RE: [Sip] about H.323 and SIP

Matt Holdrege <matt@ipverse.com> Fri, 21 September 2001 21:04 UTC

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Date: Fri, 21 Sep 2001 13:12:28 -0700
To: De Leo Walter <deleow@telefonica.com.ar>, "'sip@ietf.org'" <sip@ietf.org>
From: Matt Holdrege <matt@ipverse.com>
Subject: RE: [Sip] about H.323 and SIP
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Interworking pure ISUP and pure SIP is implementation specific. To be 
clear, I'm referring to a call setup from an ISUP network to a Softswitch, 
and the Softswitch generating a SIP/SDP setup based on policy. The same 
goes for a call in the other direction.

However, transporting or preserving ISUP information using SIP requires a 
standard, thus we have SIP-T modes of encapsulation and mapping. This is 
the way to go if you do not want the Softswitch to generate the call details.

I hope I'm making myself clear as I'm kind of jet lagged right now.


At 11:15 AM 9/18/2001, De Leo Walter wrote:
>Matt,
>Applying your reasoning to the ISUP-SIP interface, you cannot understand why
>is that interface being defined, and perhaps becoming an standar?
>Do you think it should be left implementation specific?
>I don't.
>Regards.
>Walter.
>
>
> > -----Mensaje original-----
> > De:   Matt Holdrege [SMTP:matt@ipverse.com]
> > Enviado el:   Jueves 13 de Septiembre de 2001 14:04
> > Para: Na Li; sip@ietf.org
> > Asunto:       Re: [Sip] about H.323 and SIP
> >
> > It's perfectly simple in concept that an MGC would speak H.323 on one
> > interface and SIP on another. In fact some MGC's or Softswitches do this
> > today. So if an MGC receives a call setup from an H.323 Gatekeeper which
> > is
> > intended for a SIP network, it would simply send a call setup to the SIP
> > proxy that has reachability for the destination and upon receipt of an RTP
> >
> > address would instruct the H.323 Gatekeeper or gateway to open up an RTP
> > connnection.
> >
> > A similar action would take place if a call was destined for the PSTN and
> > the MGC needed use SIP-T to contact another MGC to setup the call at the
> > remote PSTN network. The far end MGC would speak MGCP or MEGACO/H.248 to
> > control the trunking gateway to the PSTN.
> >
> > Since such protocol interworking occurs internal to a given platform/MGC,
> > there is no standard nor should there be. It is implementation specific as
> >
> > there is no interworking with other elements.
> >
> >
> > At 05:15 PM 9/12/2001, Na Li wrote:
> > >Hello,
> > >
> > >I have several questions relating to H.323
> > >
> > >1) What's the fundamental difference between H.323 (including fast
> > >setup) and SIP? I don't mean the text encoding or ANS.1 encoding.
> > >2) Could Media gateway controller use H.323 for session creation, and
> > >use Megaco to communicate with media gateway?
> > >3) Is any standard there to map H.323 with SIP?
> > >
> > >Thanks, have a nice day
> > >
> > >Lina
> > >
> > >_______________________________________________
> > >Sip mailing list  http://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  http://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  http://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  http://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