Audio (RTP?) stops 1 minute after bridging

We have a server that simply takes a call from a client via SIP and dials out to a fulfillment center. We have many clients using the system and it works perfectly for all but one. When that client calls the audio drops both directions exactly one minute after the call out to fulfillment is made. Changing fulfillment destination of technology (SIP or PRI) does nothing to change this. The call actually hangs up when the caller or destination hangs up, but will stay up and silent until then. Debugging shows nothing amiss in the dialplan code or the SIP information that is used. It matches up perfectly with calls from other clients that work properly. A TCP Dump and Wireshark analysis shows nothing wrong either, at least not at the SIP header level. I don’t know that I know enough about the RTP flow to determine if anything went wrong at the packet level there.

Has anyone else experienced something like this and if so could you suggest something to resolve it?

Thank you.

Does RTP packets stop in both directions ?
Is there any SIP message during this time ?