Thanks George I appreciate the info . Being able to see what codec is in use for call in progress is very handy sometimes.
As far as the RTP stats goes, I see there is some info with “rtp” and “rtcp” commands which can be useful for troubleshooting. A running tally of # packets or bandwidth used would be awesome in along with the codec in "pjsip show channels" or something like that. Im not certain, but I think the TLS signalling problem from this email may be happening to me again after patching for another pjsip/NAT issue which was with the external_media_address not working and the internal IP being sent in the SDP from asterisk - I applied this patch to the codebase and recompiled I am seeing the TLS “new transport” issue again , I think. Regards, Kevin Long
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