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General question about H.323 and SIP topologies

I'm used to putting together MGCP phone systems. I would like to know a bit more about H323 and SIP design. My first question is; when setting up a simple H.323 phone system are you required to set up an H323 trunk that points to the gateway which will, in turn, have the necessary dial peers to field calls? I have had this set up working before but I'm not sure if it is the only way to do it. For that matter, is the methodology the same for SIP networks (i.e; the need for a SIP trunk)? I have a SIP trunk in my cluster pointed at an IXC for long distance calls, not sure if that's how it works for complex SIP infrustructures.

So, that being the case, if trunks are needed in SIP and H323 set ups, it looks like the call agent (CallManager in this case) is always part of the phone call. I know that in MGCP after the call is set up the call agent steps aside and lets the end points communicate without further involvement. Am I understanding this correctly?



Re: General question about H.323 and SIP topologies

For all protocols, the CM is always in the signaling path. The phones signal via Skinny to the CM, the CM translates to MGCP, H323, or SIP and signals to the far end.

SIP Trunks use a MTP (ie. DSP resource) to extract/insert RFC2833 DTMF from the media path. (DTMF can be sent either mixed in the audio payload or via a dynamic RTP payload [ie. named event])

DTMF is sent via the signaling path with MGCP and H323 so the MTP is not needed.