On 04/04/2017 09:33 PM, Anthony Joseph Messina wrote:
After more digging, I see (from the Asterisk perspective) that after a certain amount of time, the "RTCP report" size gets smaller and this is the point at which the audio from Asterisk back to the softphone is dropped. Again, this audio drop occurred around 19 minutes into the call. I'm not sure this means anything, but perhaps it can point someone more knowledgeable in the right direction.
A good place to start is to inspect /proc/rtpengine/0/list and check the packet and byte counters for the respective local ports. This way you can check if incoming packets are actually arriving at rtpengine.
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