In call manager, as I understand if a CUCM SCCP phone calls a PSTN phone via an E1 / gateway then the signaling is through the CUCM but the RTP media is between the phone and the gateway.

If the PSTN connection is via a SIP trunk, does the communications method still take the same path? I think it must do for two reasons. The first is that the phone might not have been directly registered with the SIP provider, secondly the gateway's job is to be a gateway. I ran wireshark from my soft phone and all I can say is that during a SCCP call to a SIP trunk there is no direct coms but lots of VPN traffic. Since I do not have my SIP providers addresses in the encryption domain / networks I can assume the destination on the RTP packets is the CUCME.

Since technically, RTP packets could bypass the CUCME or gateways, I'm assuming if this were the case there would be one way or no audio.

Are my assumptions correct?

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"Right actions in the future are the best apologies for bad actions in the past."

I realized the first thing doing a packet capture.

Why don't try that with the second thing?

I'm in Australia and my lab is in England. It's a bit tricky

I read about media flow-through and flow around. Not sure if it can be applied here, so will do some reading soon.

Thanks for the reply

_________________
"Right actions in the future are the best apologies for bad actions in the past."