Subject: [Sip] unified call-control for SIP/PSTN i/w

From: "Info Van" <[email-masked]>
Date: Wed, 26 Dec 2001 13:37:19 -0500

hi,

i have a question for people implementing the SIP/PSTN gateways. and this is not a SIP question in itself, please excuse that. now it's quite likely that you have an integrated media-gateway, being controlled by the Call-Control thru MGCP or any proprietary interface towards the MG, but for SIP to PSTN and PSTN to SIP calls, does the call traverse one "Unified Call-Control" or two separate Call-Controls one for pure SIP to SIP call-control, the other for pure PSTN to PSTN call-control with a SIP-T translator inbetween ?? I see some obvious problems with this. 
1) the MGC portion which is kind-of sharing the same MG from both Call-Controls. both might need to do resource management, take care of race conditions (for same DSP resouce for example).
2) how are the features, say Conference (between SIP and PSTN) with Call Waiting (for SIP subscriber) interwork ?

this brings up a question in my mind, as to does the SIP call model (cc-model) completely encompass the AIN call model, or is the AIN call model completely encompass the SIP call model. If there is such a unified call-model which can encompass both SIP and AIN features (and obviously the more simplistic CLASS features), s.t. it can do :-
1) Generic Call Control (setup, manage, teardown calls between different bearers, s.a. SS7, ISDN, B-ISDN, RTP (managed by SIP))
2) Resource Management (for SS7 bearers, ISDN PRI, BRI, RTP streams)
3) Switching Control
4) Routing (between the various types of end-points or relay agents, s.a. SS7, ISDN, SIP etc.).
would it be able to server the purpose of SIP and PSTN call-control ??
does anyone see any problems with it ??

thanks,
iw.
---
---- Info Vaan is the way to go -------

--
Click here for your very own create-a-date adventure from MatchMaker
Go to http://ecard.matchmaker.com/dating.html

_______________________________________________
Sip mailing list  http://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use [email-masked] for questions on current sip
Use [email-masked] for new developments on the application of sip