[RTW] [dispatch] =?windows-1251?B?zcA=?=: RTC-Web I-D about interworking between RTC-Web and SIP-RTP

[Henry Sinnreich]

+1

> But in practice, real-time data such as vnc, is also now part of real-time
> media session, and it must follow the same Nat-traversal scenario, as audio &
> video, but not necessarily the rtp protocol

Indeed, many people spend far more time with VNC than making phone/video
calls. And as argued here before, the data content in RTP can be transported
over HTTP as well, along with all other metadata.

A possible guideline here is:

>as long as we have an architecture which one *can* instantiate a component for
>the signalling that does SIP and a component for media-transfer that does RTP,
>and so on, this requirement is met.  And I would have thought that yes, we
>would agree the architecture should make that possible.

>whether we choose those components as a baseline set to implement is much more
>a debate, I'd say.

This last sentence is the key IMO.
Thanks, Henry


On 2/9/11 2:17 PM, "Slava Borilin" <Borilin at spiritdsp.com> wrote:

> Dear all,
> 
> Let me throw in another question- when all of us are talking media, we think
> its audio-video only.
> 
> But in practice, real-time data such as vnc, is also now part of real-time
> media session, and it must follow the same Nat-traversal scenario, as audio &
> video, but not necessarily the rtp protocol (but TCP). I think if we put stun
> in browser, it should be available not only for rtp sessions.
> 
> Based on videoconferencing experience.
> 
> 
> Слава
> 
> ----- Reply message -----
> От: "Markus.Isomaki at nokia.com" <Markus.Isomaki at nokia.com>
> Дата: ср, фев 9, 2011 22:53
> Тема: [dispatch] [RTW] RTC-Web I-D about interworking between RTC-Web and
> SIP-RTP
> Кому: "xavier.marjou at orange-ftgroup.com" <xavier.marjou at orange-ftgroup.com>,
> "dispatch at ietf.org" <dispatch at ietf.org>
> Копия: "rtc-web at alvestrand.no" <rtc-web at alvestrand.no>
> 
> Hi Xavier and Jean-Francois,
> 
> Thanks for putting this together.
> 
> Based on the recent list discussion, I would say that quite many people are
> leaning towards the architecture you depict in Section 5.2, Figure 2: The
> session setup protocol is an application specific Javascript implementation
> transported over HTTP or WebSocket, while media is running on standard RTP
> supported by the browser.
> 
> In that model we can’t put many requirements on the session setup protocol or
> its interworking with SIP. If the service provider needs SIP interoperability
> (to connect to PSTN, to other service providers or SIP phones), it is indeed
> THEIR burden to make sure they use something that has a clean mapping to SIP –
> for instance, that they can do things like call hold. On the other hand if the
> service provider is not interested in SIP interoperability, they do not have
> to worry about that.  In the IETF there are probably two ways to address this
> interworking: a) do nothing and leave it completely to the implementers and
> service providers, or b) define some kind of a SIP/BOSH/HTTP or SIP/WebSocket
> mapping in the same way that the XMPP folks have done. The XMPP/BOSH spec does
> have implementations both on the client/Javascript side as well as on the
> server side, so I think that spec has had some value. (At least in a way that
> the Javascript library and the BOSH servers can be implemented somewhat
> independently.)
> 
> The RTP/media stack on the other hand is definitely in the scope of the IETF
> effort. I think we should standardize the RTP use in the browsers and that
> would be one step towards interop with SIP phones. The critical thing seems to
> be the STUN connectivity check or media authorization part. If we mandate
> browsers to get that exchange done before they are allowed to generate any RTP
> packets on behalf of the application, this will ruin the possibility of
> interop with 99% of existing SIP clients (without some kind of an SBC). DTMF
> transport capability may also be relevant interop requirement.
> 
> I think these are the key issues we should consider wrt. SIP phone interop.
> 
> Markus
> 
> 
> From: rtc-web-bounces at alvestrand.no [mailto:rtc-web-bounces at alvestrand.no] On
> Behalf Of ext Xavier Marjou
> Sent: 09 February, 2011 11:07
> To: DISPATCH list
> Cc: rtc-web at alvestrand.no; Jean-François Jestin
> Subject: [RTW] [dispatch] RTC-Web I-D about interworking between RTC-Web and
> SIP-RTP
> 
> Hi,
> 
> We have posted a draft about interworking requirements between RTC-Web and
> SIP-RTP.
> http://www.ietf.org/internet-drafts/draft-marjou-dispatch-rtcweb-sip-rtp-inter
> wk-reqs-00.txt
> 
> Cheers,
> Xavier and Jean-François
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