Re: [rtcweb] WebRTC service between SPs

Hrishikesh Kulkarni <rishi@turtleyogi.com> Sat, 29 June 2013 06:17 UTC

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Date: Sat, 29 Jun 2013 11:47:49 +0530
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From: Hrishikesh Kulkarni <rishi@turtleyogi.com>
To: Moises Silva <moises.silva@gmail.com>
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Cc: "Wangyahui \(Yahui\)" <yahui.wang@huawei.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] WebRTC service between SPs
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SIP is an established standard to interoperate domains. We at
OneKlikStreet.com developed a video/audio bridging service for WebRTC.
Although it uses JS/JSON signaling for web based clients. Our server could
very well federate with any other service using SIP. What does need to be
discussed on app to app basis is what kind of federation you are looking
for?
In case of bridging service we could merge calls from both servers or
redirect all the calls to the host service.

regards,
Rishi
Founder, OneKlikStreet.com




On Fri, Jun 28, 2013 at 11:55 PM, Moises Silva <moises.silva@gmail.com>wrote:

>
> On Fri, Jun 28, 2013 at 12:03 PM, Jim Barnett <Jim.Barnett@genesyslab.com>wrote:
>
>>  As I understand it, it’s not just a problem of identities.  WebRTC does
>> not define the signaling protocol, but leaves it  up to the application.
>> If two users download their applications/JavaScript from the same site, it
>> won’t be a problem, because the same application is handling both ends of
>> the call.  But if one user is on site A while the other is on site B, there
>> is no guarantee that either site’s application will understand the
>> signaling from the other.****
>>
>> **
>>
>
> Unless websites agree to use something standard such as SIP/Jingle for
> federation (inter website/domain communication).
>
> -
> Moy
>
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