distribution
Internetworking to make the DNS responses vary based on the
originator of the DNS query (see [RFC3466], [RFC3568] and
[RFC3570]). The applicability of such solutions needs for session
peering needs further analysis.
Finally, other techniques such as Anycast services ([RFC4786]) may
be employed at lower layers than Layer 5 to provide a solution to
this requirement. For example, anycast nodes could be defined by
SIP service providers to expose a common address for SBEs into
DNS, allowing the resolution of the anycast node address to the
appropriate peer-dependent service address based on the routing
topology or other criteria gathered from the combined use of
anycast and DNS techniques.
Notes on variability of the SBE advertisements based on the media
capabilities:
Some SSPs may have some restrictions on the type of media traffic
their SBEs can accept. For SIP sessions however, it is not
possible to communicate those restrictions in advance of the
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session initiation: a SIP target may support voice-only media,
voice and video, or voice and instant messaging communications.
While the inability to find out whether a particular type of SIP
session can be terminated by a certain SBE can cause failed
session establishment attempts, there is consensus to not add a
new requirement in this document. These aspects are essentially
covered by SSPs when discussing traffic exchange policies and are
deemed out of scope of this document.
In the use cases provided as part of direct and indirect peering
scenarios, an SSP deals with multiple SIP entities and multiple SBEs
in its own domain. There is often a many-to-many relationship
between the SIP Proxies considered inside the trusted network
boundary of the SSP and its Signaling path Border Elements at the
network boundaries.
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.
of [I-D.ietf-speermint-voip-consolidated-usecases]), it should be
possible for the SIP proxy in the originating network (O-Proxy) to
select the appropriate egress SBE (O-SBE) to reach the SIP target
based on the information the proxy receives from the Lookup Function
(O-LUF) and/or Location Routing Function (O-LRF) - message response
labeled (2). 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 Look-Up Function (LUF) and the
Location Routing Functions (LRF) MUST be capable of returning both
a target URI destination and a value providing the next SIP
hop(s).
Notes: solutions may exist depending on the choice of the protocol
used between the Proxy and its LUF/LRF. The idea is for the
O-Proxy to be provided with the next SIP hop and the equivalent of
one or more SIP Route header values. If ENUM is used as a
protocol for the LUF, the solution space is undefined.
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: some mechanisms exist. For example, the required protocol
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use of SIP over TLS may be discovered via [RFC3263] and guidelines
concerning the use of the SIPS URI scheme in SIP have been
documented in [RFC5630].
3.3. Session Establishment Data
The Session Establishment Data (SED) is defined in [RFC5486] 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
traffic.
The use of DNS domain names and hostnames is recommended in SIP URIs
and they should be resolvable on the public Internet. 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. The host part of SIP URIs
SHOULD contain a fully-qualified domain name instead of a numeric
IPv4 or IPv6 address.
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:
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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.
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.
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4. 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 [RFC5344], a document
authored by A. Houri, E. Aoki and S. Parameswar. Credits for the
original content captured in this section must go to them.
o Requirement #10:
The mechanisms recommended for the exchange of presence
information between SSPs SHOULD allow a user of one presence
community to send a presence subscription request to 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 [RFC5344].
o Requirement #11:
The mechanisms recommended for Instant Messaging exchanges between
SSPs SHOULD allow a user of one SSP's community to communicate
with users of the other SSP community via their local community
using the various methods. Note that some SSPs may exercise some
control over which methods are allowed based on service policies.
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,
etc.
Notes: see section 2.6 of [RFC5344].
o Requirement #12: Privacy Sharing
In some presence communities, users can define the list of
watchers that receive presence notifications for a given
presentity. Such privacy settings for watcher notifications per
presentity are typically not shared across SSPs causing multiple
notifications to be sent for one presentity change between SSPs.
The sharing of those privacy settings per presentity between SSPs
would allow fewer notifications: a single notification would be
sent per presentity and the terminating SSP would send
notifications to the appropriate watchers according to the
presentity's privacy information.
The mechanisms recommended for Presence information exchanges
between SSPs SHOULD allow the sharing of some user privacy
settings in order for users to convey the list of watchers that
can receive notification of presence information changes on a per
presentity basis.
The privacy sharing mechanism must be done with the express
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consent of the user whose privacy settings will be shared with to
the other community. Because of the privacy-sensitive information
exchanged between SSPs, the protocols used for the exchange of
presence information must follow the security recommendations
defined in section 6 of [RFC3863].
Notes: see section 2.3 of [RFC5344].
o Requirement #13: Multiple Watchers
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.
The systems used to exchange presence documents between SSPs
SHOULD allow more than one watchers to be passed with a presence
document.
o Requirement #14: Standard PIDF Documents and Mappings
Early deployments of SIP-based presence and Instant Messaging
gateways have been done in front of legacy proprietary systems
that use different naming schemes or name values for the elements
and properties defined in a Presence Information Data Format
(PIDF) document ([RFC3863]). For example the value "Do Not
Disturb" in one presence service may be mapped to "Busy" in
another system for the status element. Beyond this example of
status values, it is important to ensure that the meaning of the
presence information is preserved between SSPs.
The systems used to exchange presence documents between SSPs
SHOULD use standard PIDF documents and translate any non-standard
value of a PIDF element to a standard one.
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5. Security Considerations
This section describes the security properties that are desirable for
the protocol exchanges in scope of session peering. Three types of
information flows are described in the architecture and use case
documents: the acquisition of the Session Establishment Data (SED)
based on a destination target via the Lookup and Location Routing
Functions (LUF and LRF), the SIP signaling between SIP Service
Providers, and the associated media exchanges.
This section is focused on three security services, authentication,
data confidentiality and data integrity as summarized in [RFC3365].
However, this text does not specify the mandatory-to-implement
security mechanisms as required by [RFC3365]; this is left for future
protocol solutions that meet the requirements.
A security threat analysis provides additional guidance for session
peering ([I-D.niccolini-speermint-voipthreats]).
5.1. Security Properties for the Acquisition of Session Establishment
Data
The Look-Up Function (LUF) and Location Routing Function (LRF) are
defined in [RFC5486]. They provide mechanisms for determining the
SIP target address and domain the request should be sent to, and the
associated SED to route the request to that domain.
o Requirement #15:
The protocols used to query the Lookup and Location Routing
Functions SHOULD support mutual authentication.
Motivations:
A mutual authentication service is desirable for the LUF and LRF
protocol exchanges. The content of the response returned by the
LUF and LRF may depend on the identity of the requestor: the
authentication of the LUF & LRF requests is therefore a desirable
property. Mutual authentication is also desirable: the requestor
may verify the identity of the systems that provided the LUF & LRF
responses given the nature of the data returned in those
responses. Authentication also provides some protection for the
availability of the LUF and LRF against attackers that would
attempt to launch DoS attacks by sending bogus requests causing
the LUF to perform a lookup and consume resources.
o Requirement #16:
The protocols used to query the Lookup and Location Routing
Functions MUST provide support for data confidentiality and
integrity.
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Motivations:
Given the sensitive nature of the session establishment data
exchanged with the LUF and LRF functions, the protocol mechanisms
chosen for the lookup and location routing should offer data
confidentiality and integrity protection (SED data may contain
user addresses, SIP URI, location of SIP entities at the
boundaries of SIP Service Provider domains, etc.).
o Notes on the solution space for Requirements #15 and #16: ENUM,
SIP and proprietary protocols are typically used today for
accessing these functions. Even though SSPs may use lower layer
security mechanisms to guarantee some of those security
properties, candidate protocols for the LUF and LRF must meet the
above requirements.
5.2. Security Properties for the SIP signaling exchanges
The SIP signaling exchanges are out of scope of this document. This
section describes some of the security properties that are desirable
in the context of SIP interconnects between SSPs without formulating
any normative requirements.
In general, the security properties desirable for the SIP exchanges
in an inter-domain context apply to session peering. These include:
o securing the transport of SIP messages between the peers' SBEs.
Authentication of SIP communications is desirable, especially in
the context of session peering involving SIP intermediaries. Data
confidentiality and integrity of the SIP message body may be
desirable as well given some of the levels of session peering
indirection (indirect/assisted peering), but they could be harmful
as they may prevent intermediary SSPs from "inserting" SBEs/DBEs
along the signaling and data paths.
o providing an Authentication Service to authenticate the identity
of connected users based on the SIP Service Provider domains (for
both the SIP requests and the responses).
The fundamental mechanisms for securing SIP between proxy servers
intra- and inter-domain are applicable to session peering; refer to
Section 26.2 of [RFC3261] for transport-layer security of SIP
messages using TLS, [I-D.ietf-sip-connect-reuse] for establishing TLS
connections between proxies, [RFC4474] for the protocol mechanisms to
verify the identity of the senders of SIP requests in an inter-domain
context, and [RFC4916] for verifying the identity of the sender of
SIP responses).
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5.3. 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 present along the signaling
path. A number of desirable security properties emerge from this
goal.
The establishment of media security may be achieved along the media
path and not over the signaling path given the indirect peering use
cases.
For example, media carried over the Real-Time Protocol (RTP) can be
secured using secure RTP (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]).
It should also be noted that media can be carried in numerous
protocols other than RTP such as SIP (SIP MESSAGE method), MSRP (the
Message Session Relay Protocol, [RFC4975], XMPP (the Extensible
Messaging and Presence Protocol, [RFC3920]) and many others. Media
may also be carried over TCP ([RFC4571]), and it can be encrypted
over secure connection-oriented transport sessions over TLS
([RFC4572]).
A desirable security property for session peering is for SIP entities
to be transparent to the end-to-end media security negotiations: SIP
entities should not intervene in the Session Description Protocol
(SDP) exchanges for end-to-end media security.
o Requirement #17:
The protocols used to enable session peering MUST NOT interfere
with the exchanges of media security attributes in SDP. Media
attribute lines that are not understood by SBEs MUST be ignored
and passed along the signaling path untouched.
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6. Acknowledgments
This document is based on the input and contributions made by a large
number of people including: Bernard Aboba, 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, Peter Saint-
Andre, David Schwartz and Adam Uzelac.
Specials thanks go to Rohan Mahy, Brian Rosen, John Elwell for their
initial drafts describing guidelines or best current practices in
various environments, to Avshalom Houri, Edwin Aoki and Sriram
Parameswar for authoring the presence and instant messaging
requirements and to Dan Wing for providing detailed feedback on the
security consideration sections.
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7. IANA Considerations
This document does not register any values in IANA registries.
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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.ietf-avt-dtls-srtp]
McGrew, D. and E. Rescorla, "Datagram Transport Layer
Security (DTLS) Extension to Establish Keys for Secure
Real-time Transport Protocol (SRTP)",
draft-ietf-avt-dtls-srtp-07 (work in progress),
February 2009.
[I-D.ietf-pmol-sip-perf-metrics]
Malas, D. and A. Morton, "SIP End-to-End Performance
Metrics", draft-ietf-pmol-sip-perf-metrics-04 (work in
progress), September 2009.
[I-D.ietf-sip-connect-reuse]
Gurbani, V., Mahy, R., and B. Tate, "Connection Reuse in
the Session Initiation Protocol (SIP)",
draft-ietf-sip-connect-reuse-14 (work in progress),
August 2009.
[I-D.ietf-sip-dtls-srtp-framework]
Fischl, J., Tschofenig, H., and E. Rescorla, "Framework
for Establishing an SRTP Security Context using DTLS",
draft-ietf-sip-dtls-srtp-framework-07 (work in progress),
March 2009.
[I-D.ietf-speermint-architecture]
Penno, R. and S. Khan, "SPEERMINT Peering Architecture",
draft-ietf-speermint-architecture-08 (work in progress),
March 2009.
[I-D.ietf-speermint-voip-consolidated-usecases]
Uzelac, A. and Y. Lee, "VoIP SIP Peering Use Cases",
draft-ietf-speermint-voip-consolidated-usecases-14 (work
in progress), August 2009.
[I-D.niccolini-speermint-voipthreats]
Niccolini, S., Chen, E., Seedorf, J., and H. Scholz,
"SPEERMINT Security Threats and Suggested
Countermeasures", draft-niccolini-speermint-voipthreats-05
(work in progress), October 2008.
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[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.
[RFC3365] Schiller, J., "Strong Security Requirements for Internet
Engineering Task Force Standard Protocols", BCP 61,
RFC 3365, August 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.
[RFC3466] Day, M., Cain, B., Tomlinson, G., and P. Rzewski, "A Model
for Content Internetworking (CDI)", RFC 3466,
February 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.
[RFC3568] Barbir, A., Cain, B., Nair, R., and O. Spatscheck, "Known
Content Network (CN) Request-Routing Mechanisms",
RFC 3568, July 2003.
[RFC3570] Rzewski, P., Day, M., and D. Gilletti, "Content
Internetworking (CDI) Scenarios", RFC 3570, July 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)",
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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.
[RFC3863] Sugano, H., Fujimoto, S., Klyne, G., Bateman, A., Carr,
W., and J. Peterson, "Presence Information Data Format
(PIDF)", RFC 3863, August 2004.
[RFC3920] Saint-Andre, P., Ed., "Extensible Messaging and Presence
Protocol (XMPP): Core", RFC 3920, October 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.
[RFC4571] Lazzaro, J., "Framing Real-time Transport Protocol (RTP)
and RTP Control Protocol (RTCP) Packets over Connection-
Oriented Transport", RFC 4571, 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.
[RFC4786] Abley, J. and K. Lindqvist, "Operation of Anycast
Services", BCP 126, RFC 4786, December 2006.
[RFC4916] Elwell, J., "Connected Identity in the Session Initiation
Protocol (SIP)", RFC 4916, June 2007.
[RFC4975] Campbell, B., Mahy, R., and C. Jennings, "The Message
Session Relay Protocol (MSRP)", RFC 4975, September 2007.
[RFC5344] Houri, A., Aoki, E., and S. Parameswar, "Presence and
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Instant Messaging Peering Use Cases", RFC 5344,
October 2008.
[RFC5411] Rosenberg, J., "A Hitchhiker's Guide to the Session
Initiation Protocol (SIP)", RFC 5411, February 2009.
[RFC5486] Malas, D. and D. Meyer, "Session Peering for Multimedia
Interconnect (SPEERMINT) Terminology", RFC 5486,
March 2009.
[RFC5503] Andreasen, F., McKibben, B., and B. Marshall, "Private
Session Initiation Protocol (SIP) Proxy-to-Proxy
Extensions for Supporting the PacketCable Distributed Call
Signaling Architecture", RFC 5503, March 2009.
[RFC5630] Audet, F., "The Use of the SIPS URI Scheme in the Session
Initiation Protocol (SIP)", RFC 5630, October 2009.
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Appendix A. Policy Parameters for Session Peering
This informative section lists various types of parameters that
should be considered by implementers when deciding what configuration
variables to expose to system administrators or management stations,
as well as SSPs or federations of SSPs when discussing the technical
part of a session peering policy.
In the context of session peering, a policy can be defined as the set
of parameters and other information needed by an SSP to exchange
traffic with another peer. 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 be session-dependent, or the may apply
over multiple sessions or peers.
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.
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 for VoIP Session Peering 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.
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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 payload 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.
* Media variability at the Signaling path Border Elements: list
of media types supported by the various ingress points of a
peer's network.
* 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 [RFC5411].
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
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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 [RFC5503], 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.ietf-pmol-sip-perf-metrics]; 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:
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 described above.
Finally, it is possible that some requirements be imposed on
lower layers, but these are considered out of scope of session
peering.
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* 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 constraints 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
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
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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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