Hi,
I have a softswitch that supports SIP and MEGACO (H.248). Normally, when SS7 is on the play, I select SIP for the endpoints (analog gateways, softphones, IP phones etc.) - while all the PSTN interconnections are handled by MEGACO + SigTran (there is a call processor layer in the middle that "normalizes" the SIP, MEGACO and SigTran messages, so everything is compatible). This allows me to completely control the timeslots connected to the traditional carriers. So the MGC can perfectly command the PSTN gateways to select CIC "X" (for example) for an IAM to the PSTN (and vice-versa). I mean: the MGC would send an IAM to the PSTN through the PSTN gateway and the MGC would send a MEGACO command to the PSTN gateway so it would select the proper timeslot. This is what I do today without problems. The issue is: is SIP going to support this level of control? As MEGACO is a master-slave protocol, it's perfectly able to control the PSTN gateway to the timeslot level. But SIP is a peer-to-peer protocol. So it does not offer this level of control. It seems to me that SIP-T is the protocol that would allow the MGC to control the M2k in a way more similar to what we can do with MEGACO today. Is this right? Or do I need to continue using MEGACO + SigTran? In the case of SIP-T, would SigTran still be necessary? Thank you, Paulo A. Borelli Diretor de Produtos Convergentes / Product Director Monytel S.A. +55-11-6838-1006 +55-11-7851-3934 (cel) +55-11-6838-1010 (fax) <BLOCKED::BLOCKED::mailto:[EMAIL PROTECTED]> [EMAIL PROTECTED] <BLOCKED::BLOCKED::http://www.monytel.com.br/> www.monytel.com.br _______________________________________________ Sip-implementors mailing list [email protected] http://lists.cs.columbia.edu/mailman/listinfo/sip-implementors
