One document matched: draft-marjou-dispatch-rtcweb-sip-rtp-interwk-reqs-00.txt
Network Working Group X. Marjou
Internet-Draft JF. Jestin
Intended status: Informational France Telecom Orange
Expires: August 13, 2011 February 9, 2011
Requirements for interworking between RTC-Web and SIP-RTP protocols
draft-marjou-dispatch-rtcweb-sip-rtp-interwk-reqs-00
Abstract
In the context of [RTC-Web], some work is emerging to make real-time
communications possible in a web browser. This document defines a
minimal set of requirements so that such applications interoperate
with SIP-RTP applications.
Status of this Memo
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This Internet-Draft will expire on August 13, 2011.
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Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . . 3
2. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 3
3. Definitions . . . . . . . . . . . . . . . . . . . . . . . . . 4
4. Use cases . . . . . . . . . . . . . . . . . . . . . . . . . . 4
4.1. SIP Multimedia application reachability extension . . . . 4
4.2. RTC-Web applications with integration of SIP device . . . 4
4.3. RTC-Web and SIP service provider interconnection . . . . . 4
5. Possible RTC-Web/SIP interworking architectures . . . . . . . 4
5.1. SIP-RTP Stack in the Browser . . . . . . . . . . . . . . . 5
5.2. New Signalling Scheme and RTP Stack in the Browser . . . . 5
5.3. New Signalling Scheme and New Media Protocol in the
Browser . . . . . . . . . . . . . . . . . . . . . . . . . 7
5.4. Analysis with regards to interworking . . . . . . . . . . 8
6. Requirements . . . . . . . . . . . . . . . . . . . . . . . . . 9
6.1. Generic requirements: . . . . . . . . . . . . . . . . . . 9
6.2. Signalling level requirements: . . . . . . . . . . . . . . 9
6.3. Media level requirements: . . . . . . . . . . . . . . . . 9
6.4. Codec level requirements: . . . . . . . . . . . . . . . . 10
7. Security Considerations . . . . . . . . . . . . . . . . . . . 10
8. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 10
9. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 10
10. References . . . . . . . . . . . . . . . . . . . . . . . . . . 10
10.1. Normative references . . . . . . . . . . . . . . . . . . . 10
10.2. Informative references . . . . . . . . . . . . . . . . . . 11
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 11
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1. Introduction
In the context of [RTC-Web], some work is emerging to make real-time
communications possible in a web browser. Such work will allow to
use the UDP protocol to transport real time data.
This document defines a minimal set of requirements so that RTC-Web
applications interoperate with applications based on SIP ([RFC3261])
and RTP ([RFC3550]) protocols.
On the one hand, bringing real-time communication capability in the
web browser RTC-Web promises to offer great value to end-users,
developers and service providers. This value comes from the ubiquity
of the web browser and the web architecture, from the simple
programatic model the web offer and indeed the innovation perspective
of such solution.
On the other hand, SIP and RTP protocols are broadly used to
implement real-time or near real-time applications. This is
particularly true for voice, video, instant messaging, presence and
content sharing and when considering available implementations in
devices (hard phone, mobile phone...), network infrastructures (e.g.
SIP based architecture) and service provider interconnection
gateways.
Allowing both solutions to interoperate promises RTC-Web solution to
have greater value as it will allow this solution to reach legacy SIP
multimedia devices and networks and vice versa.
Section 4 describes some use cases. Section 5 reports the different
possible architectures. Section 6 finally states a set of
requirements the ongoing RTC-Web solution definition should fulfil to
be able to interoperate with SIP based multimedia applications.
Note: This document does not directly address RTC-Web service
provider interconnection except if this interconnection is based on
SIP-RTP.
2. Terminology
In this document, the key words "MUST", "MUST NOT", "REQUIRED",
"SHALL", "SHALL NOT", "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY",
and "OPTIONAL" are to be interpreted as described in RFC 2119
[RFC2119].
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3. Definitions
RTC-WEB: the term [RTC-Web] refers to the ongoing work on defining a
solution that enables real time applications such as bidirectional
audio and video within web applications
SIP-RTP: the term SIP-RTP is a generic term which refers to SIP and
RTP ecosystem protocol stack. This includes, non exhaustively: SIP,
SDP, RTP, RTCP, SRTP...
Note: To be adjusted to fit with the current on going [RTC-Web]
charter.
4. Use cases
This section presents some use cases involving interworking between
RTC-Web and SIP applications. These use cases include scenarios
where real-time audio and/or video are exchanged.
4.1. SIP Multimedia application reachability extension
Alice wants to access its services. Her service provider A (e.g.
atlanta.example.org) hosts these services on SIP-RTP servers. Alice
can use a web browser implementing an RTC-Web extension to reach its
service.
4.2. RTC-Web applications with integration of SIP device
Bob wants to access its services. His service provider B (e.g.
biloxy.example.org) hosts these services on RTC-Web servers. Bob can
use a device implementing an SIP-RTP extension to reach its service.
4.3. RTC-Web and SIP service provider interconnection
All the users of service provider A (e.g. atlanta.example.org) use an
RTC-Web application. All the users of service provider B (e.g.
biloxi.example.org) use a SIP-RTP application. Both service
providers want to make communications possible between all these
users.
This use case is typically an inter service operators use case.
5. Possible RTC-Web/SIP interworking architectures
This section outlines different architectures to realize RTC-Web/
SIP-RTP interworking. This section does not pretend to be exhaustive
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in term of architecture description but intends to propose families
of models any kind of solution should fit in.
These architectures satisfy the use cases listed above. However, It
must be noted that depending on the considered use case, additional
components may be necessary.
In this section, the name SIP used alone is a shortcut for SIP and
SDP protocols. Similarly, RTP used alone is a shortcut for RTP,
RTCP, and SRTP protocols.
5.1. SIP-RTP Stack in the Browser
This architecture consists in directly implementing a SIP-RTP
protocol stack in the browser, enabling a direct connection between
an RTC-Web application in a browser and a SIP-RTP phone.
Architecture with SIP-RTP in the browser:
----------------- -----------------
| RTC-Web | | SIP-RTP |
| application | | application |
|-----------------| | |
|rtcweb rtcweb | | |
| sig media | | |
| APIs APIs | | |
|-----------------| | |
| | | |
| ----- | | ----- |
|| SIP | |<-----------SIP----------->|| SIP | |
||stack| | ||stack| |
| ----- | | ----- |
| ----- | | ----- |
| | RTP ||<-----------RTP----------->| | RTP ||
| |stack|| | |stack||
| ----- | | ----- |
----------------- -----------------
Figure 1
5.2. New Signalling Scheme and RTP Stack in the Browser
This architecture consists in implementing a new signalling scheme
and an RTP stack in the browser.
A new signalling scheme means refers to two possible models:
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o Session management data set by API and transported by an
application protocol (e.g. HTTP or WebSockets). Figure 2
illustrates such architecture with XXX as the session management
data. The HTTP stack shown in the figure is the regular HTTP
stack available by default in all web browsers. Having SIP (or
part of it) embedded in HTTP in one possible implementation, as
indicated in [draft-sinnreich-sip-web-apis-01].
o A session management protocol different from SIP (e.g. XMPP,
MEGACO). Figure 3 illustrates such architecture with YYY as the
signalling protocol.
Both models relax constraints on the technology choice to implement
the RTC-Web solution but add constraints on end-to-end compatibility
with SIP-RTP applications by requiring the implementation of a
gateway to map one protocol into another one.
Architecture with HTTP in the browser:
----------------- -----------------
| RTC-Web | | SIP-RTP |
| application | | application |
|-----------------| | |
|rtcweb rtcweb | | |
| sig media | | |
| APIs APIs | | |
| | XXX | | | |
|---|---------|---| XXX | |
| --|-- | | over -------- | ----- |
||HTTP | | |<-HTTP->|HTTP|SIP|<-SIP-->|| SIP | |
||stack| | | | GW | ||stack| |
| ----- | | -------- | ----- |
| --|-- | | ----- |
| | RTP ||<----------RTP----------->| | RTP ||
| |stack|| | |stack||
| ----- | | ----- |
----------------- -----------------
Figure 2
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Architecture with another protocol than SIP or HTTP in the browser:
----------------- -----------------
| RTC-Web | | SIP-RTP |
| application | | application |
|-----------------| | |
|rtcweb rtcweb | | |
| sig media | | |
| APIs APIs | | |
|-----------------| | |
| ----- | -------- | ----- |
|| YYY | |<-YYY-->| YYY|SIP|<-SIP-->|| SIP | |
||stack| | | GW | ||Stack| |
| ----- | -------- | ----- |
| ----- | | ----- |
| | RTP ||<----------RTP----------->| | RTP ||
| |stack|| | |stack||
| ----- | | ----- |
----------------- -----------------
Figure 3
5.3. New Signalling Scheme and New Media Protocol in the Browser
This architecture consists in implementing different protocols in
RTC-Web and SIP-RTP frameworks, both for at the signalling level and
at the media level.
Such architecture requires interworking work (protocol mapping,
gateway) both for the signalling and the media protocols.
This architecture relaxes constraints on the technology choice to
implement the RTC-Web solution but adds constraints on end-to-end
compatibility with SIP-RTP applications by requiring the
implementation of gateway(s) to adapt protocols and media payloads.
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Architecture with another protocol than RTP as a media protocol:
----------------- -----------------
| RTC-Web | | SIP-RTP |
| application | | application |
|-----------------| | |
|rtcweb rtcweb | | |
| sig media | | |
| APIs APIs | | |
|-----------------| | |
| ----- | ------- | ----- |
||sig. | |<-sig->|sig|SIP|<-SIP-->|| SIP | |
||stack| | | GW | ||stack| |
| ----- | ------- | ----- |
| ----- | ------- | ----- |
| |med. ||<-med->|med|RTP|<-RTP-->| |RTP ||
| |stack|| | GW | | |stack||
| ----- | ------- | ----- |
----------------- -----------------
Figure 4
5.4. Analysis with regards to interworking
Using a full SIP-RTP stack in the browser (Section 5.1) would
undoubtedly be the best solution with regards to interworking: it
would avoid specifying new protocols and it would thus avoid the
control plane interworking problem described in [RFC3439] (i.e. no
need for protocol mapping). It nevertheless requires a granular API
to configure and access the protocol stack.
Using the RTP protocol suite but another than the SIP protocol
(Section 5.2) add the burden of interworking efforts at the
signalling level. The level of complexity of this gateway depends on
how much the signaling protocol will look like SIP. However, having
HTTP (or WebSocket) as a protocol transporting the signaling data is
attractive due to the central role played by this protocol in Web
environments.
Using both new signalling and media protocols in the browser
(Section 5.3) has been presented above for the sake of exhaustiveness
but this solution is not attractive for SIP-RTP interworking: it
increases the interworking efforts by requiring work at the media
level (new media protocol, complexity and cost of interworking
gateways...), whereas adding no identified advantages with regards to
the existing RTP/UDP protocol suite.
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6. Requirements
Whatever the architecture solution the RTC-Web will retain, a
reasonable way-forward is to specify its protocols and APIs taking
care of interworking with SIP-RTP devices. As such the following
requirements are proposed to the RTC-Web working group:
6.1. Generic requirements:
GENERIC-REQ-1 The [RTC-Web] solution MUST be designed in a such way
it allows interworking with SIP-RTP applications both
at the signalling and media level.
6.2. Signalling level requirements:
SIG-REQ-1 The [RTC-Web] solution MUST be designed in a way it allows
interoperability with SIP based multimedia applications.
This is typically applicable for identifiers, credentials,
state machine, and message types.
SIG-REQ-2 The [RTC-Web] solution MUST include a way to negotiate
media format as in Offer/Answer model used in SIP
([RFC3264])
SIG-REQ-3 The [RTC-Web] solution MUST include a way to interoperate
with ([RFC5939])
SIG-REQ-4 The [RTC-Web] solution MUST allow end to end codec
negotiation between RTC-web device and SIP device
SIG-REQ-5 The [RTC-Web] solution MUST include a compatibility/
mapping with SDP([RFC4566])
SIG-REQ-6 The [RTC-Web] solution SHOULD NOT require SIP-RTP
extensions.
6.3. Media level requirements:
MEDIA-REQ-1 The [RTC-Web] solution MUST be designed in a way it does
not mandate a gateway at media level when interworking
with SIP based multimedia application, consequently it
must be based on RTP/RTCP protocol suite over UDP for
real-time media.
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MEDIA-REQ-2 The [RTC-Web] solution MUST be compatible with a media
gateway architecture and not rely exclusively on a peer
to peer (between RTC-Web devices)...
MEDIA-REQ-3 The [RTC-Web] solution MUST allow the configuration of
some media-related parameters per session (e.g. buffer
size, packetization...).
6.4. Codec level requirements:
CODEC-REQ-1 The [RTC-Web] solution MUST allow codecs available in
existing SIP-RTP applications. A non exhaustive list is
the following: G.711, G.722, AMR, AMR-WB, H.264.
7. Security Considerations
SEC-REQ-1 RTC-Web and SIP-RTP interworking solution MUST NOT
compromise inherent security feature(s) developed and used
for both RTC-Web and SIP-RTP solutions.
8. IANA Considerations
None.
9. Acknowledgements
Thank you to Bruno Chatras, Christophe Eyrignoux, and Sebastien
Cubaud who provided early feedback.
10. References
10.1. Normative references
[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
Requirement Levels", BCP 14, RFC 2119, March 1997.
[RFC2616] Fielding, R., Gettys, J., Mogul, J., Frystyk, H.,
Masinter, L., Leach, P., and T. Berners-Lee, "Hypertext
Transfer Protocol -- HTTP/1.1", RFC 2616, June 1999.
[RFC3261] Rosenberg, J., Schulzrinne, H., Camarillo, G., Johnston,
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A., Peterson, J., Sparks, R., Handley, M., and E.
Schooler, "SIP: Session Initiation Protocol", RFC 3261,
June 2002.
[RFC3264] Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model
with Session Description Protocol (SDP)", RFC 3264,
June 2002.
[RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V.
Jacobson, "RTP: A Transport Protocol for Real-Time
Applications", STD 64, RFC 3550, July 2003.
[RFC3960] Camarillo, G. and H. Schulzrinne, "Early Media and Ringing
Tone Generation in the Session Initiation Protocol (SIP)",
RFC 3960, December 2004.
[RFC4566] Handley, M., Jacobson, V., and C. Perkins, "SDP: Session
Description Protocol", RFC 4566, July 2006.
[RFC5939] Andreasen, F., "Session Description Protocol (SDP)
Capability Negotiation", RFC 5939, September 2010.
10.2. Informative references
[I-D.sinnreich-sip-web-apis]
Sinnreich, H. and A. Johnston, "SIP APIs for
Communications on the Web",
draft-sinnreich-sip-web-apis-01 (work in progress),
June 2010.
[RFC3439] Bush, R. and D. Meyer, "Some Internet Architectural
Guidelines and Philosophy", RFC 3439, December 2002.
[RTC-Web] RTC-Web, "http://rtc-web.alvestrand.com/".
Authors' Addresses
Xavier Marjou
France Telecom Orange
2, avenue Pierre Marzin
Lannion 22307
France
Email: xavier.marjou@orange-ftgroup.com
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Jean-Francois Jestin
France Telecom Orange
2, avenue Pierre Marzin
Lannion 22307
France
Email: jeanfrancois.jestin@orange-ftgroup.com
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