Re: AW: AW: AW: [Sipping-tispan] Requirements 02e
Paul Kyzivat <pkyzivat@cisco.com> Fri, 30 September 2005 13:41 UTC
Received: from localhost.localdomain ([127.0.0.1] helo=megatron.ietf.org)
by megatron.ietf.org with esmtp (Exim 4.32)
id 1ELL8Y-00009c-1a; Fri, 30 Sep 2005 09:41:18 -0400
Received: from odin.ietf.org ([132.151.1.176] helo=ietf.org)
by megatron.ietf.org with esmtp (Exim 4.32) id 1ELL8V-00009Q-Fa
for sipping-tispan@megatron.ietf.org; Fri, 30 Sep 2005 09:41:16 -0400
Received: from ietf-mx.ietf.org (ietf-mx [132.151.6.1])
by ietf.org (8.9.1a/8.9.1a) with ESMTP id JAA11241
for <sipping-tispan@ietf.org>; Fri, 30 Sep 2005 09:41:14 -0400 (EDT)
Received: from rtp-iport-1.cisco.com ([64.102.122.148])
by ietf-mx.ietf.org with esmtp (Exim 4.43) id 1ELLGN-00065J-9U
for sipping-tispan@ietf.org; Fri, 30 Sep 2005 09:49:24 -0400
Received: from rtp-core-1.cisco.com ([64.102.124.12])
by rtp-iport-1.cisco.com with ESMTP; 30 Sep 2005 06:41:06 -0700
X-BrightmailFiltered: true
X-Brightmail-Tracker: AAAAAA==
X-IronPort-AV: i="3.97,162,1125903600";
d="scan'208"; a="11911354:sNHT27649236"
Received: from xbh-rtp-201.amer.cisco.com (xbh-rtp-201.cisco.com
[64.102.31.12])
by rtp-core-1.cisco.com (8.12.10/8.12.6) with ESMTP id j8UDeqTG026671;
Fri, 30 Sep 2005 09:41:03 -0400 (EDT)
Received: from xfe-rtp-202.amer.cisco.com ([64.102.31.21]) by
xbh-rtp-201.amer.cisco.com with Microsoft SMTPSVC(6.0.3790.211);
Fri, 30 Sep 2005 09:40:56 -0400
Received: from [161.44.79.87] ([161.44.79.87]) by xfe-rtp-202.amer.cisco.com
with Microsoft SMTPSVC(6.0.3790.211);
Fri, 30 Sep 2005 09:40:56 -0400
Message-ID: <433D4067.4000804@cisco.com>
Date: Fri, 30 Sep 2005 09:40:55 -0400
From: Paul Kyzivat <pkyzivat@cisco.com>
User-Agent: Mozilla Thunderbird 1.0.6 (Windows/20050716)
X-Accept-Language: en-us, en
MIME-Version: 1.0
To: "Jesske, R" <R.Jesske@t-com.net>
Subject: Re: AW: AW: AW: [Sipping-tispan] Requirements 02e
References: <E7666D92C64C2845AEF12636FF94F95202319F9C@S4DE8PSAAGQ.blf.telekom.de>
In-Reply-To: <E7666D92C64C2845AEF12636FF94F95202319F9C@S4DE8PSAAGQ.blf.telekom.de>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
X-OriginalArrivalTime: 30 Sep 2005 13:40:56.0130 (UTC)
FILETIME=[956F7A20:01C5C5C4]
X-Spam-Score: 0.0 (/)
X-Scan-Signature: 662324ecda47446db09bfd0a0092c4ba
Content-Transfer-Encoding: quoted-printable
X-MIME-Autoconverted: from 8bit to quoted-printable by ietf.org id JAA11241
Cc: sipping-tispan@ietf.org
X-BeenThere: sipping-tispan@ietf.org
X-Mailman-Version: 2.1.5
Precedence: list
List-Id: Discussion of requirements for SIP introduced by ETSI TISPAN
<sipping-tispan.ietf.org>
List-Unsubscribe: <https://www1.ietf.org/mailman/listinfo/sipping-tispan>,
<mailto:sipping-tispan-request@ietf.org?subject=unsubscribe>
List-Archive: <http://www1.ietf.org/pipermail/sipping-tispan>
List-Post: <mailto:sipping-tispan@ietf.org>
List-Help: <mailto:sipping-tispan-request@ietf.org?subject=help>
List-Subscribe: <https://www1.ietf.org/mailman/listinfo/sipping-tispan>,
<mailto:sipping-tispan-request@ietf.org?subject=subscribe>
Sender: sipping-tispan-bounces@ietf.org
Errors-To: sipping-tispan-bounces@ietf.org
Jesske, R wrote: > Paul, > with regard to the interworking to the PSTN/ PSTN Phones via a Terminal adapter the same procedures as used within the PSTN/ISDN apply. > > The P-Asserted-Id is mapped to the Connected number and the "new P-Additional ID" is mapped to the Generic number (additional Connected number). > > When at the originating Switch a backward message is received with both elements only the user provided (additional connected number) will presented to the user. > The connected number is stored within the exchange. > > We have the same mapping and procedures with the P-Asserted-ID and the >From header. So, what you are telling me is that callerid (and callee identity) is a complete joke - that it has no legitimacy at all! The value displayed *may* be a value authenticated by the provider, and it *may* be a value that was provided by the other end without authentication, and I have no way of discerning which it is. So I must assume that callerid is always a lie. And this is a service you wish to simulate ??? Paul > I hope this clarifies. > > Bets Regards > > Roland > > >>-----Ursprüngliche Nachricht----- >>Von: Paul Kyzivat [mailto:pkyzivat@cisco.com] >>Gesendet: Donnerstag, 29. September 2005 15:05 >>An: Jesske, Roland >>Cc: Miguel.An.Garcia@nokia.com; sipping-tispan@ietf.org >>Betreff: Re: AW: AW: [Sipping-tispan] Requirements 02e >> >> >> >> >>Jesske, R wrote: >> >>>Paul, >>>The callee has the possibility to subscribe to an not >> >>screening option so that the additional identity can be >>included by the subscriber. >> >>>So the final SIP response will include a P-Asserted-id and >> >>an additional identity. >> >>>So the originating user can receive either two identities >> >>or only one. >> >>>It is up to the UA to decide which one shall be presented. >>> >>>With regard to the PSTN/ISDN service this is true. But If I >> >>see the presentation of the From header and the >>P-Asserted-Identity header in forward direction, I think we >>should do the same in backward direction, that both >>identities will be sent to the Caller back. >> >>OK. That relieves some of my concern, since you are leaving >>the decision >>of what to display to the UA, and it is clear to the UA which is >>authenticated and which is not. >> >>But I am still interested in what you expect to happen when this is >>gatewayed to the PSTN, or when the UA is an analog interface >>connected >>to a phone with "callerid box". AFAIK, the callerid box will only >>display one name+number per call and doesn't have any way to indicate >>whether the id it is displaying has been authenticated or >>not. This is >>distressing if the displayed id may sometimes be authenticated and >>sometimes not. >> >>However I don't find the precedent of From vs P-Asserted-ID as >>compelling for doing something similar in the reverse direction. The >> From and To headers are turning out to be useless in >>environments that >>use P-Asserted-ID, and probably should just be ignored. In this >>environment it would make some sense to always populate them >>with Anonymous. >> >> Paul >> >> >>>Roland >>> >>> >>> >>>>-----Ursprüngliche Nachricht----- >>>>Von: Paul Kyzivat [mailto:pkyzivat@cisco.com] >>>>Gesendet: Montag, 26. September 2005 17:02 >>>>An: Jesske, Roland >>>>Cc: Miguel.An.Garcia@nokia.com; sipping-tispan@ietf.org >>>>Betreff: Re: AW: [Sipping-tispan] Requirements 02e >>>> >>>> >>>> >>>> >>>>Jesske, R wrote: >>>> >>>> >>>>>>-----Ursprüngliche Nachricht----- >>>>>>Von: sipping-tispan-bounces@ietf.org >>>>>>[mailto:sipping-tispan-bounces@ietf.org] Im Auftrag von >> >>Paul Kyzivat >> >>>>>>Gesendet: Donnerstag, 22. September 2005 21:09 >>>>>>An: Miguel Garcia >>>>>>Cc: sipping-tispan@ietf.org >>>>>>Betreff: Re: [Sipping-tispan] Requirements 02e >>>>>> >>>>>> >>>>>> >>>>>> >>>>>>Miguel Garcia wrote: >>>>>> >>>>>> >>>>>> >>>>>>>That is my understanding. If I recall correctly, the idea >>>>>> >>>>>>is to emulate >>>>>> >>>>>> >>>>>> >>>>>>>the PBX, where the callee can indicate a direct-to-call >>>>>> >>>>>>number (URI). >>>>>> >>>>>>OK, so the callee can indicate it. What happens at the >>>>>>calling end? Is >>>>>>it displayed instead of the callerid? Is it displayed in >>>>>>addition to the >>>>>>callerid? Is it retained by the calling UA and used in some >>>>>>circumstance? >>>>> >>>>> >>>>>[RJ] This ID is displayed at the callers UA instead of the >>>> >>>>dialled number or the P-Asserted-ID sent from the >> >>terminating network. >> >>>>Which network element makes the decision about which one to >>>>display? Is >>>>it a network element operating on behalf of the callee, one >>>>operating on >>>>behalf of the caller, or is it the caller itself that decides? >>>> >>>> >>>> >>>>>The Reply-to header can only be used in from the Caller so >>>> >>>>it is not possible to use this ID for the callee. >>>> >>>> >>>>>Also the Contact header is not appropriate. >>>>> >>>>>The equivalent in PSTN is the COLP/COLR service. There we >>>> >>>>have a connected and additional connected number. Normally >>>>only the connected number that is a network asserted number >>>>will be shown. But there is also the possibility to allow the >>>>user to present a number sent from his phone directly. (so >>>>called no screening option) So we trust the user that it is a >>>>dialable and valid number. This feature will be used from PBX. >>>> >>>> >>>>>So we want the have the same within SIP. In forward >>>> >>>>direction we have the From header and the >>>>P-Asserted-Identity. But in Backward direction the equivalent >>>>to the From header is missing. >>>> >>>> >>>>>So we need such an identity header for the backward >>>> >>>>direction setup from the callee. >>>> >>>>It sounds like the calling device gets one number, which >>>>could be either >>>> a number that the network can vouch for, or one that the network >>>>cannot vouch for. Does the calling device have indication >> >>about which >> >>>>case holds? If it doesn't trust the number from the callee, >>>>can it get >>>>the asserted id instead? >>>> >>>> Paul >>>> >>>> >>>> >>>>>Roland >>>>> >>>>> >>>>> >>>>> >>>>>>This sounds a lot like the SIP Reply-To header, which can be >>>>>>there and >>>>>>carry information, but which has never had any clear utility, >>>>>>and so has >>>>>>essentially never been used. >>>>>> >>>>>> Paul >>>>>> >>>>>> >>>>>> >>>>>> >>>>>>>/Miguel >>>>>>> >>>>>>>Paul Kyzivat wrote: >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>>>Re TIP: What is the caller supposed to do with this >>>>>> >>>>>>multiplicity of >>>>>> >>>>>> >>>>>> >>>>>>>>addresses? Are we to imagine that the device will present >>>>>> >>>>>>the caller >>>>>> >>>>>> >>>>>> >>>>>>>>with all of them so that he may pick which to use when >>>>>> >>>>>>making a future >>>>>> >>>>>> >>>>>> >>>>>>>>call? >>>>>>>> >>>>>>>> Paul >>>>>>>> >>>>>>>>Miguel Garcia wrote: >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>>>Inline discussion >>>>>>>>> >>>>>>>>>Rocky Wang wrote: >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>>>Hi, Miguel, >>>>>>>>>> >>>>>>>>>> For REQ-TIP-1, I agree with you and it is the same of >>>>>> >>>>>>ours. But >>>>>> >>>>>> >>>>>> >>>>>>>>>>previously I misunderstand it that the direct >>>>>> >>>>>>communication means it >>>>>> >>>>>> >>>>>> >>>>>>>>>>should reach to the same terminal. >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>>Ok, I can add a clarification that there is not a >>>>>> >>>>>>requirement to end >>>>>> >>>>>> >>>>>> >>>>>>>>>up in the same terminal. >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>>> But for REQ-TIP-3, maybe there are still some >>>>>> >>>>>>security problem. >>>>>> >>>>>> >>>>>> >>>>>>>>>>Suppose user B set and his terminal returns a >>>> >>>>subscriber C's URI >>>> >>>> >>>>>>>>>>back to >>>>>>>>>>the calling party A, then what will happen? It will be >>>>>> >>>>>>possible that >>>>>> >>>>>> >>>>>> >>>>>>>>>>the >>>>>>>>>>user A will be misleaded to make a call to C but he >>>>>> >>>>>>wants to make >>>>>> >>>>>> >>>>>> >>>>>>>>>>one to >>>>>>>>>>user B actually. >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>>Shame. As I said, there isn't a requirement for the >>>>>> >>>>>>network to assert >>>>>> >>>>>> >>>>>> >>>>>>>>>the additional identity. This has been clarified a few >>>>>> >>>>>>times in the >>>>>> >>>>>> >>>>>> >>>>>>>>>past, and remember, I am not the owner of the >>>>>> >>>>>>requirements, just the >>>>>> >>>>>> >>>>>> >>>>>>>>>scribe. So from TISPAN point of view, the requirement is >>>>>> >>>>>>as is, and >>>>>> >>>>>> >>>>>> >>>>>>>>>the security implications are accepted. >>>>>>>>> >>>>>>>>>/Miguel >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>>>Best Regards, >>>>>>>>>>Rocky Wang >>>>>>>>>> >>>>>>>>>>-----Original Message----- >>>>>>>>>>From: Miguel Garcia [mailto:Miguel.An.Garcia@nokia.com] Sent: >>>>>>>>>>2005?9?21? 14:37 >>>>>>>>>>To: Rocky Wang >>>>>>>>>>Cc: sipping-tispan@ietf.org >>>>>>>>>>Subject: Re: [Sipping-tispan] Requirements 02e >>>>>>>>>> >>>>>>>>>> >>>>>>>>>>Hi Rocky: >>>>>>>>>> >>>>>>>>>>inline discussion. >>>>>>>>>> >>>>>>>>>>Rocky Wang wrote: >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>>>But I have some more doubts on TIP: >>>>>>>>>>>REQ-TIP-1: In addition to any network asserted >>>>>> >>>>>>identity, it must be >>>>>> >>>>>> >>>>>> >>>>>>>>>>> possible for the callee to indicate in a >>>>>> >>>>>>SIP response an >>>>>> >>>>>> >>>>>> >>>>>>>>>>> additional identity where he is reachable >>>>>> >>>>>>for future >>>>>> >>>>>> >>>>>> >>>>>>>>>>> direct communications. >>>>>>>>>>>Does it mean the back identifier must be something >>>>>> >>>>>>like gruu? The >>>>>> >>>>>> >>>>>> >>>>>>>>>>>one who answers the call from one endpoint does not >>>>>> >>>>>>always mean the >>>>>> >>>>>> >>>>>> >>>>>>>>>>>caller should make a call to this endpoint directly later. >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>>As far as I understand, there is not a requirement for >>>>>> >>>>>>this to be a >>>>>> >>>>>> >>>>>> >>>>>>>>>>GRUU. This is just another URI that would lead to the >>>>>> >>>>>>same person, >>>>>> >>>>>> >>>>>> >>>>>>>>>>but not necessarily to the same terminal. >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>>>REQ-TIP-3: The identity mentioned in REQ-TIP-1 is >>>>>> >>>>>>considered an end >>>>>> >>>>>> >>>>>> >>>>>>>>>>> user supplied information that is not >>>>>> >>>>>>asserted by the >>>>>> >>>>>> >>>>>> >>>>>>>>>>> network. >>>>>>>>>>>Does it mean the network equipments just transfer >> >>the called >> >>>>>>>>>>>party's identifier provided by the called party >>>>>> >>>>>>endpoint? It must >>>>>> >>>>>> >>>>>> >>>>>>>>>>>be also asserted, I think. Or, it will cause some >>>>>> >>>>>>security problem. >>>>>> >>>>>> >>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>>It means that the network does not verify the validity >>>>>> >>>>>>of this URI. >>>>>> >>>>>> >>>>>> >>>>>>>>>>It is up to the user to put something meaningful there. >>>>>> >>>>>>This is the >>>>>> >>>>>> >>>>>> >>>>>>>>>>requirement. >>>>>>>>>> >>>>>>>>>>BR, >>>>>>>>>> >>>>>>>>>> Miguel >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>>>Regards, >>>>>>>>>>>Rocky Wang >>>>>>>>>>> >>>>>>>>>>>-----Original Message----- >>>>>>>>>>>From: Miguel Garcia [mailto:Miguel.An.Garcia@nokia.com] >>>>>>>>>>>Sent: 2005?9?20? 14:46 >>>>>>>>>>>To: Rocky Wang >>>>>>>>>>>Cc: sipping-tispan@ietf.org >>>>>>>>>>>Subject: Re: [Sipping-tispan] Requirements 02e >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>>Hi Rocky: >>>>>>>>>>> >>>>>>>>>>>Yes, I think we agreed that this is a missing >>>>>> >>>>>>requirement, but we want >>>>>> >>>>>> >>>>>> >>>>>>>>>>>to write it down when we have finished the other >>>>>> >>>>>>requirements, because >>>>>> >>>>>> >>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>>>it may happen that the calling part categories >>>>>> >>>>>>requirements help us in >>>>>> >>>>>> >>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>>>creating a proper dependency. >>>>>>>>>>> >>>>>>>>>>>So this is not forgotten, it will be added at a later time. >>>>>>>>>>> >>>>>>>>>>>BR, >>>>>>>>>>> >>>>>>>>>>> Miguel >>>>>>>>>>> >>>>>>>>>>>Rocky Wang wrote: >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>>>Hi, Migues, >>>>>>>>>>>>As I considered, the requirement of overrding ACR >>>>>> >>>>>>service must be >>>>>> >>>>>> >>>>>> >>>>>>>>>>>>there as a formal requirement. Maybe there are some >>>>>> >>>>>>ways that the >>>>>> >>>>>> >>>>>> >>>>>>>>>>>>called AS to get enough information to determine that >>>>>> >>>>>>it can or >>>>>> >>>>>> >>>>>> >>>>>>>>>>>>cannot >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>>>override the ACR, but there are some cases the >>>>>> >>>>>>simulation service >>>>>> >>>>>> >>>>>> >>>>>>>>>>>>must >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>>>be overrided, including the case you listed in the 02a >>>>>> >>>>>>version. It >>>>>> >>>>>> >>>>>> >>>>>>>>>>>>can >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>>>be like this: 1. The originating network shall be able >>>>>> >>>>>>to take enough >>>>>> >>>>>> >>>>>> >>>>>>>>>>>>calling party's information to the terminating network >>>>>> >>>>>>to help to >>>>>> >>>>>> >>>>>> >>>>>>>>>>>>override the callee's ACR service. 2. The >> >>terminating network >> >>>>>>>>>>>>shall be >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>>>able to override the callee's ACR simulation service >>>>>> >>>>>>if the caller is >>>>>> >>>>>> >>>>>> >>>>>>>>>>>>authorized to do this based on the caller's information. >>>>>>>>>>>> >>>>>>>>>>>>Rocky Wang >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>>-----Original Message----- >>>>>>>>>>>>From: sipping-tispan-bounces@ietf.org >>>>>>>>>>>>[mailto:sipping-tispan-bounces@ietf.org] On Behalf Of >>>>>> >>>>>>Miguel Garcia >>>>>> >>>>>> >>>>>> >>>>>>>>>>>>Sent: 2005?9?9? 20:57 >>>>>>>>>>>>To: 'sipping-tispan@ietf.org' >>>>>>>>>>>>Subject: [Sipping-tispan] Requirements 02e >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>>Hi: >>>>>>>>>>>> >>>>>>>>>>>>I have another working copy of the TISPAN >>>>>> >>>>>>requirements, now version >>>>>> >>>>>> >>>>>> >>>>>>>>>>>>02e: >>>>>>>>>>>> >>>>>>>>>>>> >>>>>> >>>>>>http://people.nokia.net/~miguel/drafts/pre/draft-jesske-sippin >>>>>>g-tispan >>>>>> >>>>>> >>>>>> >>>>>>>>>>>>-r >>>>>>>>>>>>equirements-02e.html >>>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>http://people.nokia.net/~miguel/drafts/pre/draft-jesske-sipp >>> >>>ing-tispan >>> >>> >>>>>>>>>>-r >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>>>equirements-02e.txt >>>>>>>>>>> >>>>>>>>>>>And a diff version with respect 02d: >>>>>>>>>>> >>>>> >>>>>http://people.nokia.net/~miguel/drafts/pre/draft-jesske-sippin >>>> >>>>g-tispan >>>> >>>> >>>> >>>>>>>>>>-r >>>>>>>>>>equirements-02d-to-e.html >>>>>>>>>> >>>>>>>>>>The main changes this time are: >>>>>>>>>> >>>>>>>>>>- Rewording of the CCBS/CCNR requirements as per our >> >>discussion. I >> >>>>>>>>>>have >>>>>>>>>>added a few definitions and try to reorder them >> >>sequentially, so >> >>>>>>>>>>that they are a bit easier to understand. >>>>>>>>>>- Clarification that the TIP requirements are for an >> >>additional (to >> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>>the >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>>>asserted) identity. >>>>>>>>>>- Addition of the Communication Waiting requirements >>>>>>>>>> >>>>>>>>>>Please feel free to comment, but bear in mind that due to the >>>>>>>>>>TISPAN meeting next week most of the TISPANers will >> >>have limited >> >>>>>>>>>>time to discuss. >>>>>>>>>> >>>>>>>>>>/Miguel >>>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>_______________________________________________ >>>>Sipping-tispan mailing list >>>>Sipping-tispan@ietf.org >>>>https://www1.ietf.org/mailman/listinfo/sipping-tispan >>>> >>> >>> > _______________________________________________ Sipping-tispan mailing list Sipping-tispan@ietf.org https://www1.ietf.org/mailman/listinfo/sipping-tispan
- AW: AW: AW: [Sipping-tispan] Requirements 02e Jesske, R
- Re: AW: AW: AW: [Sipping-tispan] Requirements 02e Paul Kyzivat
- Re: AW: AW: AW: [Sipping-tispan] Requirements 02e Tom-PT Taylor
- RE: AW: AW: AW: [Sipping-tispan] Requirements 02e GARCIN Sebastien RD-CORE-ISS
- Re: AW: AW: AW: [Sipping-tispan] Requirements 02e Paul Kyzivat