Simple SIP Usage Scenario for Applications in the Endpoints
RFC 5638
| Document | Type |
RFC - Informational
(September 2009; No errata)
Was draft-sinnreich-sip-tools (individual in rai area)
|
|
|---|---|---|---|
| Authors | Kundan Singh , Eunsoo Shim , Alan Johnston , Henry Sinnreich | ||
| Last updated | 2018-12-20 | ||
| 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 5638 (Informational) | |
| Action Holders |
(None)
|
||
| Consensus Boilerplate | Unknown | ||
| Telechat date | |||
| Responsible AD | Cullen Jennings | ||
| Send notices to | mary.barnes@nortel.com, kundan@adobe.com | ||
Network Working Group H. Sinnreich, Ed.
Request for Comments: 5638 Adobe
Category: Informational A. Johnston
E. Shim
Avaya
K. Singh
Columbia University Alumni
September 2009
Simple SIP Usage Scenario for Applications in the Endpoints
Abstract
For Internet-centric usage, the number of SIP-required standards for
presence and IM and audio/video communications can be drastically
smaller than what has been published by using only the rendezvous and
session-initiation capabilities of SIP. The simplification is
achieved by avoiding the emulation of telephony and its model of the
intelligent network. 'Simple SIP' relies on powerful computing
endpoints. Simple SIP desktop applications can be combined with rich
Internet applications (RIAs). Significant telephony features may
also be implemented in the endpoints.
This approach for SIP reduces the number of SIP standards with which
to comply -- from roughly 100 currently, and still growing, to about
11.
References for NAT traversal and for security are also provided.
Status of This Memo
This memo provides information for the Internet community. It does
not specify an Internet standard of any kind. Distribution of this
memo is unlimited.
Copyright Notice
Copyright (c) 2009 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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RFC 5638 SIP Usage for Applications in Endpoints September 2009
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 BSD License.
Table of Contents
1. Introduction ....................................................3
2. The Endpoint in the SIP and Web Architectures ...................5
2.1. The Telephony Gateway as a SIP Endpoint ....................6
3. Applicability for Simple SIP in the Endpoints ...................7
3.1. What Simple SIP Can Accomplish .............................7
3.2. Baseline for Simple SIP ....................................7
3.3. What Simple SIP May or May Not Accomplish ..................8
3.4. What Is Out of Scope for Simple SIP ........................8
3.5. Borderline Cases ...........................................9
4. Mandatory SIP References for Internet-Centric Usage .............9
4.1. RFC 3261: "SIP: Session Initiation Protocol" ..............10
4.2. RFC 4566: "SDP: Session Description Protocol" .............10
4.3. RFC 3264: "An Offer/Answer Model with Session
Description Protocol (SDP)" ...............................10
4.4. RFC 3840: "Indicating User Agent Capabilities in
the Session Initiation Protocol (SIP)" ....................10
4.5. RFC 3263: "Session Initiation Protocol (SIP):
Locating SIP Servers" .....................................11
4.6. RFC 3265: "Session Initiation Protocol
(SIP)-Specific Event Notification" ........................11
4.7. RFC 3856: "A Presence Event Package for the
Session Initiation Protocol (SIP)" ........................11
4.8. RFC 3863: "Presence Information Data Format (PIDF)" .......11
4.9. RFC 3428: "Session Initiation Protocol (SIP)
Extension for Instant Messaging" ..........................12
4.10. RFC 4474: "Enhancements for Authenticated
Identity Management in the Session Initiation
Protocol (SIP)" ..........................................12
4.11. RFC 3581: "An Extension to the Session Initiation
Protocol (SIP) for Symmetric Response Routing" ...........12
4.12. Updates to SIP-Related Protocols .........................12
5. SIP Applications in the Endpoints ..............................12
6. NAT Traversal ..................................................14
7. Security Considerations ........................................14
8. Acknowledgements ...............................................15
9. References .....................................................16
9.1. Normative References ......................................16
9.2. Informative References ....................................17
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1. Introduction
The Session Initiation Protocol (SIP) has become the global standard
for real-time multimedia communications over the Internet and in
private IP networks, due to its adoption by service providers and in
enterprise networks alike. The cost of this success has been a
continuing increase in complexity to accommodate the various
requirements for such networks. At the same time, the World Wide Web
has become the platform for a boundless variety of rich Internet
applications (RIAs), both in the browser and on the desktop. For SIP
to be useful for RIAs, requirements for legacy voice-service
providers that add unnecessary complexity may be avoided by
delegating the interworking to telephony gateway endpoints. This
usage scenario for SIP requires following the end-to-end principle of
the Internet architecture at the application level or, in other
words, placing SIP applications in the endpoints.
There are several reasons, from the Web service's perspective, to
place most or all SIP applications in the endpoints and just use the
client-server (CS) or peer-to-peer (P2P) rendezvous function for SIP:
1. Value proposition: SIP applications in the endpoints can be easily
mixed with RIAs and thus enable service providers to offer new
services in a scalable and flexible manner. Mixing SIP
applications with RIAs also significantly enhances the value of
SIP applications. Rich Internet applications support unrestricted
user choice as an alternative that is beyond what is traditionally
prepackaged as network-based communication service plans.
2. Eliminating the problems associated with distributed SIP
applications in various feature servers across the network allows
us to greatly simplify SIP. There is also the Internet end-to-end
principle, which argues that network intermediaries cannot
completely understand the applications and their state in the
endpoints.
'Simple SIP' in this document refers the SIP functions necessary to
support only the rendezvous and session-setup functions of SIP,
voice, video, basic presence, instant messaging, and also security.
Simple SIP is focused on providing a basic multimedia, real-time
communications "call". This includes presence, instant messaging,
voice, and video for point-to-point and various conference
applications. One or a very small number of additional servers may
also be provided; for example, a voice-mail server may be provided as
an auxiliary to make a simple one-to-one call to voice mail if the
callee does not answer or to check voice mail.
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Once the applications in the endpoints have established basic
communications, it is up to them to support available features
selected by users. This paper is targeted to such scenarios. In
telephony, most of the value to users and service providers alike is
added by signaling. By contrast, on the Web, RIAs add most of the
value. The integrated use of SIP and RIAs in the endpoints can
combine the best of both.
This approach limits the number of SIP standards to roughly 11 that
are listed here as the core for simple SIP. At the time of this
writing, the Real-Time Applications and Infrastructure (RAI) area of
the IETF is focused on a dedicated working group for the core SIP
protocol, separate from various SIP applications. We anticipate this
emerging work will also be the core of what is termed here as simple
SIP and will actually further reduce the number of references that
reflect the present core SIP standards.
This memo aims to shield Web application developers from the need to
know or understand more than the core SIP protocol. The total number
of references has been kept to a minimum and includes other related
topics, such as examples for providing telephony services in the
endpoints, NAT traversal, and security. The referenced papers are,
however, entry points to these knowledge resources. Readers
interested in a more detailed list of SIP topics, especially
telephony, can follow up the short list here with the extensive list
in "A Hitchhikers' Guide to SIP", RFC 5411 [12]. The guide has over
140 references for understanding most, but not all, of the published
features of SIP in the IETF and elsewhere. There is also a Web site
that automatically tracks the number of SIP-related RFCs [13]. Other
standards and commercial organizations have greatly enlarged the
published features of SIP as well. We could not actually provide a
complete count on everything that has been published as some form of
SIP-standard document.
NAT traversal is also a basic requirement for simple SIP. However,
given the potential option of using the Host Identity Protocol (HIP)
in SIP-enabled endpoints, as shown in Section 4, simple SIP may not
require any standards other than those mentioned here. The
alternative to HIP is to use SIP-specific protocols for NAT
traversal, such as STUN (Simple Traversal of the UDP Protocol through
NAT), TURN (Traversal Using Relay NAT), and ICE (Interactive
Connectivity Establishment), as discussed in Section 4.
The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL
NOT","SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in
this document are to be interpreted as described in RFC 2119.
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2. The Endpoint in the SIP and Web Architectures
SIP has been defined in RFC 3261 for rendezvous and session
initiation. The usual example is the trapezoid model for
communications between two endpoints placed in two different SIP
service-provider domains. SIP is also flexible, since SIP
applications beyond the rendezvous function can reside either in the
SIP networks in additional feature and media servers or in the
endpoints. SIP endpoints are our focus in this memo.
Since SIP has been invented, with much initial similarity between SIP
and HTTP, the Web has evolved from a global access mechanism to
static documents to a universal platform with rich interaction
between the user and client. In most cases, the client is the
browser, though recently dedicated Web desktop clients have emerged
as well.
The Web provides access to applications as well as to documents. It
is beyond the scope of this memo to describe the application and
network architectures of the Web. We will note, however, some of the
new application and communication forms that have emerged on the Web
as a result of a Darwinian evolution [30] rather than as a result of
being defined in standards organizations. They are referred to as
Rich Internet Applications.
Examples of RIAs include social networks, blogs, wikis, web-based
office and collaboration tools, as well as task-related apps for
creating to-do lists, tracking time, combining geographic information
with various applications (such as tracking exercise paths and
recording the metrics), tracking airline flights, combining live
video from events with results and comments, etc.
More information can be found at [31] and in the vast collection of
books about RIAs.
RIAs have positioned the browser (and associated Web desktop
applications) as the dominant platform for a large variety of
applications. They are universal application platforms, independent
of network location, operating system, processor, or display size.
Behind the better-known Web applications are a wealth of new
technologies that can enhance SIP-based communications, for example,
the aggregation of data at runtime from several resources on the
Internet. A variety of RIA components, such as found on interactive
Web pages, can significantly improve the user experience of SIP-based
communications. This is in contrast to the fixed interfaces found in
most SIP user agents (UA), such as phones and desktop clients.
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The Web network and application architecture is very different from
SIP service-provider networks at present, but the one point where
they both meet is the end-user device of any shape: fixed or mobile.
The desire of SIP service providers to support new services in a
scalable and flexible manner is incidentally easier to implement by
the loose service coupling on the Web, as it is possible to
characterize a service, or actually a mix of several service
components (such as in a mash-up), with a URI. This is in contrast
to network services registration being done by a central registrar.
The Web architecture is also better suited for users to select and
configure their applications and interaction mode with the client.
The boundless variety of configurations of services and client
settings on the Web is in contrast with the prepackaged services and
fixed user-agent configurations in present SIP services.
Last but not least, program execution locally on the client is faster
if the interaction with servers across the network is minimized.
The motivation behind this memo is the potential of integrating SIP-
based multimedia communications with access to RIAs on the Web. To
mention a few scenarios: adding SIP- and RTP-based real-time
communications to RIAs, integrating (from a user perspective) the SIP
location service (not to be confused with geographic location
services) with other desktop- and network-based geographic location
services, using social networks as part of the contact list, etc.
2.1. The Telephony Gateway as a SIP Endpoint
In order to accomplish interoperability with the installed base of
telephone networks of various kinds, integrating SIP communications
into RIAs precludes, in our opinion, carrying legacy telephony
features over to the Web. Interoperability between the Internet and
telephone networks is best left to gateways that look to the Web as
special endpoints serving large numbers of users. Plain one-to-one
phone calls are already supported by Internet-to-telephony gateways.
If added, PSTN (Public Switched Telephone Network) or ISDN telephony
features must be exposed to Web users; visual Web display and
interaction with the user is preferable to carrying the extremely
complex SIP equivalents over into the Internet. On the Internet side
of telephony gateways, simple SIP is all that needs to be deployed,
in our opinion. Additional telephony features can be just another
RIA hosted in the gateway. The market is the best indicator to show
if such an effort is worthwhile to be productized.
Overloading simple SIP with telephony features is a non-objective, as
detailed in Section 3.
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3. Applicability for Simple SIP in the Endpoints
This section aims to clarify the scope of applicability by
considering what can be done better in the endpoints, what simple SIP
for user agents can and cannot accomplish, and what is out of scope.
We will use emergency calls as an example to illustrate these points
on applicability. Emergency calls are also a good example for
considering if and when SIP-plus-RIA applications could be used as
emergency telephony enhancements or even replacements.
3.1. What Simple SIP Can Accomplish
The main goal for SIP applications on the desktop or in the browser
is to support the integration of SIP- and RTP-based real-time
communications with RIAs. This assumes powerful endpoints, such as
PC/laptop, smart mobile phones, or various dedicated devices.
Example of better functionality: emergency calls not limited to a
Public Safety Access Point (PSAP), but extended to a medical service
taking care of patients or elderly people.
In this example, besides alerting the right medical provider of the
emergency, vital body-sign data and video can also be transmitted.
In the opposite direction, the caller may get visual and audio
information and instructions for instant self-help. In this
scenario, there is no need to invoke a PSAP service. A dedicated
device for such scenarios may actually have an emergency medical call
button, though for telephone calls to a PSAP this is not recommended
[14]. Powerful endpoints may also have various means to determine
the geographic location of the caller and transmit it to the
emergency care provider. In this and other examples, SIP voice may
be a component of several other communications means, but not always
the central one; some emergency communications and data transfer may
actually be performed without voice, such as instances when the
"caller" cannot speak for some reason.
3.2. Baseline for Simple SIP
The focus of the memo is to define the baseline for simple SIP: the
establishment of a one-to-one real-time multimedia communication
session for presence, IM, voice, and video. Adequate security must
also be provided; authentication and encryption for the media and for
parts of the signaling should be done in a manner consistent with the
routing of SIP messages.
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3.3. What Simple SIP May or May Not Accomplish
There are border cases where simple SIP may or may not accomplish
some necessary legacy function. Example: an emergency call to a PSAP
over the Internet may be supported using the SOS URN [15] and the
LoST protocol [16] to determine where to route the call. If,
however, emergency calls must be routed over the PSTN to a country-
specific telephone number, the assistance of a SIP proxy and also of
a SIP-PSTN gateway is required to recognize and route the emergency
call. Depending on the local jurisdiction, emergency calls from a
SIP UA may require other features that are beyond the scope of this
memo.
3.4. What Is Out of Scope for Simple SIP
The simple usage of SIP is applicable when avoiding the traditional
voice-provider approaches for charging (or monetizing) that aim to
provide, manage, and charge for what is referred to as services (not
applications); some examples of such approaches to charging are
listed here. Simple SIP means to avoid placing any functions in the
network other than the rendezvous function of SIP. This includes
avoiding:
o support of legacy telephony functions, such as emulating public-
telephone-switch services and voice-only private branch exchanges.
o SIP network architectures designed to support telephony-type
network models. Examples include long chains of SIP proxies and
feature servers (more than the two SIP servers shown in RFC 3261)
that may be encountered inside and between closed Voice over IP
(VoIP) networks and in-transit VoIP networks in between. Long
chains of intermediaries of any type not only add complexity, they
pose a security risk that increases with the number of SIP network
elements. Complex server-based networks also make it more
difficult to introduce new services. A special problem in SIP
server chains is forking, which leads to the well-known problems
of concurrency in computing; the so-called race conditions in
telephony. This is amplified by redesigning the whole network
every time there is a new SIP routing requirement.
o support for legacy telephony models, such as identifying end-user
devices for the purpose of differentiated charging by type of
service or for charging for roaming between networks.
o policies and the associated policy servers and network elements
for Quality of Service (QoS) to enforce service-rate-specific
policies for real-time communications.
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o design considerations for SIP for compatibility with legacy
telephony networks, traditional telephony services, and various
telephone numbering plans. This pushes the responsibility of
mapping the URI to telephone numbers to edge networks where the
IP-PSTN gateway functions are performed. The handling of
telephony-specific functions, such as early media, are also pushed
to edge gateway networks. Other design considerations for
interworking with the PSTN and 'looking like the PSTN' are also
avoided.
This list is not exhaustive, but conveys the concept of what to avoid
when using SIP as a simpler protocol to understand and to implement.
3.5. Borderline Cases
There are also some interesting borderline cases for what to avoid,
such as Provisional Response Acknowledgements (PRACKs), specified in
RFC 3262. PRACK is targeted for multi-hop SIP server networks and
PSTN interworking, especially to assure reliable early media. PRACK
can be delegated, albeit with some limitations to the SIP-PSTN
gateway. PRACK does little to improve the user experience and has no
relevance on true broadband networks with minimal SIP hop counts.
Using PRACK may therefore be a decision best left to designers.
Another interesting example of a borderline case are the issues with
SIP's Non-Invite transactions as discussed in RFC 4320 [17]. Long
chains of SIP intermediaries complicate the handling of provisional
responses and may create several problems, such as storms of late
responses from forked SIP forwarding paths. We mentioned that long
chains of SIP intermediaries are out of scope for simple SIP, but
since designers may encounter various scenarios, even those they
don't like, the decision to conform the user agent (UA) to RFC 4320
is best left to them.
The list of borderline cases is also not exhaustive and the above are
only examples. So where is the borderline? We believe that SIP usage
on the Internet, without any intermediaries designed to support
closed VoIP networks, eliminates the borderline cases. Enterprise
SIP networks are also most useful when designed to work with the
Internet model in mind, by giving enterprise users the benefit of
SIP-enhanced Web applications for productivity. Handling of SIP in
enterprise firewalls is out of the scope of this memo.
4. Mandatory SIP References for Internet-Centric Usage
Here is the minimal set of mandatory references to support the
Internet-centric approach to SIP, outlined above. The minimal set of
references defines simple SIP.
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The proposed change process [29] for SIP in the IETF RAI area will
define the updated SIP core specification and thus reduce even more
the required SIP standards for what is referred to here as simple
SIP.
4.1. RFC 3261: "SIP: Session Initiation Protocol"
RFC 3261 [1] is the core specification for SIP. The trapezoid model
for SIP, found in RFC 3261, is only an example and a use case
applicable to two service providers featuring an outgoing SIP proxy
and an incoming SIP proxy in each domain respectively. However, SIP
can also work in peer-to-peer (P2P) communications without SIP
servers.
4.2. RFC 4566: "SDP: Session Description Protocol"
SDP [2] is the standard format for the representation of media
parameters, transport addresses, and other session data irrespective
of the protocol used to transport the SDP data. SIP is one of the
protocols used to transport SDP data, to enable the setting up of
multimedia communication sessions. Other Internet application
protocols use SDP as well.
4.3. RFC 3264: "An Offer/Answer Model with Session Description Protocol
(SDP)"
Though SDP has the capability to describe SIP sessions, how to arrive
at a common description by two SIP endpoints requires a negotiation
procedure to agree on common media codecs, along with IP addresses
and ports where the media can be received. This negotiation
procedure is specified in RFC 3264 [3]. As will be seen in Section
6, this negotiation is usually considerably complicated due to the
existence of NAT between the SIP endpoints.
4.4. RFC 3840: "Indicating User Agent Capabilities in the Session
Initiation Protocol (SIP)"
A SIP UA can convey its capability in the Contact header field,
indicating if it can support presence, IM, audio, or video, and if
the device is fixed, mobile, or other, such as the endpoint being an
automaton (voice mail for example). Which SIP methods are supported
may also be indicated as specified in RFC 3840 [4]. SIP registrars
(SIP servers or the P2P SIP overlay) can be informed of endpoint
capabilities. Missing capabilities can be displayed for the user by,
for example, grayed out or missing icons.
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4.5. RFC 3263: "Session Initiation Protocol (SIP): Locating SIP
Servers"
RFC 3263 [5] adds key clarifications to the base SIP specification in
RFC 3261 by specifying how a SIP user agent (UA) or SIP server can
determine with DNS queries not only the IP addresses of the target
SIP servers, but also which SIP servers can support UDP or TCP
transport, as required. TCP may be required to support secure SIP
(SIPS) using Transport Layer Security (TLS) transport or when SIP
messages are too large to fit into UDP packets without fragmentation.
Successive DNS queries yield finer-grain location by providing NAPTR,
SRV, and A type records. Note that finding a SIP server requires
several successive DNS queries to access these records.
Locating SIP servers is also required for P2P SIP when a peer node
wishes to communicate with a SIP UA outside its own P2P SIP overlay
network.
4.6. RFC 3265: "Session Initiation Protocol (SIP)-Specific Event
Notification"
RFC 3265 [6] provides an extensible framework by which SIP nodes can
request notification from remote nodes indicating that certain events
have occurred. The most prominent event notifications are those used
for presence, though SIP events are used for many other SIP services,
some of which can be useful for simple SIP.
4.7. RFC 3856: "A Presence Event Package for the Session Initiation
Protocol (SIP)"
RFC 3856 [7] defines the usage of SIP as a presence protocol and
makes use of the SUBSCRIBE and NOTIFY methods for presence events.
SIP location services already contain presence information in the
form of registrations and, as such, can be reused to establish
connectivity for subscriptions and notifications. This can enable
either endpoints or servers to support rich applications based on
presence.
4.8. RFC 3863: "Presence Information Data Format (PIDF)"
RFC 3863 [8] defines the Presence Information Data Format (PIDF) and
the media type "application/pidf+xml" to represent the XML MIME
entity for PIDF. PIDF is used by SIP to carry presence information.
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4.9. RFC 3428: "Session Initiation Protocol (SIP) Extension for Instant
Messaging"
The SIP extension for IM in RFC 3428 [9] consists in the MESSAGE
method (defined in RFC 3428) only for the pager model of IM, based on
the assumption that an IM conversation state exists in the client
interface in the endpoints or in the mind of the users.
4.10. RFC 4474: "Enhancements for Authenticated Identity Management in
the Session Initiation Protocol (SIP)"
RFC 4474 [10] defines (1) an identity header and (2) an identity info
header for SIP requests that carry, respectively, the signature of
the issuer over parts of the SIP request and the signed identity
information. The signature includes the FROM header and the identity
of the sender. The associated identity info header identifies the
sender of the SIP request, such as INVITE. The issuer of the
signature can present their certificate as well. It is assumed the
issuer may be the domain owner. Strong authentication is thus
provided for SIP requests. Authentication for SIP responses is not
defined in this document.
4.11. RFC 3581: "An Extension to the Session Initiation Protocol (SIP)
for Symmetric Response Routing"
RFC 3581 [11] specifies an extension to SIP called "rport" so that
responses are sent back to the source IP address and port from which
the request originated. This correction to RFC 3261 is helpful for
NAT traversal, debugging, and support of multi-homed hosts.
4.12. Updates to SIP-Related Protocols
Several of the above are being updated to benefit from the experience
of large deployments and frequent interoperability testing. We
recommend readers to constantly check for revisions. One update
example is "Correct Transaction Handling for 200 Responses to the
Session Initiation Protocol INVITE Requests" [18]. This is an update
to RFC 3261; the added security risk for misbehaving SIP UAs is
handled in the forwarding SIP proxy.
5. SIP Applications in the Endpoints
Although the present adoption of SIP is mainly due to telephony
applications, its roots are in the Web and it has initial similarity
to HTTP. As a result, SIP may play other roles in adequately
powerful endpoints (their number keeps increasing with Moore's law).
SIP-based multimedia communications may be linked with various other
applications on the Web. Either some non-SIP application or the
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communication feature may be perceived as the primary usage. An
example is mixing SIP-based real-time communications with some Web
content of high interest to the user.
Examples:
1. In a conversation between a consumer and the contact center, a Web
conference can be invoked to present to the user buying options or
help information. This information can make use of mashups to
combine real-time data from various sources on the Web.
2. In a social network, multimedia conversations combined with Web
mashups can be invoked, thus strengthening the bond between its
members.
3. Conversations can be invoked while watching some events on the Web
in real time. However, the main beneficiary in this case may be
the Web site, since the conversation can prolong the time for
users watching that Web site.
This shows the value of combining RIAs with SIP-based communications.
It is a matter for the end user's judgment whether the Web content or
the associated communication capability is more important, or if a
mix of both is most attractive.
Example: a Web-based enterprise directory where employees can find a
wealth of data. Adding SIP multimedia communications to the
enterprise directory to call someone (if online and not too busy)
enhances its usefulness, but is not critical to the directory.
SIP applications in the endpoints can, however, accomplish most
telephony functions as well. This has been amply documented in SIP-
related work in the IETF, such as:
o "A Call Control and Multi-party usage framework for SIP" [19]
presents a large assortment of telephony applications where the
call control resides in the participating endpoints that use the
peer-to-peer feature invocation model. The peer-to-peer design
and its principles are based on multiparty call control.
o "Session Initiation Protocol Service Examples" [20] contains a
collection of SIP call flows for traditional telephony, many of
which require no server support for the respective features. The
SIP service examples for telephony are extremely useful since they
illustrate in detail the concepts and applications supported by
the core simple SIP references.
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In conclusion, SIP applications in the endpoints can support both a
mix of real-time communications with new rich Internet applications
and traditional telephony features as well.
6. NAT Traversal
SIP devices behind one or more NATs are, at present, the rule rather
than the exception.
"Best Current Practices for NAT Traversal for SIP" [22]
comprehensively summarizes the use of STUN, TURN, and ICE, and
provides a definitive set of 'Best Common Practices' to demonstrate
the traversal of SIP and its associated RTP media packets through NAT
devices.
The use of ICE has been developed mainly for SIP. Other proposals,
such as NICE (generic for non-SIP) and "D-ICE" for Real Time
Streaming Protocol (RTSP) streaming media, have also been proposed.
Internet games have different NAT traversal techniques of their own.
This list is not exhaustive and such approaches are based on
different NAT traversal protocols for each application protocol,
separately.
A general, non-application-protocol-specific approach for NAT
traversal is therefore highly desirable.
One approach for NAT traversal that is generic and applicable for all
application protocols is to deploy the Host Identity Protocol (HIP)
and solve NAT traversal only once, at the HIP level. HIP has many
other useful features (such as support for the IPv6 transition in
endpoints, mobility, and multihoming) that are beyond the scope of
this paper. "Basic HIP Extensions for Traversal of Network Address
Translators" [23] provides an extensive coverage of the use of HIP
for NAT traversal.
Using HIP-enabled endpoints can provide the functions required for
NAT traversal [24] for all applications, for both IPv4 and IPv6. HIP
can thus simplify the SIP UA since it takes away the burden of NAT
traversal from the SIP UA and moves it to the HIP protocol module in
the endpoint.
7. Security Considerations
All protocols discussed in this paper have their own specific
security requirements that MUST be considered. The special security
considerations for SIP signaling security and RTP media security are
discussed here.
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SIP security has two main parts: transport security and identity.
o Transport security for SIP is specified in RFC 3261. Secure SIP
has the notation SIPS in the request URI and uses TLS over TCP.
Note that SIP over UDP cannot be secured in this way. Transport
security works only hop by hop. Specifying SIPS requires the user
to trust all intermediate servers and no end-to-end media
encryption is assumed. There is no insurance for misbehaving
intermediaries in the path. SIPS is therefore really adequate
only in single-hop scenarios.
o RFC 4474, "Enhancements for Authenticated Identity Management in
the Session Initiation Protocol (SIP)", which is mentioned
previously, specifies the use of certificates for secure
identification of the parties involved in SIP signaling requests.
o The Datagram Transport Layer Security (DTLS) specified in RFC 4347
[25] has wide applicability for other applications that require
UDP transport. DTLS has been designed to have maximum commonality
with TLS, yet does not require TCP transport and works over UDP.
The DTLS-SRTP (Secure Realtime Transport Protocol) Framework [26]
can support encrypted communications between endpoints using
self-signed certificates whose fingerprints are exchanged over an
integrity-protected SIP signaling channel. The SRTP master secret
is derived using the DTLS exchange as described in [27].
o ZRTP [28] provides key agreement for SRTP for multimedia
communication with voice without depending on SIP signaling,
though it can utilize an integrity-protected SIP signaling path
for authentication. ZRTP does not require the use of certificates
or any Public Key Infrastructure (PKI). ZRTP provides best-effort
SRTP encryption without any additional SIP extensions.
8. Acknowledgements
The authors would like to thank Cullen Jennings, Ralph Droms, and
Adrian Farrel for helpful comments in the most recent stage of this
memo.
Special thanks are due to Paul Kyzivat for challenging the authors to
clarify the role of telephony network gateways and also to Keith
Drage for challenging them to discuss the use of emergency calls
using simple SIP.
Robert Sparks has pointed to some missing references, which we have
added.
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The authors would also like to thank Jiri Kuthan, Adrian Georgescu,
and others for the detailed discussion on the SIPPING WG list. As a
result, we have added clarification of what simple SIP can do, what
it does not aim to do, and some scenarios in between. We would also
like to thank Wilhelm Wimmreuter for the detailed review of the
initial draft and to Arjun Roychaudhury for the comments regarding
the need to clarify the difference between network-based services and
endpoint applications.
9. References
9.1. Normative References
[1] 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.
[2] Handley, M., Jacobson, V., and C. Perkins, "SDP: Session
Description Protocol", RFC 4566, July 2006.
[3] Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model with
Session Description Protocol (SDP)", RFC 3264, June 2002.
[4] Rosenberg, J., Schulzrinne, H., and P. Kyzivat, "Indicating
User Agent Capabilities in the Session Initiation Protocol
(SIP)", RFC 3840, August 2004.
[5] Rosenberg, J. and H. Schulzrinne, "Session Initiation Protocol
(SIP): Locating SIP Servers", RFC 3263, June 2002.
[6] Roach, A., "Session Initiation Protocol (SIP)-Specific Event
Notification", RFC 3265, June 2002.
[7] Rosenberg, J., "A Presence Event Package for the Session
Initiation Protocol (SIP)", RFC 3856, August 2004.
[8] Sugano, H., Fujimoto, S., Klyne, G., Bateman, A., Carr, W., and
J. Peterson, "Presence Information Data Format (PIDF)", RFC
3863, August 2004.
[9] Campbell, B., Ed., Rosenberg, J., Schulzrinne, H., Huitema, C.,
and D. Gurle, "Session Initiation Protocol (SIP) Extension for
Instant Messaging", RFC 3428, December 2002.
[10] Peterson, J. and C. Jennings, "Enhancements for Authenticated
Identity Management in the Session Initiation Protocol (SIP)",
RFC 4474, August 2006.
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[11] Rosenberg, J. and H. Schulzrinne, "An Extension to the Session
Initiation Protocol (SIP) for Symmetric Response Routing", RFC
3581, August 2003.
9.2. Informative References
[12] Rosenberg, J., "A Hitchhiker's Guide to the Session Initiation
Protocol (SIP)", RFC 5411, February 2009.
[13] Ohlmeier, N., "VoIP RFC Watch", http://rfc3261.net/.
[14] Rosen, B. and J. Polk, "Best Current Practice for
Communications Services in support of Emergency Calling", Work
in Progress, July 2009.
[15] Schulzrinne, H., "A Uniform Resource Name (URN) for Emergency
and Other Well-Known Services", RFC 5031, January 2008.
[16] Hardie, T., Newton, A., Schulzrinne, H., and H. Tschofenig,
"LoST: A Location-to-Service Translation Protocol", RFC 5222,
August 2008.
[17] Sparks, R., "Actions Addressing Identified Issues with the
Session Initiation Protocol's (SIP) Non-INVITE Transaction",
RFC 4320, January 2006.
[18] Sparks, R. and T. Zourzouvillys, "Correct Transaction Handling
for 200 Responses to Session Initiation Protocol INVITE
Requests", Work in Progress, July 2009.
[19] Mahy, R., Sparks, R., Rosenberg, J., Petrie, D., and A.
Johnson, "A Call Control and Multi-party usage framework for
the Session Initiation Protocol (SIP)", Work in Progress, March
2009.
[20] Johnston, A., Ed., Sparks, R., Cunningham, C., Donovan, S., and
K. Summers, "Session Initiation Protocol Service Examples", BCP
144, RFC 5359, October 2008.
[22] Boulton, C., Rosenberg, J., Camarillo, G. and F. Audet, "Best
Current Practices for NAT Traversal for Client-Server SIP",
Work in Progress, September 2008.
[23] Komu, M., Henderson, T., Tschofenig, H., Melen, J. and A.
Keraenen, "Basic HIP Extensions for Traversal of Network
Address Translators", Work in Progress, June 2009.
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[24] Moskowitz, R., "HIP Experimentation using Teredo", July 2008,
http://www.ietf.org/proceedings/08jul/slides/HIPRG-3.pdf.
[25] Rescorla, E. and N. Modadugu, "Datagram Transport Layer
Security", RFC 4347, April 2006.
[26] Fischl, J., Tschofenig, H. and E. Rescorla, "Framework for
Establishing an SRTP Security Context using DTLS", Work in
Progress, March 2009.
[27] McGrew, D. and E. Rescorla, "Datagram Transport Layer Security
(DTLS) Extension to Establish Keys for Secure Real-time
Transport Protocol (SRTP)", Work in Progress, February 2009.
[28] Zimmerman, P., Johnston, A. and J. Callas, "ZRTP: Media Path
Key Agreement for Secure RTP", Work in Progress, March 2009
[29] Peterson, J., Jennings, C. and R. Sparks, "Change Process for
the Session Initiation Protocol (SIP)", Work in Progress, July
2009.
[30] Raman, T.V., "Toward 2 exp(W), Beyond Web 2.0", Communications
of the ACM, Vol. 52, No.2, p. 52-59, February 2009.
[31] Wikipedia, "Rich Internet application",
http://en.wikipedia.org/wiki/Rich_Internet_Applications.
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Authors' Addresses
Henry Sinnreich
Adobe Systems, Inc.
601 Townsend Street,
San Francisco, CA 94103, USA
EMail: henrys@adobe.com
Alan Johnston
Avaya
Saint Louis, MO, USA
EMail: alan@sipstation.com
Eunsoo Shim
Avaya Labs Research
233 Mount Airy Road
Basking Ridge, NJ 07920 USA
EMail: eunsooshim@gmail.com
Kundan Singh
Columbia University Alumni
1214 Amsterdam Ave., MC0401
New York, NY, USA
EMail: kns10@cs.columbia.edu
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