Identification of Communications Services in the Session Initiation Protocol (SIP)
RFC 5897
| Document | Type | RFC - Informational (June 2010; No errata) | |
|---|---|---|---|
| Author | Jonathan Rosenberg | ||
| Last updated | 2015-10-14 | ||
| Stream | Internet Engineering Task Force (IETF) | ||
| Formats | plain text html pdf htmlized bibtex | ||
| Reviews | |||
| Stream | WG state | (None) | |
| Document shepherd | No shepherd assigned | ||
| IESG | IESG state | RFC 5897 (Informational) | |
| Action Holders |
(None)
|
||
| Consensus Boilerplate | Unknown | ||
| Telechat date | |||
| Responsible AD | Robert Sparks | ||
| Send notices to | (None) | ||
Internet Engineering Task Force (IETF) J. Rosenberg
Request for Comments: 5897 jdrosen.net
Category: Informational June 2010
ISSN: 2070-1721
Identification of Communications Services
in the Session Initiation Protocol (SIP)
Abstract
This document considers the problem of service identification in the
Session Initiation Protocol (SIP). Service identification is the
process of determining the user-level use case that is driving the
signaling being utilized by the user agent (UA). This document
discusses the uses of service identification, and outlines several
architectural principles behind the process. It identifies perils
when service identification is not done properly -- including fraud,
interoperability failures, and stifling of innovation. It then
outlines a set of recommended practices for service identification.
Status of This Memo
This document is not an Internet Standards Track specification; it is
published for informational purposes.
This document is a product of the Internet Engineering Task Force
(IETF). It represents the consensus of the IETF community. It has
received public review and has been approved for publication by the
Internet Engineering Steering Group (IESG). Not all documents
approved by the IESG are a candidate for any level of Internet
Standard; see Section 2 of RFC 5741.
Information about the current status of this document, any errata,
and how to provide feedback on it may be obtained at
http://www.rfc-editor.org/info/rfc5897.
Copyright Notice
Copyright (c) 2010 IETF Trust and the persons identified as the
document authors. All rights reserved.
This document is subject to BCP 78 and the IETF Trust's Legal
Provisions Relating to IETF Documents
(http://trustee.ietf.org/license-info) in effect on the date of
publication of this document. Please review these documents
carefully, as they describe your rights and restrictions with respect
to this document. Code Components extracted from this document must
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include Simplified BSD License text as described in Section 4.e of
the Trust Legal Provisions and are provided without warranty as
described in the Simplified BSD License.
Table of Contents
1. Introduction ....................................................3
2. Services and Service Identification .............................4
3. Example Services ................................................6
3.1. IPTV vs. Multimedia ........................................6
3.2. Gaming vs. Voice Chat ......................................7
3.3. Gaming vs. Voice Chat #2 ...................................7
3.4. Configuration vs. Pager Messaging ..........................7
4. Using Service Identification ....................................8
4.1. Application Invocation in the User Agent ...................8
4.2. Application Invocation in the Network ......................9
4.3. Network Quality-of-Service Authorization ..................10
4.4. Service Authorization .....................................10
4.5. Accounting and Billing ....................................11
4.6. Negotiation of Service ....................................11
4.7. Dispatch to Devices .......................................11
5. Key Principles of Service Identification .......................12
5.1. Services Are a By-Product of Signaling ....................12
5.2. Identical Signaling Produces Identical Services ...........13
5.3. Do What I Say, Not What I Mean ............................14
5.4. Declarative Service Identifiers Are Redundant .............15
5.5. URIs Are Key for Differentiated Signaling .................15
6. Perils of Declarative Service Identification ...................16
6.1. Fraud .....................................................16
6.2. Systematic Interoperability Failures ......................17
6.3. Stifling of Service Innovation ............................18
7. Recommendations ................................................20
7.1. Use Derived Service Identification ........................20
7.2. Design for SIP's Negotiative Expressiveness ...............20
7.3. Presence ..................................................21
7.4. Intra-Domain ..............................................21
7.5. Device Dispatch ...........................................21
8. Security Considerations ........................................22
9. Acknowledgements ...............................................22
10. Informative References ........................................22
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1. Introduction
The Session Initiation Protocol (SIP) [RFC3261] defines mechanisms
for initiating and managing communications sessions between agents.
SIP allows for a broad array of session types between agents. It can
manage audio sessions, ranging from low-bitrate voice-only up to
multi-channel high-fidelity music. It can manage video sessions,
ranging from small, "talking-head" style video chat, up to high-
definition multipoint video conferencing and ranging from low-
bandwidth user-generated content, up to high-definition movie and TV
content. SIP endpoints can be anything -- adaptors that convert an
old analog telephone to Voice over IP (VoIP), dedicated hardphones,
fancy hardphones with rich displays and user entry capabilities,
softphones on a PC, buddy-list and presence applications on a PC,
dedicated videoconferencing peripherals, and speakerphones.
This breadth of applicability is SIP's greatest asset, but it also
introduces numerous challenges. One of these is that, when an
endpoint generates a SIP INVITE for a session, or receives one, that
session can potentially be within the context of any number of
different use cases and endpoint types. For example, a SIP INVITE
with a single audio stream could represent a Push-To-Talk session
between mobile devices, a VoIP session between softphones, or audio-
based access to stored content on a server.
Each of these different use cases represents a different service.
The service is the user-visible use case that is driving the behavior
of the user agents and servers in the SIP network.
The differing services possible with SIP have driven implementors and
system designers to seek techniques for service identification.
Service identification is the process of determining and/or signaling
the specific use case that is driving the signaling being generated
by a user agent. At first glance, this seems harmless and easy
enough. It is tempting to define a new header, "Service-ID", for
example, and have a user agent populate it with any number of well-
known tokens that define what the service is. It could then be
consumed for any number of purposes. A token placed into the
signaling for this purpose is called a service identifier.
Service identification and service identifiers, when used properly,
can be beneficial. However, when done improperly, service
identification can lead to fraud, systemic interoperability failures,
and a complete stifling of the innovation that SIP was meant to
achieve. The purpose of this document is to describe service
identification in more detail and describe how these problems arise.
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Section 2 begins by defining a service and the service identification
problem. Section 3 gives some concrete examples of services and why
they can be challenging to identify. Section 4 explores the ways in
which a service identification can be utilized within a network.
Next, Section 5 discusses the key architectural principles of service
identification. Section 6 describes what declarative service
invocation is, and how it can lead to fraud, interoperability
failures, and stifling of service innovation.
Consequently, this document concludes that declarative service
identification -- the process by which a user agent inserts a moniker
into a message that defines the desired service, separate from
explicit and well-defined protocol mechanisms -- is harmful.
Instead of performing declarative service identification, this
document recommends derived service identification, and gives several
recommendations around it in Section 7:
1. The identity of a service should always be derived from the
explicit signaling in the protocol messages and other contextual
information, and never indicated by the user through a separate
identifier placed into the message.
2. The process of service identification based on signaling messages
must be designed to SIP's negotiative expressiveness, and
therefore handle heterogeneity and not assume a fixed set of use
cases.
3. Presence can help in providing URIs that can be utilized to
connect to specific services, thereby creating explicit
indications in the signaling that can be used to derive a service
identity.
4. Service identities placed into signaling messages for the
purposes of caching the service identity are strictly for intra-
domain usage.
5. Device dispatch should be based on feature tags that map to well-
defined SIP extensions and capabilities. Service dispatch should
not be based on abstract service identifiers.
2. Services and Service Identification
The problem of identifying services within SIP is not a new one. The
problem has been considered extensively in the context of presence.
In particular, the presence data model for SIP [RFC4479] defines the
concept of a service as one of the core notions that presence
describes. Services are described in Section 3.3 of RFC 4479.
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Essentially, the service is the user-visible use case that is driving
the behavior of the user agents and servers in the SIP network.
Being user-visible means that there is a difference in user
experience between two services that are different. That user
experience can be part of the call, or outside of the call. Within a
call, the user experience can be based on different media types (an
audio call vs. a video chat), different content within a particular
media type (stored content, such as a movie or TV session), different
devices (a wireless device for "telephony" vs. a PC application for
"voice chat"), different user interfaces (a buddy-list view of voice
on a PC application vs. a software emulation of a hardphone),
different communities that can be accessed (voice chat with other
users that have the same voice chat client vs. voice communications
with any endpoint on the Public Switched Telephone Network (PSTN)),
or different applications that are invoked by the user (manually
selecting a Push-To-Talk application from a wireless phone vs. a
telephony application). Outside of a call, the difference in user
experience can be a billing one (cheaper for one service than
another), a notification feature for one and not another (for
example, an IM that gets sent whenever a user makes a call), and
so on.
In some cases, there is very little difference in the underlying
technology that will support two different services, and in other
cases, there are big differences. However, for the purposes of this
discussion, the key definition is that two services are distinct when
there is a perceived difference by the user in the two services.
This leads naturally to the desire to perform service identification.
Service identification is defined as the process of:
1. determining the underlying service that is driving a particular
signaling exchange,
2. associating that service with a service identifier, and
3. attaching that moniker to a signaling message (typically a SIP
INVITE).
Once service identification is performed, the service identifier can
then be used for various purposes within the network. Service
identification can be done in the endpoints, in which case the UA
would insert the moniker directly into the signaling message based on
its awareness of the service. Or, it can be done within a server in
the network (such as a proxy), based on inspection of the SIP
message, or based on hints placed into the message by the user.
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When service identification is performed entirely by inspecting the
signaling, this is called derived service identification. When it is
done based on knowledge possessed only by the invoking user agent, it
is called declarative service identification. Declarative service
identification can only be done in user agents, by definition.
3. Example Services
It is very useful to consider several example services, especially
ones that appear difficult to differentiate from each other. In
cases where it is hard to differentiate, service identification --
and in particular, declarative service identification -- appears
highly attractive (and indeed, required).
3.1. IPTV vs. Multimedia
IP Television (IPTV) is the usage of IP networks to access
traditional television content, such as movies and shows. SIP can be
utilized to establish a session to a media server in a network, which
then serves up multimedia content and streams it as an audio and
video stream towards the client. Whether SIP is ideal for IPTV is,
in itself, a good question. However, such a discussion is outside
the scope of this document.
Consider multimedia conferencing. The user accesses a voice and
video conference at a conference server. The user might join in
listen-only mode, in which case the user receives audio and video
streams, but does not send.
These two services -- IPTV and listen-only multimedia conferencing --
clearly appear as different services. They have different user
experiences and applications. A user is unlikely to ever be confused
about whether a session is IPTV or listen-only multimedia
conferencing. Indeed, they are likely to have different software
applications or endpoints for the two services.
However, these two services look remarkably alike based on the
signaling. Both utilize audio and video. Both could utilize the
same codecs. Both are unidirectional streams (from a server in the
network to the client). Thus, it would appear on the surface that
there is no way to differentiate them, based on inspection of the
signaling alone.
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3.2. Gaming vs. Voice Chat
Consider an interactive game, played between two users from their
mobile devices. The game involves the users sending each other game
moves, using a messaging channel, in addition to voice. In another
service, users have a voice and IM chat conversation using a buddy-
list application on their PC.
In both services, there are two media streams -- audio and messaging.
The audio uses the same codecs. Both use the Message Session Relay
Protocol (MSRP) [RFC4975]. In both cases, the caller would send an
INVITE to the Address of Record (AOR) of the target user. However,
these represent fairly different services, in terms of user
experience.
3.3. Gaming vs. Voice Chat #2
Consider a variation on the example in Section 3.2. In this
variation, two users are playing an interactive game between their
phones. However, the game itself is set up and controlled using a
proprietary mechanism -- not using SIP at all. However, the client
application allows the user to chat with their opponent. The chat
session is a simple voice session set up between the players.
Compare this with a basic telephone call between the two users. Both
involve a single audio session. Both use the same codecs. They
appear to be identical. However, different user experiences are
needed. For example, we desire traditional telephony features (such
as call forwarding and call screening) to be applied in the telephone
service, but not in the gaming chat service.
3.4. Configuration vs. Pager Messaging
The SIP MESSAGE method [RFC3428] provides a way to send one-shot
messages to a particular AOR. This specification is primarily aimed
at Short Message Service (SMS)-style messaging, commonly found in
wireless phones. Receipt of a MESSAGE request would cause the
messaging application on a phone to launch, allowing the user to
browse the message history and respond.
However, a MESSAGE request is sometimes used for the delivery of
content to a device for other purposes. For example, some providers
use it to deliver configuration updates, such as new phone settings
or parameters, or to indicate that a new version of firmware is
available. Though not designed for this purpose, the MESSAGE method
gets used since, in existing wireless networks, SMS is used for this
purpose, and the MESSAGE request is the SIP equivalent of SMS.
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Consequently, the MESSAGE request sent to a phone can be for two
different services. One would require invocation of a messaging app,
whereas the other would be consumed by the software in the phone,
without any user interaction at all.
4. Using Service Identification
It is important to understand what the service identity would be
utilized for, if known. This section discusses the primary uses.
These are application invocation in user agents and the network,
Quality of Service authorization, service authorization, accounting
and billing, service negotiation, and device dispatch.
4.1. Application Invocation in the User Agent
In some of the examples above, there were multiple software
applications executing on the host. One common way of achieving this
is to utilize a common SIP user agent implementation that listens for
requests on a single port. When an incoming INVITE or MESSAGE
arrives, it must be delivered to the appropriate application
software. When each service is bound to a distinct software
application, it would seem that the service identity is needed to
dispatch the message to the appropriate piece of software. This is
shown in Figure 1.
+---------------------------------+
| |
| +-------------+ +-------------+ |
| | UI | | UI | |
| +-------------+ +-------------+ |
| +-------------+ +-------------+ |
| | | | | |
| | Service 1 | | Service 2 | |
| | | | | |
| +-------------+ +-------------+ |
| +-----------------------------+ |
| | | |
| | SIP | |
| | Layer | |
| | | |
| +-----------------------------+ |
| |
+---------------------------------+
Physical Device
Figure 1
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The role of the SIP layer is to parse incoming messages, handle the
SIP state machinery for transactions and dialogs, and then dispatch
requests to the appropriate service. This software architecture is
analogous to the way web servers frequently work. An HTTP server
listens on port 80 for requests, and based on the HTTP Request-URI,
dispatches the request to a number of disparate applications. The
same is happening here. For the example services in Section 3.2, an
incoming INVITE for the gaming service would be delivered to the
gaming application software. An incoming INVITE for the voice chat
service would be delivered to the voice chat application software.
The example in Section 3.3 is similar. For the examples in
Section 3.4, a MESSAGE request for user-to-user messaging would be
delivered to the messaging or SMS app, and a MESSAGE request
containing configuration data would be delivered to a configuration
update application.
Unlike the web, however, in all three use cases, the user initiating
communications has (or appears to have -- more below) only a single
identifier for the recipient -- their AOR. Consequently, the SIP
Request-URI cannot be used for dispatching, as it is identical in all
three cases.
4.2. Application Invocation in the Network
Another usage of a service identifier would be to cause servers in
the SIP network to provide additional processing, based on the
service. For example, an INVITE issued by a user agent for IPTV
would pass through a server that does some kind of content rights
management, authorizing whether the user is allowed to access that
content. On the other hand, an INVITE issued by a user for
multimedia conferencing would pass through a server providing
"traditional" telephony features, such as outbound call screening and
call recording. It would make no sense for the INVITE associated
with IPTV to have outbound call screening and call recording applied,
and it would make no sense for the multimedia conferencing INVITE to
be processed by the content rights management server. Indeed, in
these cases, it's not just an efficiency issue (invoking servers when
not needed), but rather, truly incorrect behavior can occur. For
example, if an outbound call screening application is set to block
outbound calls to everything except for the phone numbers of friends
and family, an IPTV request that gets processed by such a server
would be blocked (as it's not targeted to the AOR of a friend or
family member). This would block a user's attempt to access IPTV
services, when that was not the goal at all.
Similarly, a MESSAGE request as described in Section 3.4 might need
to pass through a message server for filtering when it is associated
with chat, but not when it is associated with a configuration update.
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Consider a filter that gets applied to MESSAGE requests, and that
filter runs in a server in the network. The filter operation
prevents user Joe from sending messages to user Bob that contain the
words "stock" or "purchase", due to some regulations that disallow
Joe and Bob from discussing stock trading. However, a MESSAGE for
configuration purposes might contain an XML document that uses the
token "stock" as some kind of attribute. This configuration update
would be discarded by the filtering server, when it should not have
been.
4.3. Network Quality-of-Service Authorization
The IP network can provide differing levels of Quality of Service
(QoS) to IP packets. This service can include guaranteed throughput,
latency, or loss characteristics. Typically, the user agent will
make some kind of QoS request, either using explicit signaling
protocols (such as the Resource ReSerVation Protocol (RSVP)
[RFC2205]) or through marking of a Diffserv value in packets. The
network will need to make a policy decision based on whether or not
these QoS treatments are authorized. One common authorization policy
is to check if the user has invoked a service using SIP that they are
authorized to invoke, and that this service requires the level of QoS
treatment the user has requested.
For example, consider IPTV and multimedia conferencing as described
in Section 3.1. IPTV is a non-real-time service. Consequently,
media traffic for IPTV would be authorized for bandwidth guarantees,
but not for latency or loss guarantees. On the other hand,
multimedia conferencing is in real time. Its traffic would require
bandwidth, loss, and latency guarantees from the network.
Consequently, if a user should make an RSVP reservation for a media
stream, and ask for latency guarantees for that stream, the network
would choose to be able to authorize it if the service was multimedia
conferencing, but not if it was IPTV. This would require the server
performing the QoS authorization to know the service associated with
the INVITE that set up the session.
4.4. Service Authorization
Frequently, a network administrator will want to authorize whether a
user is allowed to invoke a particular service. Not all users will
be authorized to use all services that are provided. For example, a
user may not be authorized to access IPTV services, whereas they are
authorized to utilize multimedia processing. A user might not be
able to utilize a multiplayer gaming service, whereas they are
authorized to utilize voice chat services.
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Consequently, when an INVITE arrives at a server in the network, the
server will need to determine what the requested service is, so that
the server can make an authorization decision.
4.5. Accounting and Billing
Service authorization and accounting/billing go hand in hand. One of
the primary reasons for authorizing that a user can utilize a service
is that they are being billed differently based on the type of
service. Consequently, one of the goals of a service identity is to
be able to include it in accounting records, so that the appropriate
billing model can be applied.
For example, in the case of IPTV, a service provider can bill based
on the content (US $5 per movie, perhaps), whereas for multimedia
conferencing, they can bill by the minute. This requires the
accounting streams to indicate which service was invoked for the
particular session.
4.6. Negotiation of Service
In some cases, when the caller initiates a session, they don't
actually know which service will be utilized. Rather, they might
choose to offer up all of the services they have available to the
called party, and then let the called party decide, or let the system
make a decision based on overlapping service capabilities.
As an example, a user can do both the game and the voice chat service
described in Section 3.2. The user initiates a session to a target
AOR, but the devices used by the target can only support voice chat.
The called device returns, in its call acceptance, an indication that
only voice chat can be used. Consequently, voice chat gets utilized
for the session.
4.7. Dispatch to Devices
When a user has multiple devices, each with varying capabilities in
terms of service, it is useful to dispatch an incoming request to the
right device based on whether the device can support the service that
has been requested.
For example, if a user initiates a gaming session with voice chat,
and the target user has two devices -- one that can support the
gaming service, and another that cannot -- the INVITE should be
dispatched to the device that supports the gaming session.
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5. Key Principles of Service Identification
In this section, we describe several key principles of service
identification:
1. Services are a by-product of signaling
2. Identical signaling produces identical services
3. Declarative service identification is an example of "Do What I
Mean" (DWIM)
4. Declarative service identifiers are redundant
5. URIs are a key mechanism for producing differentiated signaling
5.1. Services Are a By-Product of Signaling
Declarative service identification -- the addition of a service
identifier by clients in order to inform other entities of what the
service is -- is a very compelling solution to solving the use cases
described above. It provides a clear way for each of the use cases
to be differentiated. On the other hand, derived service
identification appears "hard", since the signaling appears to be the
same for these different services.
Declarative service identification misses a key point, which cannot
be stressed enough, and which represents the core architectural
principle to be understood here:
A service is the byproduct of the signaling and the context around
it (the user profile, time of day, and so on) -- the effects of
the signaling message once it is launched into the network. The
service identity is therefore always derivable from the signaling
and its context without additional identifiers. In other words,
derived service identification is always possible when signaling
is being properly handled.
When a user sends an INVITE request to the network and targets that
request at an IPTV server, and includes the Session Description
Protocol (SDP) for audio and video streaming, the *result* of sending
such an INVITE is that an IPTV session occurs. The entire purpose of
the INVITE is to establish such a session, and therefore, invoke the
service. Thus, a service is not something that is different from the
rest of the signaling message. A service is what the user gets after
the network and other user agents have processed a signaling message.
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It may seem that delayed offers (SIP INVITE requests that lack SDP)
make it impossible to perform derived service identification. After
all, in some of the cases above, the differentiation was done using
the SDP in the request. What if it's not there? The answer is
simple -- if it's not there, and the SDP is being offered by the
called party, you cannot in fact know the service at the time of the
INVITE. That's the whole point of delayed offer -- to give the
called party the chance to offer up what it wants for the session.
In cases where service identification is needed at request time,
delayed offer cannot be used.
5.2. Identical Signaling Produces Identical Services
This principle is a natural conclusion of the previous assertion. If
a service is the byproduct of signaling, how can a user have
different experiences and different services when the signaling
message is the same? They cannot.
But how can that be? From the examples in Section 3, it would seem
that there are services that are different, but have identical
signaling. If we hold true to the assertion, there is in fact only
one logical conclusion:
If two services are different, but their signaling appears to be
the same, it is because one or more of the following is true:
1. there is in fact something different that has been overlooked
2. something has been implied from the signaling, when in fact it
should have been signaled explicitly
3. the signaling mechanism should be changed so that there is, in
fact, something that is different
To illustrate this, let us take each of the example services in
Section 3 and investigate whether there is, or should be, something
different in the signaling in each case.
IPTV vs. Multimedia Conferencing: The two services described in
Section 3.1 appear to have identical signaling. They both involve
audio and video streams, both of which are unidirectional. Both
might utilize the same codecs. However, there is another
important difference in the signaling -- the target URI. In the
case of IPTV, the request is targeted at a media server or to a
particular piece of content to be viewed. In the case of
multimedia conferencing, the target is a conference server. The
administrator of the domain can therefore examine the Request-URI
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and figure out whether it is targeted for a conference server or a
content server, and use that to derive the service associated with
the request.
Gaming vs. Voice Chat: Though both sessions involve MSRP and voice,
and both are targeted to the same AOR of the called user, there is
a difference. The MSRP messages for the gaming session carry
content that is game specific, whereas the MSRP messages for the
voice chat are just regular text, meant for rendering to a user.
Thus, the MSRP session in the SDP will indicate the specific
content type that MSRP is carrying, and this type will differ in
both cases. Even if the game moves look like text, since they are
being consumed by an automata, there is an underlying schema that
dictates their content, and therefore, this schema represents the
actual content type that should be signaled.
Gaming vs. Voice Chat #2: In this case, both sessions involve only
voice, and both are targeted at the same AOR. Indeed, there truly
is nothing different -- if indeed the signaling works this way.
However, there is an alternative mechanism for performing the
signaling. For the gaming session, the proprietary protocol can
be used to exchange a URI that can be used to identify the voice
chat function on the phone that is associated with the game (for
example, a Globally Routable User Agent URI (GRUU) can be used
[RFC5627]). Indeed, the gaming chat is not targeting the USER --
it's targeting the gaming instance on the phone. Thus, if a
special GRUU is used for the gaming chat, this makes the signaling
different between these two services.
Configuration vs. Pager Messaging: Just as in the case of gaming vs.
voice chat, the content type of the messages differentiates the
service that occurs as a consequence of the messages.
5.3. Do What I Say, Not What I Mean
"Do What I Mean", abbreviated as DWIM, is a concept in computer
science. It is sometimes used to describe a function that tries to
intelligently guess at what the user intended. It is in contrast to
"Do What I Say", or DWIS, which describes a function that behaves
concretely based on the inputs provided. Systems built on the DWIM
concept can have unexpected behaviors, because they are driven by
unstated rules.
Declarative service identification is an example of DWIM. The
service identifier has no well-defined impact on the state machinery
or protocols in the system; it has various side effects based on an
assumption of what is meant by the service identifier. Derived
service identification, on the other hand, is an expression of the
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principle of DWIS -- the behavior of the system is based entirely on
the specifics of the protocol and are well defined by the protocol
specification. The service identifier is just a shorthand for
summarizing things that are well defined by signaling.
As a litmus test to differentiate the two cases, consider the
following question. If a request contained a service identifier, and
that request were processed by a domain that didn't understand the
concept of service identifiers at all, would the request be rejected
if that service were not supported, or would it complete but do the
wrong thing? If it is the latter case, it's DWIM. If it's the
former, it's DWIS.
5.4. Declarative Service Identifiers Are Redundant
Because a declarative service identifier is, by definition, inside of
the signaling message, and because the signaling itself completely
defines the behavior of the service, another natural conclusion is
that a declarative service identifier is redundant with the signaling
itself. It says nothing that could not or should not otherwise be
derived from examination of the signaling.
5.5. URIs Are Key for Differentiated Signaling
In the IPTV example and in the second gaming example, it was
ultimately the Request-URI that was (or should be) different between
the two services. This is important. In many cases where services
appear the same, it is because the resource that is being targeted is
not, in fact, the user. Rather, it is a resource that is linked with
the user. This resource might be an instance of a software
application on the particular device of a user, or a resource in the
network that acts on behalf of the user.
The Request-URI is an infinitely large namespace for identifying
these resources. It is an ideal mechanism for providing
differentiation when there would otherwise be none.
Returning again to the example in Section 3.3, we can see that it
does make more sense to target the gaming chat session at a software
instance on the user's phone, rather than at the user themselves.
The gaming chat session should really only go to the phone on which
the user is playing the game. The software instance does indeed live
only on that phone, whereas the user themselves can be contacted in
many ways. We don't want telephony features invoked for the gaming
chat session, because those features only make sense when someone is
trying to communicate with the USER. When someone is trying to
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communicate with a software instance that acts on behalf of the user,
a different set of rules apply, since the target of the request is
completely different.
6. Perils of Declarative Service Identification
Based on these principles, several perils of declarative service
identification can be described. They are:
1. Declarative service identification can be used for fraud
2. Declarative service identification can hurt interoperability
3. Declarative service identification can stifle service innovation
6.1. Fraud
Declarative service identification can lead to fraud. If a provider
uses the service identifier for billing and accounting purposes, or
for authorization purposes, it opens an avenue for attack. The user
can construct the signaling message so that its actual effect (which
is the service the user will receive), is what the user desires, but
the user places a service identifier into the request (which is what
is used for billing and authorization) that identifies a cheaper
service, or one that the user is not authorized to receive. In such
a case, the user will receive service, and not be billed properly for
it.
If, however, the domain administrator derived the service identifier
from the signaling itself (derived service identification), the user
cannot lie. If they did lie, they wouldn't get the desired service.
Consider the example of IPTV vs. multimedia conferencing. If
multimedia conferencing is cheaper, the user could send an INVITE for
an IPTV session, but include a service identifier that indicates
multimedia conferencing. The user gets the service associated with
IPTV, but at the cost of multimedia conferencing.
This same principle shows up in other places -- for example, in the
identification of an emergency services call [ECRIT-FRAMEWORK]. It
is desirable to give emergency services calls special treatment, such
as being free and authorized even when the user cannot otherwise make
calls, and to give them priority. If emergency calls were indicated
through something other than the target of the call being an
emergency services URN [RFC5031], it would open an avenue for fraud.
The user could place any desired URI in the request-URI, and indicate
separately, through a declarative identifier, that the call is an
emergency services call. This would then get special treatment but
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of course would get routed to the target URI. The only way to
prevent this fraud is to consider an emergency call as any call whose
target is an emergency services URN. Thus, the service
identification here is based on the target of the request. When the
target is an emergency services URN, the request can get special
treatment. The user cannot lie, since there is no way to separately
indicate that this is an emergency call, besides targeting it to an
emergency URN.
6.2. Systematic Interoperability Failures
How can declarative service identification cause loss of
interoperability? When an identifier is used to drive functionality
-- such as dispatch on the phones, in the network, or QoS
authorization -- it means that the wrong thing can happen when this
field is not set properly. Consider a user in domain 1, calling a
user in domain 2. Domain 1 provides the user with a service they
call "voice chat", which utilizes voice and IM for real-time
conversation, driven off of a buddy-list application on a PC.
Domain 2 provides their users with a service they call "text
telephony", which is a voice service on a wireless device that also
allows the user to send text messages. Consider the case where
domain 1 and domain 2 both have their user agents insert a service
identifier into the request, and then use that to perform QoS
authorization, accounting, and invocation of applications in the
network and in the device. The user in domain 1 calls the user in
domain 2, and inserts the identifier "Voice Chat" into the INVITE.
When this arrives at the server in domain 2, the service identifier
is unknown. Consequently, the request does not get the proper QoS
treatment, even if the call itself will succeed.
If, on the other hand, derived service identification were used, the
service identifier could be removed by domain 2, and then recomputed
based on the signaling to match its own notion of services. In this
case, domain 2 could derive the "text telephony" identifier, and the
request completes successfully.
Declarative service identification, used between domains, causes
interoperability failures unless all interconnected domains agree on
exactly the same set of services and how to name them. Of course,
lack of service identifiers does not guarantee service
interoperability. However, SIP was built with rich tools for
negotiation of capabilities at a finely granular level. One user
agent can make a call using audio and video, but if the receiving UA
only supports audio, SIP allows both sides to negotiate down to the
lowest common denominator. Thus, communication is still provided.
As another example, if one agent initiates a Push-To-Talk session
(which is audio with a companion floor control mechanism), and the
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other side only did regular audio, SIP would be able to negotiate
back down to a regular voice call. As another example, if a calling
user agent is running a high-definition video conferencing endpoint,
and the called user agent supports just a regular video endpoint, the
codecs themselves can negotiate downward to a lower rate, picture
size, and so on. Thus, interoperability is achieved. Interestingly,
the final "service" may no longer be well characterized by the
service identifier that would have been placed in the original
INVITE. For example, in this case, if the original INVITE from the
caller had contained the service identifier "hi-fi video", but the
video gets negotiated down to a lower rate and picture size, the
service identifier is no longer really appropriate. That is why
services need to be derived by signaling -- because the signaling
itself provides negotiation and interoperability between different
domains.
This illustrates another key aspect of the interoperability problem.
Declarative service identification will result in inconsistencies
between its service identifiers and the results of any SIP
negotiation that might otherwise be applied in the session.
When a service identifier becomes something that both proxies and the
user agent need to understand in order to properly treat a request
(which is the case for declarative service identification), it
becomes equivalent to including a token in the Proxy-Require and
Require header fields of every single SIP request. The very reason
that [RFC4485] frowns upon usage of Require and certainly Proxy-
Require is the huge impact on interoperability it causes. It is for
this same reason that declarative service identification needs to be
avoided.
6.3. Stifling of Service Innovation
The probability that any two service providers end up with the same
set of services, and give those services the same names, becomes
smaller and smaller as the number of providers grow. Indeed, it
would almost certainly require a centralized authority to identify
what the services are, how they work, and what they are named. This,
in turn, leads to a requirement for complete homogeneity in order to
facilitate interconnection. Two providers cannot usefully
interconnect unless they agree on the set of services they are
offering to their customers and each do the same thing. This is
because each provider has become dependent on inclusion of the proper
service identifier in the request, in order for the overall treatment
of the request to proceed correctly. This is, in a very real sense,
anathema to the entire notion of SIP, which is built on the idea that
heterogeneous domains can interconnect and still get
interoperability.
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Declarative service identification leads to a requirement for
homogeneity in service definitions across providers that
interconnect, ruining the very service heterogeneity that SIP was
meant to bring.
Indeed, Metcalfe's Law says that the value of a network grows with
the square of the number of participants. As a consequence of this,
once a bunch of large domains did get together, agree on a set of
services, and then agree on a set of well-known identifiers for those
services, it would force other providers to also deploy the same
services, in order to obtain the value that interconnection brings.
This, in turn, will stifle innovation, and quickly force the set of
services in SIP to become fixed and never expand beyond the ones
initially agreed upon. This, too, is anathema to the very framework
on which SIP is built, and defeats much of the purpose of why
providers have chosen to deploy SIP in their own networks.
Consider the following example. Several providers get together and
standardize on a bunch of service identifiers. One of these uses
audio and video (say, "multimedia conversation"). This service is
successful and is widely utilized. Endpoints look for this
identifier to dispatch calls to the right software applications, and
the network looks for it to invoke features, perform accounting, and
provide QoS. A new provider gets the idea for a new service (say,
"avatar-enhanced multimedia conversation"). In this service, there
is audio and video, but there is a third stream, which renders an
avatar. A caller can press buttons on their phone, to cause the
avatar on the other person's device to show emotion, make noise, and
so on. This is similar to the way emoticons are used today in IM.
This service is enabled by adding a third media stream (and
consequently, a third m-line) to the SDP.
Normally, this service would be backwards-compatible with a regular
audio-video endpoint, which would just reject the third media stream.
However, because a large network has been deployed that is expecting
to see the token, "multimedia conversation" and its associated audio+
video service, it is nearly impossible for the new provider to roll
out this new service. If they did, it would fail completely, or
partially fail, when their users call users in other provider
domains.
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7. Recommendations
From these principles, several recommendations can be made.
7.1. Use Derived Service Identification
Derived service identification -- where an identifier for a service
is obtained by inspection of the signaling and of other contextual
data (such as subscriber profile) -- is reasonable, and when done
properly, does not lead to the perils described above. However,
declarative service identification -- where user agents indicate what
the service is, separate from the rest of the signaling -- leads to
the perils described above.
If it appears that the signaling currently defined in standards is
not sufficient to identify the service, it may be due to lack of
sufficient signaling to convey what is needed, or may be because
request URIs should be used for differentiation and they are not
being used. By applying the litmus tests described in Section 5.3,
network designers can determine whether or not the system is
attempting to perform declarative service identification.
7.2. Design for SIP's Negotiative Expressiveness
One of SIP's key strengths is its ability to negotiate a common view
of a session between participants. This means that the service that
is ultimately received can vary wildly, depending on the types of
endpoints in the call and their capabilities. Indeed, this fact
becomes even more evident when calls are set up between domains.
As such, when performing derived service identification, domains
should be aware that sessions may arrive from different networks and
different endpoints. Consequently, the service identification
algorithm must be complete -- meaning it computes the best answer for
any possible signaling message that might be received and any session
that might be set up.
In a homogeneous environment, the process of service identification
is easy. The service provider will know the set of services they are
providing, and based on the specific call flows for each specific
service, can construct rules to differentiate one service from
another. However, when different providers interconnect, or when
different endpoints are introduced, assumptions about what services
are used, and how they are signaled, no longer apply. To provide the
best user experience possible, a provider doing service
identification needs to perform a "best-match" operation, such that
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any legal SIP signaling -- not just the specific call flows running
within their own network amongst a limited set of endpoints -- is
mapped to the appropriate service.
7.3. Presence
Presence can help a great deal with providing unique URIs for
different services. When a user wishes to contact another user, and
knows only the AOR for the target (which is usually the case), the
user can fetch the presence document for the target. That document,
in turn, can contain numerous service URIs for contacting the target
with different services. Those URIs can then be used in the Request-
URI for differentiation. When possible, this is the best solution to
the problem.
7.4. Intra-Domain
Service identifiers themselves are not bad; derived service
identification allows each domain to cache the results of the service
identification process for usage by another network element within
the same domain. However, service identifiers are fundamentally
useful within a particular domain, and any such header must be
stripped at a network boundary. Consequently, the process of service
identification and their associated service identifiers is always an
intra-domain operation.
7.5. Device Dispatch
Device dispatch should be done following the principles of [RFC3841],
using implicit preferences based on the signaling. For example,
[RFC5688] defines a new UA capability that can be used to dispatch
requests based on different types of application media streams.
However, it is a mistake to try and use a service identifier as a UA
capability. Consider a service called "multimedia telephony", which
adds video to the existing PSTN experience. A user has two devices,
one of which is used for multimedia telephony and the other strictly
for a voice-assisted game. It is tempting to have the telephony
device include a UA capability [RFC3840] called "multimedia
telephony" in its registration. A calling multimedia telephony
device can then include the Accept-Contact header field [RFC3841]
containing this feature tag. The proxy serving the called party,
applying the basic algorithms of [RFC3841], will correctly route the
call to the terminating device.
However, if the calling party is not within the same domain, and the
calling domain does not know about or use this feature tag, there
will be no Accept-Contact header field, even if the calling party was
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using a service that is a good match for "multimedia telephony". In
such a case, the call may be delivered to both devices, but it will
yield a poorer user experience. That's because device dispatch was
done using declarative service identification.
The best way to avoid this problem is to use feature tags that can be
matched to well-defined signaling features -- media types, required
SIP extensions, and so on. In particular, the golden rule is that
the granularity of feature tags must be equivalent to the granularity
of individual features that can be signaled in SIP.
8. Security Considerations
Oftentimes, the service associated with a request is utilized for
purposes such as authorization, accounting, and billing. When
service identification is not done properly, the possibility of
unauthorized service use and network fraud is introduced. It is for
this reason, discussed extensively in Section 6.1, that the usage of
declarative service identifiers inserted by a UA is not recommended.
9. Acknowledgements
This document is based on discussions with Paul Kyzivat and
Andrew Allen, who contributed significantly to the ideas here. Much
of the content in this document is a result of discussions amongst
participants in the SIPPING mailing list, including Dean Willis,
Tom Taylor, Eric Burger, Dale Worley, Christer Holmberg, and
John Elwell, amongst many others. Thanks to Spencer Dawkins,
Tolga Asveren, Mahesh Anjanappa, and Claudio Allochio for reviews of
this document.
10. Informative References
[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.
[RFC4479] Rosenberg, J., "A Data Model for Presence", RFC 4479,
July 2006.
[RFC4485] Rosenberg, J. and H. Schulzrinne, "Guidelines for Authors
of Extensions to the Session Initiation Protocol (SIP)",
RFC 4485, May 2006.
[RFC4975] Campbell, B., Mahy, R., and C. Jennings, "The Message
Session Relay Protocol (MSRP)", RFC 4975, September 2007.
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[RFC5031] Schulzrinne, H., "A Uniform Resource Name (URN) for
Emergency and Other Well-Known Services", RFC 5031,
January 2008.
[ECRIT-FRAMEWORK]
Rosen, B., Schulzrinne, H., Polk, J., and A. Newton,
"Framework for Emergency Calling using Internet
Multimedia", Work in Progress, July 2009.
[RFC5627] Rosenberg, J., "Obtaining and Using Globally Routable User
Agent URIs (GRUUs) in the Session Initiation Protocol
(SIP)", RFC 5627, October 2009.
[RFC5688] Rosenberg, J., "A Session Initiation Protocol (SIP) Media
Feature Tag for MIME Application Subtypes", RFC 5688,
January 2010.
[RFC3428] Campbell, B., Rosenberg, J., Schulzrinne, H., Huitema, C.,
and D. Gurle, "Session Initiation Protocol (SIP) Extension
for Instant Messaging", RFC 3428, December 2002.
[RFC3841] Rosenberg, J., Schulzrinne, H., and P. Kyzivat, "Caller
Preferences for the Session Initiation Protocol (SIP)",
RFC 3841, August 2004.
[RFC3840] Rosenberg, J., Schulzrinne, H., and P. Kyzivat,
"Indicating User Agent Capabilities in the Session
Initiation Protocol (SIP)", RFC 3840, August 2004.
[RFC2205] Braden, B., Zhang, L., Berson, S., Herzog, S., and S.
Jamin, "Resource ReSerVation Protocol (RSVP) -- Version 1
Functional Specification", RFC 2205, September 1997.
Author's Address
Jonathan Rosenberg
jdrosen.net
Monmouth, NJ
USA
EMail: jdrosen@jdrosen.net
URI: http://www.jdrosen.net
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