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Versions: 00 01 02 03 04 05 06 07 RFC 5638
SIPPING Working Group H. Sinnreich/Adobe, editor
Internet Draft A. Johnston/Avaya
E. Shim/Avaya
K. Singh/Columbia U. Alumni
Intended status: Informational
June 29, 2009
Expires: December 2009
Simple SIP Usage Scenario for Applications in the Endpoints
<draft-sinnreich-sip-tools-07>
Status of this Memo
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Abstract
For Internet-centric usage, the number of SIP required standards for
presence; 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
based on avoiding emulating telephony and its model of the
intelligent network. 'Simple SIP' by contrast relies on powerful
computing endpoints. Simple SIP desktop applications can be combined
with rich Internet applications (RIA). Significant telephony features
may also be implemented in the endpoints.
This approach for SIP reduces the number of SIP standards to comply
with, currently from roughly 100 and still growing, to about 11.
References for NAT traversal and for security are also provided.
Table of Contents
1. Introduction................................................3
2. The Endpoint in the SIP and Web Architectures...............5
The Telephony Gateway as a SIP Endpoint........................7
3. Applicability for 'simple SIP' in the Endpoints.............7
What 'simple SIP' can accomplish...............................7
Baseline for 'simple SIP'......................................8
What 'simple SIP' may or may not accomplish....................8
What is out of scope for 'simple SIP'..........................8
Borderline cases...............................................9
4. Mandatory SIP References for Internet-Centric Usage........10
RFC 3261: "SIP: Session Initiation Protocol"..................10
RFC 4566: "SDP: Session Description Protocol".................10
RFC 3264: "An Offer/Answer Model with SDP"....................11
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RFC 3840: "Indicating User Agent Capability in SIP"...........11
RFC 3263: "SIP: Locating SIP Servers".........................11
RFC 3265: "SIP-Specific Event Notification"...................11
RFC 3856: "A Presence Event Package for SIP"..................12
RFC 3863: "Presence Information Data Format (PIDF)"...........12
RFC 3428: "SIP Extension for Instant Messaging"...............12
RFC 4474: "Enhancements for Authenticated Identity Management in
SIP"..........................................................12
RFC 3581: "An Extension to SIP for Symmetric Response Routing"12
Updates to SIP Related Protocols..............................13
5. SIP Applications in the Endpoints..........................13
6. NAT Traversal..............................................14
7. Security Considerations....................................15
8. IANA Considerations........................................16
9. Acknowledgements...........................................16
10. References.................................................16
10.1 Mandatory References.....................................17
10.2 Informative References...................................17
Acknowledgment...................................................19
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 (RIA), both in the browser and on the desktop. For SIP
to be useful for RIA, legacy voice service provider requirements 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 services 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:
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1. Value proposition: SIP applications in the endpoints can be easily
mixed with RIA and thus enable service providers to offer new
services in a scalable and flexible manner as well as
significantly enhance the value of SIP applications. Rich Internet
applications support unrestricted user choice as an alternative to
and 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 that argues that network intermediaries cannot
completely understand the applications and their state in the
endpoints.
We will refer in the following by 'simple SIP' to the SIP functions
necessary to only support the rendezvous and session setup functions
of SIP, support voice, video, basic presence and instant messaging
and also support 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 as an
auxiliary to making a simple one-to-one call to voice mail if the
callee does not answer, or to check voice mail.
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, RIA adds most of the
value. The integrated use of SIP and RIA 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
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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. Given
however 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 other standards than those mentioned here. The
alternative to HIP is to use SIP specific protocols for NAT traversal
such as STUN, TURN and ICE 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."
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 servers 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 the 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 to describe here 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 being defined in
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standards organizations. They are referred to as Rich Internet
Applications.
Examples of RIA include social networks, blogs, wikis, web based
office and collaboration tools, task related apps for 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 RIA.
RIA 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 such as phones and desktop clients.
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 by 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: Characterizing a service, or
actually a mix of several service components (such as in a mash-up)
by a URI. This is in contrast to network services registration 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 potential of integrating SIP based multimedia communications
with access to RIA on the Web is the motivation behind this memo. To
mention a few scenarios: Adding SIP and RTP based real-time
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communications to RIA, 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.
The Telephony Gateway as a SIP Endpoint
Integrating SIP communications into RIA precludes in our opinion
carrying over to the Web legacy telephony features in order to
accomplish interoperability with the installed base of telephone
networks of various kinds. 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 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.
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 to consider
if and when SIP plus RIA applications could be used as an emergency
telephony enhancement or even replacement.
What 'simple SIP' can accomplish
The main driver for SIP applications on the desktop or in the browser
is to support the integration of SIP and RTP based real time
communications with RIA. 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 also extended to a medical
service taking care of patients or elderly people.
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In this case, 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 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 instances as when the "caller" cannot speak for
some reason.
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, encryption for the media and for
parts of the signaling in a manner consistent with the routing of SIP
messages.
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 [16] protocol 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 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.
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 which are listed here. This means to
avoid placing any functions in the network other than the rendezvous
function of SIP.
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o Avoiding the support of legacy telephony functions, such as
emulating public telephone switch services and voice-only
private branch exchanges.
o Avoiding 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 VoIP networks and in transit VoIP networks in between.
Long chains of intermediaries of any type are not only adding
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 that 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 new SIP
routing requirement.
o Avoiding the support for legacy telephony models such as
identifying end user devices for the purpose of differentiated
charging by type of service or charging for roaming between
networks.
o Avoiding policies and the associated policy servers and network
elements for Quality of Service (QoS) to enforce service rate
specific policies for real-time communications.
o Avoiding design considerations for SIP for compatibility with
legacy telephony networks, traditional telephony services and
the 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 to 'look like
the PSTN' are also avoided.
This list is not exhaustive, but conveys the concept on what to avoid
when using SIP as a simpler protocol to understand and to implement.
Borderline cases
There are also some interesting borderline cases for what to avoid,
such as the Reliability of Provisional Responses (PRACK) specified in
RFC 3262. PRACK is targeted for multi-hop SIP server networks and
PSTN interworking, especially to assure reliable early media. PRACK
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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 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 in 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'.
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 here to as 'simple
SIP'.
RFC 3261: "SIP: Session Initiation Protocol"
RFC 3261 [1] is the core specification for SIP. The trapezoid model
for SIP 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. SIP however can also work in peer-
to-peer (P2P) communications without SIP servers.
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
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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.
RFC 3264: "An Offer/Answer Model with 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 the following on NAT
traversal, this negotiation is usually considerably complicated due
to the existence of NAT between the SIP endpoints.
RFC 3840: "Indicating User Agent Capability in SIP"
A SIP UA can convey its capability in the Contact header field
indicating if it can support presence, IM, audio, video, if the
device is fixed or mobile and 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
such as grayed out or missing icons.
RFC 3263: "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 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.
RFC 3265: "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
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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.
RFC 3856: "A Presence Event Package for 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.
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.
RFC 3428: "SIP Extension for Instant Messaging"
The SIP extension for IM in RFC 3428 [9] consists in the MESSAGE
method defined here 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.
RFC 4474: "Enhancements for Authenticated Identity Management in 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.
RFC 3581: "An Extension to 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.
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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 SIP 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
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. The main beneficiary in this case may be however the
Web site, since the conversation can prolong the time for users
watching that Web site.
This shows the value of combining RIA with SIP based communications.
It is a matter of judgment by the end user if the Web content or the
associated communication capability is more important or if the mix
of both is most attractive.
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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 "SIP Service Examples" [20] contain 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.
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 NAT 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. This
document 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 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.
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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 the 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 and Firewalls" [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 [23] 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.
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 SIP" mentioned previously specifies the use of certificates
for secure identification of the parties involved in SIP
signaling requests.
o The Datagram Transport Layer Security specified in RFC 4347 [27]
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 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].
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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. IANA Considerations
There are no IANA considerations associated with this memo.
9. 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 to discuss the use of emergency calls using simple SIP.
Robert Sparks has pointed to some missing references, which we have
added.
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 the clarifications 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.
This document was prepared using 2-Word-v2.0.template.dot.
10. References
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10.1 Mandatory References
[1] RFC 3261: "SIP: Session Initiation Protocol." by J. Rosenberg et
al. IETF, June 2002.
[2] RFC 4566: "SDP: Session Description Protocol" by M. Handley et
al. IETF, July 2006.
[3] RFC 3264: "An Offer/Answer Model with the Session Description
Protocol (SDP)" by J. Rosenberg et al. IETF, June 2002.
[4] RFC 3840: "Indicating User Agent Capabilities in SIP" by J.
Rosenberg et al. IETF, August 2004.
[5] RFC 3263: "SIP: Locating SIP Servers" by J. Rosenberg and H.
Schulzrinne. IETF, June 2002.
[6] RFC 3265: "SIP-Specific Event Notification" by A. Roach. IETF,
June 2002.
[7] RFC 3856: "A Presence Event Package for the Session Initiation
Protocol" by J. Rosenberg. IETF, August 2004.
[8] RFC 3863: "Presence Information Data Format (PIDF)" by J. Sugano
et al. IETF, August 2004.
[9] RFC 3428: "SIP Extension for Instant Messaging" by B.Campbell et
al. IETF, December 2002.
[10] RFC 4474: "Enhancement for Authenticated Identity Management in
the Session Initiation Protocol (SIP)" by J. Peterson et al. IETF,
August 2006.
[11] RFC 3581: "An Extension to SIP for Symmetric Response Routing"
by J. Rosenberg and H. Schulzrinne. IETF, August 2003.
10.2 Informative References
[12] RFC 54: "A Hitchhiker's Guide to the Session Initiation
Protocol(SIP)" by J. Rosenberg, IETF November 2008.
[13] "VoIP RFC Watch by Nils Ohlmeier". Web site counting SIP related
standards at http://rfc3261.net/"
[14] B. Rosen and J. Polk: "Best Current Practice for Communications
Services in support of Emergency Calling". Internet-Draft of the
Ecrit WG, IETF, June 2009. Work in progress.
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[15] RFC 5031: "A Uniform Resource Name (URN) f or Emergency and
Other Well-Known Services" by H. Schulzrinne. IETF, January 2008.
[16] RFC 5222: "LoST: A Location-to-Service Translation Protocol" by
T. Hardie al. IETF, August 2008.
[17] RFC 4320: "Actions Addressing Identified Issues with SIP Non-
Invite Transactions" by R. Sparks. IETF, January 2006.
[18] "Correct Transaction Handling for 200 Responses to the SIP
INVITE Requests" by R. Sparks, Internet-Draft, work in progress,
IETF, March 2009.
[19] "A Call Control and Multi-party usage framework for the Session
Initiation Protocol (SIP)" by R. Mahy et al. Internet-Draft. IETF
Macrh 2009, work in progress.
[20] RFC 5359: "Session Initiation Protocol Service Examples" by A.
Johnston et al. IETF, November 2008.
[22] "Best Current Practices for NAT Traversal for SIP" by C. Boulton
et al., IETF, September 2008. Internet-Draft.
[23] "Basic HIP Extensions for Traversal of Network Address
Translators" by M. Komu et al. Internet-Draft, IETF, June 2009, work
in progress.
[24] "HIP Experimentation using Teredo" by R. Moskowitz. HIPRG in the
IETF, June 2008. http://www.ietf.org/proceedings/08jul/slides/HIPRG-
3.pdf
[25] "RFC 4347: "Datagram Transport layer Security" by E. Rescorla et
al. IETF, April 2006.
[26] Framework for Establishing an SRTP Security Context using DTLS
by J. Fischl et al. Internet-Draft <draft-ietf-sip-dtls-srtp-
framework-07>, March 2009. To be published as an RFC.
[27] Datagram Transport Layer Security (DTLS) Extension to Establish
Keys for Secure Real-time Transport Protocol (SRTP) by D. McGrew and
E. Rescorla. Internet Draft, February 2009, work in progress.
[28] "ZRTP: Media Path Key Agreement for Secure RTP" by P. Zimmermann
et al. Internet-Draft, IETF, March 2009, work in progress.
[29]"Change Process for SIP" by J. Peterson and C. Jennings.
Internet-Draft, IETF, February 26 2009, work in progress.
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[30] Toward 2 exp(W), Beyond Web 2.0" by T.V. Raman. Communications
of the ACM, February 2009, Vol. 52, No.2, p. 52-59.
[31] A convenient starting point for information on Rich Internet
Applications can be found at
http://en.wikipedia.org/wiki/Rich_Internet_Applications
Author's Addresses
Henry Sinnreich
Adobe Systems, Inc.
601 Townsend Street,
San Francisco, CA 94103, USA
Email: henrys@adobe.com
Alan B. 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
Acknowledgment
Funding for the RFC Editor function is provided by the IETF
Administrative Support Activity (IASA).
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