5 Apr
2017
5 Apr
'17
4:55 p.m.
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.
Cheers