Re: AW: AW: [Sip] Extension of conference procedures

"Peili Xu" <xupeili@gmail.com> Wed, 12 September 2007 14:14 UTC

Return-path: <sip-bounces@ietf.org>
Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1IVSzY-0002p0-MU; Wed, 12 Sep 2007 10:14:56 -0400
Received: from sip by megatron.ietf.org with local (Exim 4.43) id 1IVSzX-0002ot-2c for sip-confirm+ok@megatron.ietf.org; Wed, 12 Sep 2007 10:14:55 -0400
Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1IVSzW-0002ol-P9 for sip@ietf.org; Wed, 12 Sep 2007 10:14:54 -0400
Received: from fk-out-0910.google.com ([209.85.128.188]) by ietf-mx.ietf.org with esmtp (Exim 4.43) id 1IVSzV-00083W-H4 for sip@ietf.org; Wed, 12 Sep 2007 10:14:54 -0400
Received: by fk-out-0910.google.com with SMTP id z23so190185fkz for <sip@ietf.org>; Wed, 12 Sep 2007 07:14:52 -0700 (PDT)
DKIM-Signature: v=1; a=rsa-sha256; c=relaxed/relaxed; d=gmail.com; s=beta; h=domainkey-signature:received:received:message-id:date:from:to:subject:cc:in-reply-to:mime-version:content-type:content-transfer-encoding:content-disposition:references; bh=zM8JQhCuP7RC9rUJyJBnHiFYip8ReAsEJWMQRNEj+DM=; b=MQqlslxtuEw/+oInVbb1ZLf2IyRVgCFCTpoqqbMW9QqTjqLyu66UyoUp2S0aZeq8OkcU4VunPJr5uZKWtTRIhoSspmc5ki5BSIYm+eFB30PvYCEjn0p2r+5PNMYqXxgzPe7LrafQ8zfxvHA6UZY3jp+wU/k5MBp/8HYfR5jrb1I=
DomainKey-Signature: a=rsa-sha1; c=nofws; d=gmail.com; s=beta; h=received:message-id:date:from:to:subject:cc:in-reply-to:mime-version:content-type:content-transfer-encoding:content-disposition:references; b=k+CBOrD3q9fVuE8TM+4g04MgH/O9vev3WGOxDktI9cfV9HccKKLKPSJZcw19GD2AFcZwktGFY9aBOqf337HbLbaOJPU+MVYXrefTH/plE4nwKrarnk1niQEHCNFoAlTs/lTVq5Ed4oqHa5iug3K8aa/yxfOfIadFj/FpSTXTv7A=
Received: by 10.82.100.1 with SMTP id x1mr7742337bub.1189606492514; Wed, 12 Sep 2007 07:14:52 -0700 (PDT)
Received: by 10.82.107.19 with HTTP; Wed, 12 Sep 2007 07:14:52 -0700 (PDT)
Message-ID: <317ceda50709120714y2f17422dxbb49cb943d1928b4@mail.gmail.com>
Date: Wed, 12 Sep 2007 22:14:52 +0800
From: Peili Xu <xupeili@gmail.com>
To: Jerry Yin <jerry.yin@yahoo.com>
Subject: Re: AW: AW: [Sip] Extension of conference procedures
In-Reply-To: <412678.76904.qm@web63212.mail.re1.yahoo.com>
MIME-Version: 1.0
Content-Type: text/plain; charset="ISO-8859-1"
Content-Transfer-Encoding: 7bit
Content-Disposition: inline
References: <46DF05BF.8080007@zonnet.nl> <412678.76904.qm@web63212.mail.re1.yahoo.com>
X-Spam-Score: 0.0 (/)
X-Scan-Signature: 8b30eb7682a596edff707698f4a80f7d
Cc: sip@ietf.org
X-BeenThere: sip@ietf.org
X-Mailman-Version: 2.1.5
Precedence: list
List-Id: Session Initiation Protocol <sip.ietf.org>
List-Unsubscribe: <https://www1.ietf.org/mailman/listinfo/sip>, <mailto:sip-request@ietf.org?subject=unsubscribe>
List-Post: <mailto:sip@ietf.org>
List-Help: <mailto:sip-request@ietf.org?subject=help>
List-Subscribe: <https://www1.ietf.org/mailman/listinfo/sip>, <mailto:sip-request@ietf.org?subject=subscribe>
Errors-To: sip-bounces@ietf.org

Hi Jerry,

please see my comments in line.

2007/9/12, Jerry Yin <jerry.yin@yahoo.com>:
>
> I think there are two ways to invoke a conference. One is to invoke the
> conference by the conference server. The other is ad-hoc conference invoked
> by the participants. The
> draft-ietf-sip-uri-list-conferencing was trying to solve
> the problem by initiating a conference from the server.
> Here's what I think for the ad-hoc conference.
> Participants: A calls B (a UA or a conference room) and put B on-hold, and
> then A calls C. Now A presses the conf button.
> 1. If B has a conference room url, A will transfer C to B (by REFER), as
> some of you discussed already. It actually is supported by some companies
> already as I know.
> 2. But if B is a UA, when the conf button is pressed, the only SIP messages
> send out by A is the re-Invite (off-hold) to B since most SIP phones support
> 3-way conference locally. Then A will do the audio mixing locally. So far I
> didn't find any solution to transfer the local 3-way conference to a
> centralized conference yet. Currently in our system, we adopted the "Join"
> header (RFC3911). When A sends the re-Invite to B, it also includes a Join
> header contains the C's dialog info. The B2B server will translate the Join
> to a centralized conference. It will Invite C with a Replace header to

[Peili] Solution 2 is interesting, but as my interpretation of
RFC3911, it seems join header should only be added in a new INVITE. I
copy the text here "The UAC places the Call-ID, to-tag, and from-tag
information for the target dialog in a single Join  header field and
sends the new INVITE to the target."

> replace the session between A and C. C will sends a BYE to A. The server
> will update the media to A and B (reInvite). Then all three parties are in
> the centralized conference room.
> I hope the new RFC for conference also capture the behavior described in 2.
> Whether it's Join header or something else. The user should be able to call
> someone first and then decided to setup a conference.Jerry
>


_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip