Re: [rtcweb] WebRTC service between SPs

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

Return-Path: <rishi@turtleyogi.com>
X-Original-To: rtcweb@ietfa.amsl.com
Delivered-To: rtcweb@ietfa.amsl.com
Received: from localhost (localhost [127.0.0.1]) by ietfa.amsl.com (Postfix) with ESMTP id 1619921F9E54 for <rtcweb@ietfa.amsl.com>; Fri, 28 Jun 2013 23:17:55 -0700 (PDT)
X-Virus-Scanned: amavisd-new at amsl.com
X-Spam-Flag: NO
X-Spam-Score: -2.976
X-Spam-Level:
X-Spam-Status: No, score=-2.976 tagged_above=-999 required=5 tests=[BAYES_00=-2.599, FM_FORGED_GMAIL=0.622, HTML_MESSAGE=0.001, RCVD_IN_DNSWL_LOW=-1]
Received: from mail.ietf.org ([12.22.58.30]) by localhost (ietfa.amsl.com [127.0.0.1]) (amavisd-new, port 10024) with ESMTP id 6kB+XWflzoYl for <rtcweb@ietfa.amsl.com>; Fri, 28 Jun 2013 23:17:50 -0700 (PDT)
Received: from mail-ie0-f174.google.com (mail-ie0-f174.google.com [209.85.223.174]) by ietfa.amsl.com (Postfix) with ESMTP id 8916E21F9E43 for <rtcweb@ietf.org>; Fri, 28 Jun 2013 23:17:50 -0700 (PDT)
Received: by mail-ie0-f174.google.com with SMTP id 9so5800421iec.33 for <rtcweb@ietf.org>; Fri, 28 Jun 2013 23:17:50 -0700 (PDT)
X-Google-DKIM-Signature: v=1; a=rsa-sha256; c=relaxed/relaxed; d=google.com; s=20120113; h=mime-version:x-originating-ip:in-reply-to:references:date :message-id:subject:from:to:cc:content-type:x-gm-message-state; bh=ZHPpnbdB+7Yas4XEv/Q0lqy7++RBxbwsX1LjRaVM2SE=; b=GxOIIhlVUwQWCT1dbm/PCDcpqhz1vRECt8ek9PhR8ZUHli3vwKLR5/2hTqPWKOILjL EiRDkm+riypmhF4w+kY7I2FobL/k4PzW0d9B1EHpdEfUDIsS6k/ddmP+wOyOHuJPD83U FTgIomQUIEt/BXxB0xn3nKBQwNKlWA+74IWeIifNgeTuCIPl0+PKU43BDRckUyGHWLNx M5mvtnkm8cTyULw4B+zt+iiNNFM2XgpihDpUCpGYYnhILna9B+X4aQJDTEIJYSd2yoM6 AmAGc1lHBqNJrV0kEhGcJP63Rspdi8vJSqaPUhe7H6lfaqIuBOkBkGAJ9a/zcLe4hc/x 1sRQ==
MIME-Version: 1.0
X-Received: by 10.50.136.196 with SMTP id qc4mr7265055igb.21.1372486669941; Fri, 28 Jun 2013 23:17:49 -0700 (PDT)
Received: by 10.64.231.36 with HTTP; Fri, 28 Jun 2013 23:17:49 -0700 (PDT)
X-Originating-IP: [122.166.178.191]
In-Reply-To: <CAA4nhyCLd_JGdGaqGFN3e5qi7eDy4yLVdpSLYU76HCa4AmcUUQ@mail.gmail.com>
References: <034C870DB898BE43B5787C7A79107CD94BFA4E1B@nkgeml507-mbx.china.huawei.com> <57A15FAF9E58F841B2B1651FFE16D281052E6B@GENSJZMBX01.msg.int.genesyslab.com> <CAA4nhyCLd_JGdGaqGFN3e5qi7eDy4yLVdpSLYU76HCa4AmcUUQ@mail.gmail.com>
Date: Sat, 29 Jun 2013 11:47:49 +0530
Message-ID: <CALFWOz4EqVXOTJAUpJ1dU22fpx5J3S5VowFMd=EwkM4sXSSHEA@mail.gmail.com>
From: Hrishikesh Kulkarni <rishi@turtleyogi.com>
To: Moises Silva <moises.silva@gmail.com>
Content-Type: multipart/alternative; boundary="089e01494eee6bd20804e044f49c"
X-Gm-Message-State: ALoCoQkbASdKID9zJqLeaG2PTjDSdPblvWugiieuUPpPudd/erhsvBIm8CAGIMvQ/OzaQvPl7mYo
Cc: "Wangyahui (Yahui)" <yahui.wang@huawei.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] WebRTC service between SPs
X-BeenThere: rtcweb@ietf.org
X-Mailman-Version: 2.1.12
Precedence: list
List-Id: Real-Time Communication in WEB-browsers working group list <rtcweb.ietf.org>
List-Unsubscribe: <https://www.ietf.org/mailman/options/rtcweb>, <mailto:rtcweb-request@ietf.org?subject=unsubscribe>
List-Archive: <http://www.ietf.org/mail-archive/web/rtcweb>
List-Post: <mailto:rtcweb@ietf.org>
List-Help: <mailto:rtcweb-request@ietf.org?subject=help>
List-Subscribe: <https://www.ietf.org/mailman/listinfo/rtcweb>, <mailto:rtcweb-request@ietf.org?subject=subscribe>
X-List-Received-Date: Sat, 29 Jun 2013 06:17:55 -0000

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
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>
>