Re: [rtcweb] RTCWeb default signaling protocol [was RE: About defining a signaling protocol for WebRTC (or not)]

Henry Sinnreich <henry.sinnreich@gmail.com> Tue, 20 September 2011 19:48 UTC

Return-Path: <henry.sinnreich@gmail.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 447E91F0C41 for <rtcweb@ietfa.amsl.com>; Tue, 20 Sep 2011 12:48:26 -0700 (PDT)
X-Virus-Scanned: amavisd-new at amsl.com
X-Spam-Flag: NO
X-Spam-Score: -3.207
X-Spam-Level:
X-Spam-Status: No, score=-3.207 tagged_above=-999 required=5 tests=[AWL=0.392, BAYES_00=-2.599, 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 bmTW3jU+vZ69 for <rtcweb@ietfa.amsl.com>; Tue, 20 Sep 2011 12:48:25 -0700 (PDT)
Received: from mail-gy0-f172.google.com (mail-gy0-f172.google.com [209.85.160.172]) by ietfa.amsl.com (Postfix) with ESMTP id 94BBE1F0C40 for <rtcweb@ietf.org>; Tue, 20 Sep 2011 12:48:25 -0700 (PDT)
Received: by gyd12 with SMTP id 12so778393gyd.31 for <rtcweb@ietf.org>; Tue, 20 Sep 2011 12:50:52 -0700 (PDT)
DKIM-Signature: v=1; a=rsa-sha256; c=relaxed/relaxed; d=gmail.com; s=gamma; h=user-agent:date:subject:from:to:cc:message-id:thread-topic :thread-index:in-reply-to:mime-version:content-type :content-transfer-encoding; bh=h3KLOxlgP0UaygT/6Nco11FIVpgvmBGvaH/8qWooYw4=; b=CbZUp7N26kIrHKyDFZ/DjhI3bFZ5Dg98rj3KWoUpOO+wtU6VeCM33oQFfmXx1qWfwX JjjzqrqyrMrzpaBBHhNebBtRpIMrcKSSqBcr0l7lQy5WTlA6Emc/Girj781fkiqCH5MT SU4+amql5mkrqkjV3SwfiRWu40drM468EBPxI=
Received: by 10.236.195.10 with SMTP id o10mr7460313yhn.115.1316548252053; Tue, 20 Sep 2011 12:50:52 -0700 (PDT)
Received: from [192.168.15.2] (cpe-76-184-249-163.tx.res.rr.com. [76.184.249.163]) by mx.google.com with ESMTPS id t56sm3274550yhh.20.2011.09.20.12.50.45 (version=TLSv1/SSLv3 cipher=OTHER); Tue, 20 Sep 2011 12:50:45 -0700 (PDT)
User-Agent: Microsoft-Entourage/12.30.0.110427
Date: Tue, 20 Sep 2011 14:50:42 -0500
From: Henry Sinnreich <henry.sinnreich@gmail.com>
To: Hadriel Kaplan <HKaplan@acmepacket.com>, Randell Jesup <randell-ietf@jesup.org>
Message-ID: <CA9E58C2.1DBDF%henry.sinnreich@gmail.com>
Thread-Topic: [rtcweb] RTCWeb default signaling protocol [was RE: About defining a signaling protocol for WebRTC (or not)]
Thread-Index: AQHMd2Ww2V9ZOtZta0qF5HzNdmfwDJVWrlwZ
In-Reply-To: <E4C646E9-44E5-4EBE-9AA1-D97500FAEE66@acmepacket.com>
Mime-version: 1.0
Content-type: text/plain; charset="US-ASCII"
Content-transfer-encoding: 7bit
Cc: "<rtcweb@ietf.org>" <rtcweb@ietf.org>
Subject: Re: [rtcweb] RTCWeb default signaling protocol [was RE: About defining a signaling protocol for WebRTC (or not)]
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: Tue, 20 Sep 2011 19:48:26 -0000

+1

> A full mesh is what *should* happen, but SIP/SDP can't do it, afaict.  It
> would treat them either as independent calls even at a media layer, or as a
> full-mixer conference focus model.  The closest thing we have would probably
> be the Join header, but I believe it's semantics is to join as a full mixer
> conf call.  Isn't this full-mesh media-forking thing actually a new semantic
> for SIP/SDP?  (it's hard to believe with 100+ drafts/RFCs this scenario hasn't
> already been addressed in SIP - I must be just having a memory leak)

Thanks, Henry


On 9/20/11 2:19 AM, "Hadriel Kaplan" <HKaplan@acmepacket.com> wrote:

> 
> On Sep 20, 2011, at 2:39 AM, Randell Jesup wrote:
> 
>> On 9/20/2011 1:59 AM, Hadriel Kaplan wrote:
>>> OK, let's take the game example... only 2-player games would be able to use
>>> a simple rtcweb-SIP agent.  Anything more than 2-player would want to use
>>> the multi-party "conferencing" model of rtcweb, which can't even be signaled
>>> with SIP today as far as I can tell. (not that I've thought about it too
>>> much, but I can't see how it would without some changes to SIP)
>> 
>> It should be easy - either as N-1 2-person calls to the person hosting the
>> game, or
>> N calls via a central server (equivalent to a mixer), or as a full mesh of
>> direct
>> calls (3 2-person calls for a 3-person game, 6 for a 4-person game, etc), or
>> even
>> sparse meshes (makes sense in a game where not all players are 'near' each
>> other).
> 
> Can't do N-1 2-person calls to the person hosting the game, because rtcweb
> doesn't support "full" mixing in the browser, only local media mixing. (ie,
> everyone will hear the person hosting the game, and the person hosting the
> game will hear everyone else, but the rest won't hear each other)
> 
> Calls via a central media server require a central media server, which kinda
> defeats the "easy" concept and using our shiny new toy of rtcweb.
> 
> A full mesh is what *should* happen, but SIP/SDP can't do it, afaict.  It
> would treat them either as independent calls even at a media layer, or as a
> full-mixer conference focus model.  The closest thing we have would probably
> be the Join header, but I believe it's semantics is to join as a full mixer
> conf call.  Isn't this full-mesh media-forking thing actually a new semantic
> for SIP/SDP?  (it's hard to believe with 100+ drafts/RFCs this scenario hasn't
> already been addressed in SIP - I must be just having a memory leak)
> 
> -hadriel
> 
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb