Re: AW: AW: AW: [Sip] Extension of conference procedures
Paul Kyzivat <pkyzivat@cisco.com> Thu, 13 September 2007 14:00 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 1IVpFE-0007oW-4u; Thu, 13 Sep 2007 10:00:36 -0400
Received: from sip by megatron.ietf.org with local (Exim 4.43) id 1IVpFC-0007oO-Dz for sip-confirm+ok@megatron.ietf.org; Thu, 13 Sep 2007 10:00:34 -0400
Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1IVpFC-0007oG-3M for sip@ietf.org; Thu, 13 Sep 2007 10:00:34 -0400
Received: from rtp-iport-1.cisco.com ([64.102.122.148]) by ietf-mx.ietf.org with esmtp (Exim 4.43) id 1IVpFA-0005Tu-Op for sip@ietf.org; Thu, 13 Sep 2007 10:00:34 -0400
X-IronPort-AV: E=Sophos;i="4.20,250,1186372800"; d="scan'208";a="70950670"
Received: from rtp-dkim-1.cisco.com ([64.102.121.158]) by rtp-iport-1.cisco.com with ESMTP; 13 Sep 2007 10:00:33 -0400
Received: from rtp-core-2.cisco.com (rtp-core-2.cisco.com [64.102.124.13]) by rtp-dkim-1.cisco.com (8.12.11/8.12.11) with ESMTP id l8DE0Wg3024923; Thu, 13 Sep 2007 10:00:32 -0400
Received: from xbh-rtp-201.amer.cisco.com (xbh-rtp-201.cisco.com [64.102.31.12]) by rtp-core-2.cisco.com (8.12.10/8.12.6) with ESMTP id l8DE0SBY026987; Thu, 13 Sep 2007 14:00:28 GMT
Received: from xfe-rtp-202.amer.cisco.com ([64.102.31.21]) by xbh-rtp-201.amer.cisco.com with Microsoft SMTPSVC(6.0.3790.1830); Thu, 13 Sep 2007 10:00:28 -0400
Received: from [161.44.174.118] ([161.44.174.118]) by xfe-rtp-202.amer.cisco.com with Microsoft SMTPSVC(6.0.3790.1830); Thu, 13 Sep 2007 10:00:27 -0400
Message-ID: <46E9427B.7020101@cisco.com>
Date: Thu, 13 Sep 2007 10:00:27 -0400
From: Paul Kyzivat <pkyzivat@cisco.com>
User-Agent: Thunderbird 2.0.0.6 (Windows/20070728)
MIME-Version: 1.0
To: "Huelsemann, Martin" <Martin.Huelsemann@t-com.net>
Subject: Re: AW: AW: AW: [Sip] Extension of conference procedures
References: <CCA850DCD3FBE2479D5076C5C1873222029F5102@S4DE9JSAAHW.ost.t-com.de>
In-Reply-To: <CCA850DCD3FBE2479D5076C5C1873222029F5102@S4DE9JSAAHW.ost.t-com.de>
Content-Type: text/plain; charset="ISO-8859-1"; format="flowed"
Content-Transfer-Encoding: 8bit
X-OriginalArrivalTime: 13 Sep 2007 14:00:27.0555 (UTC) FILETIME=[70307F30:01C7F60E]
DKIM-Signature: v=0.5; a=rsa-sha256; q=dns/txt; l=14505; t=1189692032; x=1190556032; c=relaxed/simple; s=rtpdkim1001; h=Content-Type:From:Subject:Content-Transfer-Encoding:MIME-Version; d=cisco.com; i=pkyzivat@cisco.com; z=From:=20Paul=20Kyzivat=20<pkyzivat@cisco.com> |Subject:=20Re=3A=20AW=3A=20AW=3A=20AW=3A=20[Sip]=20Extension=20of=20conf erence=20procedures |Sender:=20 |To:=20=22Huelsemann,=20Martin=22=20<Martin.Huelsemann@t-com.net>; bh=6yKG9A0ImjbbP7ZZDTzFfHqrccEFt+DWCaBSWEQi7R0=; b=angHj76JG4nM0PMXiCUGRc86hkbPlxZ+SOkaE7zFWotfgkheBlS292AU+PQZt6KxHuZs62hb m5tbSqfG4oHVK0SakN8Bx0cb5hieyqnuVqrUONZSl7oAwci2GCUhH4zo;
Authentication-Results: rtp-dkim-1; header.From=pkyzivat@cisco.com; dkim=pass ( sig from cisco.com/rtpdkim1001 verified; );
X-Spam-Score: -4.0 (----)
X-Scan-Signature: 5ba9b8496764663b12c333825fbf6b3d
Cc: sip@ietf.org, drage@alcatel-lucent.com, alan@sipstation.com, mary.barnes@nortel.com
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
Why not simply REFER the focus to send an INVITE/Replaces to the calls with existing dialogs? Paul Huelsemann, Martin wrote: > Hi all, > > thanks for your interest in this, but I think my question pointed in a > littlebit different direction. > > I think the procedures to start a centralized ad-hoc dial-out conference > are fine, you send an INVITE to the conference factory to start the > conference, and then you send REFERS to the focus referring INVITEs to > the other participants. Even better is an INVITE to the confernce with a > URI list of the participants who shall be invited to the conference, > because this needs lees messages. > > The problem is the INVITE from the focus to the participants. As this is > an ad-hoc conference, it's clear that the invited user is already in a > dialog with the inviting user, and because of this it depends on the > invited user if he accepts the invitation to the conference, or if he > e.g. just answers with a 486. The proposal is to trigger a re-INVITE in > the original dialog by using the "?" mechanism to transfer the dialog ID > of the original dialog to the focus, as this wouldn't cause a 2nd dialog > at the invited UE and therefore means the fewest requirements on the > invited user. > To mee it seems that the Join header cannot solve this problem. If you > include the Join header in the INVITE to the focus, that means you want > to join some dialog at the focus which doesn't exist. > If you include the Join header also using "?" in the header portion in > the Refer-to header of a REFER or in the header protions in the URI > list, this means the focus shall generate an INVITE including a Join > header and sends this to the invited user. The invited user then would > have to start the procedures to include the focus in the original > dialog. If he accepts the 2nd INVITE and is compatible to RFC3911. > Triggering a re-INVITE because of the Join header part in the Refer-to > header to me seems to be not in accordance with RFC3911. > Because of this I don't think the usage of the Join header can solve the > problem of 2 dialogs at the invited user in case of an ad-hoc dial-out > conference. > > BR, Martin > > > > > -----Ursprüngliche Nachricht----- > *Von:* Jerry Yin [mailto:jerry.yin@yahoo.com] > *Gesendet:* Mittwoch, 12. September 2007 22:14 > *An:* Alan Johnston > *Cc:* sip@ietf.org; DRAGE, Keith (Keith); Mary Barnes > *Betreff:* Re: AW: AW: [Sip] Extension of conference procedures > > Hi Alan and Peili, > > Thanks for your responses. I saw Alan's RFC before. I revisited it > today briefly. But I still can't find the solution I am looking for. > Basically in your RFC, even for the ad-hoc conference, the user > always starts with calling the conference server first. What I am > looking for is that the user puts two (or more) lines on hold, then > he decides to conference them together in a conference room. > > The solution I proposed will allow user to have a smooth experience > to conduct a conference in a conference room, if the server supports > this feature. If server does not support it then it will be a local > 3way conference on the phone that most SIP phones support. The UI > design and user experience will be exactly the same in both the cases. > > As to the RFC3911, I agree that it didn't say explicitly that the > Join header can be used in a re-Invite. I also admit that my > proposal does not interpret the Join header as the RFC defines. But > if it works, then we might want to re-interpret the defintion of > Join header? Or we might want to consider introducing a new header > for this purpose! > > To answer your question, if the join does not match an existing > dialog, then according to the RFC 3911, it will be ignored. > > BTW, I saw the RFC 4579 depends on out-dialog REFER. Would it be a > security concern? > > thanks, > Jerry > > > */Alan Johnston <alan@sipstation.com>/* wrote: > > Hi Jerry, > > The use of a Join header field in a re-INVITE in your system is > problematic. RFC 3911 does not seem to specifically rule it out but > implies that it can only be used in a initial (dialog creating) > INVITE. > Also, the error response generation described in RFC 3911 can > not be > applied to a re-INVITE scenario, so I'm curious what you do if > the Join > does not match an existing one. > > There are solutions in RFC 4579 for transitioning conferences to > and > from point to point sessions that may be of use to you. > > Thanks, > Alan > > > Jerry Yin wrote: > > 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 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 > > > > */Jeroen van Bemmel /* wrote: > > > > Concretely, would we be looking at something like > > > > > sip:b-party@provider.com?From=sip%3ca-party@provider.com;tag=x&To=sip:b-party@provider.com;tag=y&Call-ID=i&CSeq=1234&Route=rrr&body= > > with proper session versions etc> > > > > in order to help the conference server fake a reINVITE towards B? > > > > RFC3261 provides some guidance on the types of headers that > elements > > might accept as part of a URI. Specifically, it states in 19.1.5: > > "An implementation SHOULD NOT honor these obviously dangerous > header > > fields: From, Call-ID, CSeq, Via, and Record-Route." > > > > I believe the usage that was foreseen for this mechanism (as > > illustrated > > by some of the examples in RFC3261) was to provide some context > > for the > > request, such as Subject and Priority fields. In other words, > > optional > > information that might help the receiver understand the context. > > > > The above are not different semantics for headers in a URI > (concept > > remains: form a new request based on the URI, inserting the > headers), > > but it does imply a deviation from the basic SIP call model > > (basically a > > way of encoding dialog state in a SIP URI, and sending that to > > another > > element such that it can reconstruct that state and assume the > > role of > > the party which shared the state). > > > > Apart from the fact that this approach will fall short for SDP > > related > > state: is this desirable? > > > > Regards, > > Jeroen > > > > Mary Barnes wrote: > > > RFC 4244 (History-Info) also uses this mechanism to capture the > > Reason > > > and Privacy associated with the URIs that are included as part > > of the > > > History-Info header. My understanding is that it's really > just a > > nifty > > > way to compactly reuse existing headers (i.e., it makes the > > History-Info > > > much more compact as I didn't need to define additional > > parameters for > > > the header, but could rather reuse the existing ones, whose > existing > > > semantics perfectly applicable). I do think that the use of the > > headers > > > that might be escaped using this mechanism should be explained, > > > particularly in cases where you might be extending the use of > > existing > > > headers as I did for the Privacy header. > > > > > > Mary. > > > > > > -----Original Message----- > > > From: Peili Xu [mailto:xupeili@gmail.com] > > > Sent: Wednesday, September 05, 2007 10:41 AM > > > To: DRAGE, Keith (Keith) > > > Cc: sip@ietf.org > > > Subject: Re: AW: AW: [Sip] Extension of conference procedures > > > > > > Yes, It's vague in RFC3261. I'm only aware of the usage in > REFER > > now. > > > It'll be good to clarify the semantics in the usage in > url-list. > > > > > > 2007/9/5, DRAGE, Keith (Keith) : > > > > > >> So this is a convenient way to bring us back to the other half > > of the > > >> > > > issue which we do not seem to have discussed yet. When the > > syntax was > > > defined that allowed ?headers: > > > > > >> Headers: Header fields to be included in a request constructed > > >> from the URI. > > >> > > >> Headers fields in the SIP request can be specified with the > > >> > > > "?" > > > > > >> mechanism within a URI. The header names and values are > > >> encoded in ampersand separated hname = hvalue pairs. The > > >> special hname "body" indicates that the associated hvalue is > > >> the message-body of the SIP request. > > >> > > >> What usage did the SIP WG envisage for this, and thus what > > semantics > > >> > > > did they define for that usage. > > > > > >> Is it appropriate to assign new semantics to such usage? > > >> > > >> Regards > > >> > > >> Keith > > >> > > >> > > > Note: I snipped the rest of this thread as it was getting > really > > LONG. > > > > > > > > > _______________________________________________ > > > 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 > > > > > > > > > > > > _______________________________________________ > > 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 > > > > > > > ------------------------------------------------------------------------ > > Check out > > > > the hottest 2008 models today at Yahoo! Autos. > > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > 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 > > > ------------------------------------------------------------------------ > Be a better Globetrotter. Get better travel answers > <http://us.rd.yahoo.com/evt=48254/*http://answers.yahoo.com/dir/_ylc=X3oDMTI5MGx2aThyBF9TAzIxMTU1MDAzNTIEX3MDMzk2NTQ1MTAzBHNlYwNCQUJwaWxsYXJfTklfMzYwBHNsawNQcm9kdWN0X3F1ZXN0aW9uX3BhZ2U-?link=list&sid=396545469>from > someone who knows. > Yahoo! Answers - Check it out. > > > ------------------------------------------------------------------------ > > _______________________________________________ > 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 _______________________________________________ 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
- [Sip] Extension of conference procedures Alexeitsev, D
- Re: [Sip] Extension of conference procedures Jeroen van Bemmel
- AW: [Sip] Extension of conference procedures Huelsemann, Martin
- Re: AW: [Sip] Extension of conference procedures Jeroen van Bemmel
- Re: AW: [Sip] Extension of conference procedures Dale.Worley
- AW: AW: [Sip] Extension of conference procedures Huelsemann, Martin
- RE: AW: [Sip] Extension of conference procedures Even, Roni
- RE: AW: [Sip] Extension of conference procedures Brian Rosen
- AW: AW: [Sip] Extension of conference procedures Huelsemann, Martin
- RE: AW: [Sip] Extension of conference procedures Even, Roni
- Re: AW: [Sip] Extension of conference procedures Peili Xu
- RE: AW: [Sip] Extension of conference procedures Even, Roni
- RE: AW: [Sip] Extension of conference procedures Peili Xu
- AW: AW: [Sip] Extension of conference procedures Huelsemann, Martin
- Re: AW: AW: [Sip] Extension of conference procedu… Eric Burger
- AW: AW: AW: [Sip] Extension of conference procedu… Huelsemann, Martin
- RE: AW: AW: [Sip] Extension of conference procedu… DRAGE, Keith (Keith)
- Re: AW: AW: [Sip] Extension of conference procedu… Peili Xu
- ?headers ( was RE: AW: AW: [Sip] Extension of con… Mary Barnes
- Re: ?headers ( was RE: AW: AW: [Sip] Extension of… Jeroen van Bemmel
- Re: AW: AW: [Sip] Extension of conference procedu… Dean Willis
- RE: AW: AW: [Sip] Extension of conference procedu… Jerry Yin
- AW: ?headers ( was RE: AW: AW: [Sip] Extension of… Huelsemann, Martin
- Re: AW: AW: [Sip] Extension of conference procedu… Peili Xu
- Re: AW: AW: [Sip] Extension of conference procedu… Alan Johnston
- Re: AW: AW: [Sip] Extension of conference procedu… Jerry Yin
- AW: AW: AW: [Sip] Extension of conference procedu… Huelsemann, Martin
- Re: AW: AW: AW: [Sip] Extension of conference pro… Paul Kyzivat
- AW: AW: AW: AW: [Sip] Extension of conference pro… Huelsemann, Martin
- RE: AW: AW: [Sip] Extension of conference procedu… DRAGE, Keith (Keith)
- Re: AW: AW: [Sip] Extension of conference procedu… Adam Roach
- Re: AW: AW: [Sip] Extension of conference procedu… Paul Kyzivat