IETF Logo Mail Archive

[rtcweb] Let's define the purpose of WebRTC

Iñaki Baz Castillo <ibc@aliax.net> Sat, 05 November 2011 13:35 UTC

Return-Path: <ibc@aliax.net>
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 9E32421F8726 for <rtcweb@ietfa.amsl.com>; Sat, 5 Nov 2011 06:35:38 -0700 (PDT)
X-Virus-Scanned: amavisd-new at amsl.com
X-Spam-Flag: NO
X-Spam-Score: -2.038
X-Spam-Level:
X-Spam-Status: No, score=-2.038 tagged_above=-999 required=5 tests=[AWL=-0.561, BAYES_00=-2.599, FM_FORGED_GMAIL=0.622, J_CHICKENPOX_34=0.6, J_CHICKENPOX_43=0.6, MIME_8BIT_HEADER=0.3, 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 wxVxl+O8HC8J for <rtcweb@ietfa.amsl.com>; Sat, 5 Nov 2011 06:35:37 -0700 (PDT)
Received: from mail-vx0-f172.google.com (mail-vx0-f172.google.com [209.85.220.172]) by ietfa.amsl.com (Postfix) with ESMTP id 9E28521F84AF for <rtcweb@ietf.org>; Sat, 5 Nov 2011 06:35:37 -0700 (PDT)
Received: by vcbfl11 with SMTP id fl11so3328637vcb.31 for <rtcweb@ietf.org>; Sat, 05 Nov 2011 06:35:36 -0700 (PDT)
MIME-Version: 1.0
Received: by 10.52.33.84 with SMTP id p20mr19167563vdi.32.1320500135955; Sat, 05 Nov 2011 06:35:35 -0700 (PDT)
Received: by 10.220.107.206 with HTTP; Sat, 5 Nov 2011 06:35:35 -0700 (PDT)
Date: Sat, 05 Nov 2011 14:35:35 +0100
Message-ID: <CALiegfkVNVAs_MyU_-4koA4zRwSn1-FwLjY9g_oZVkhi9rSK5Q@mail.gmail.com>
From: Iñaki Baz Castillo <ibc@aliax.net>
To: rtcweb@ietf.org
Content-Type: text/plain; charset="UTF-8"
Content-Transfer-Encoding: quoted-printable
Subject: [rtcweb] Let's define the purpose of WebRTC
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, 05 Nov 2011 13:35:38 -0000

Hi, in theory WebRTC is about realtime communications for the Web, but
there is interest in making it interoperable with SIP networks. So:

- Who is interested in interoperability with SIP? telcos (and me).

- Are telcos the main target of *Web*RTC? I don't think so. There are
millons of websites out there that would be interested in realtime
communications without any kind of interaction with SIP networks.
There are also other VoIP protocols.

- What does require "interoperability with SIP"? does it mean that
WebRTC should allow plain RTP and no ICE? This has been discussed many
times in this WG: Security in the media plane MUST NOT be optional, it
MUST be a MUST. So sorry, but a legacy SIP device not implementing
SRTP+ICE cannot interoperate with a WebRTC endoint. Period.


I'm the first one interested in making WebRTC interoperable with SIP,
but NOT with insecure and legacy SIP. So I assume that 99% of current
SIP devices will NOT interoperate in the media plane with a WebRTC
endpoint without the help of a smart media gateway implementing
ICE+SRTP. Neither I know whether such media gateway is feasible or
not. Sorry if not.

So IMHO this WG should clarify these points so we can move on faster.
If these points are already stated by WebRTC specs then forget this
mail please, but I still see folks asking for "legacy SIP
interoperability without requiring SRTP or ICE".

Thanks a lot.

-- 
Iñaki Baz Castillo
<ibc@aliax.net>

v2.23.0 | Report a Bug | By Email