Environment: Cisco UCM 7.1, SIP trunking to PSTN via XO Communications. No shortage of bandwidth. Latency etc all in spec.
On some inbound calls my users complain that they pick up the call and there is no audio for some 4-5 seconds. I pulled call traces and CDR but I don't see anything that helps me identify the delay in the start of the audio stream. The call is coming in the correct gateway. The MOS scores shown in destination CMR look ok:
First check if there is SDP in the INVITE or not (early offer). If there isn't check when the phone answers the call in the CCM trace, and compare that to how long it takes for the 200 OK to go out, and then how long does it take for the ACK to come back in. At that point we should be just about ready to have audio flowing, we just need to send an open receive channel and start media transmission to the phone (if SCCP) or ACK to the phone if it's SIP. If all those times look fine, I would take a packet capture from the call manager server the IP phone is registered to and the IP phone to see if there is any network delay between CUCM sending out packets and them arriving at the phone.
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