RE: [Sip] Eating our own Dog Food...could the IAB and IESG use SIP forconference calls

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>So "pure Internet SIP" won't work for all of us any time soon.

Glad to clear up the confusion on this point. People on the PSTN can
dial in and can be called from the SIP conferencing server by using a
service provider that has standard PSTN-SIP gateways. The typical SIP
voice conference has both PSTN users and SIP users. Works quite well for
everybody.

IM can also be used at the same time, by those who prefer it to real
time voice. Several SIP clients do both, actually video and data as
well.

Thanks, Henry

> -----Original Message-----
> From: sip-admin@ietf.org [mailto:sip-admin@ietf.org] On 
> Behalf Of Harald Tveit Alvestrand
> Sent: Tuesday, March 25, 2003 1:14 PM
> To: Richard Shockey; ietf@ietf.org
> Cc: sip@ietf.org
> Subject: Re: [Sip] Eating our own Dog Food...could the IAB 
> and IESG use SIP for conference calls
> 
> 
> Thanks for the idea!
> injecting a slight dash of cold water....
> 
> - the actual cost of "Teleconferences and Long Distance 
> charges" in 2002 
> were USD 82.210 (unaudited) (vs 54.400 for 2001). A 
> significant fraction of 
> that, but far from all, is the IESG teleconferences.
> - we've already switched teleconference providers once to 
> reduce costs, 
> going from call-everyone to most-people-call-in.
> - the costliest part of the IESG teleconferences has been the 
> callout: 
> international participants (last year, France, the 
> Netherlands, Norway and 
> Sweden when they are at home, hotels literally anywhere in 
> the world when 
> they are travelling) are called rather than calling in. You 
> don't want to 
> discuss with your boss why you had to make a 2 1/2 hour 
> international call 
> at hotel room rates if you can avoid it......
> - it's absolutely essential that one be able to participate 
> in the IESG 
> telecon from just about anything that one can dial from - we've had 
> participants on cellphones from trains, in airport lounges, 
> hotel rooms and 
> other places. So SIP-only won't work for some time yet.
> - even at work, several of us have problems with firewalls; 
> about half the 
> IESG uses Jabber during sessions - the reason many of the 
> others don't is 
> that they can't get Jabber through their corporate firewalls. 
> So "pure 
> Internet SIP" won't work for all of us any time soon.
> 
> So I think this is a good idea, PROVIDED THAT:
> 
> - The SIP teleconference bridge provider is able to provide either 
> 800-number access or callout services to normal telephones in 
> most corners 
> of the world
> - The voice quality, operator quality and call stability is 
> competitive 
> (yes, we've got the "there's an echo on this conference - can 
> you figure 
> out who is echoing and fix it" request down to a matter of routine)
> 
> A "normal" teleconference provider that *also* allows SIP 
> dialin over the 
> Internet would probably be perfect. If you have one - send 
> email to me - 
> PRIVATELY - and I will forward to the relevant parties at the 
> secretariat.
> 
>                Harald
> 
> --On mandag, mars 24, 2003 19:04:17 -0500 Richard Shockey 
> <richard@shockey.us> wrote:
> 
> >
> > Like many of us I was moved my Harald's appeal for suggestions for 
> > helping to cut down costs in the IETF.
> >
> > I certainly endorse the idea of considering Canada or Mexico as 
> > possible sites for future IETF meetings, but I suspect that 
> the weekly 
> > teleconferece calls that the IAB and IESG have represent a 
> significant 
> > line item for the Secretariat.
> >
> > In case anyone has not heard, SIP is quite capable of handling this 
> > type of task and there are a variety of commercial as well as open 
> > source Client User Agents as well as commercial products 
> and services 
> > that could help reduce this cost.
> >
> > I'm sure the SIP working group could help the Secretariat identify 
> > products and services that could make this essential function more 
> > productive and operate at less cost.
> 
> 
> _______________________________________________
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> 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
> 





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