Hello!
I'm experiencing an issue with Polycom VVX 411 (UC version 6.3.1.11465 )
Schema of the call using OPUS codec
Some other endpoint -> Asterisk (TLS/SRTP) -> Polycom VVX
Call establishes correctly, but when pressing HOLD on Polycom, on unHOLD I'm getting one-way audio on the phone. This means what I'm saying on some other endpoint can't be heard on Polycom, but everything I'm saying on Polycom is heard by Asterisk/another endpoint. Recording made on Asterisk is shown both streams are reaching Asterisk and (S)RTP is sent towards Polycom.
But this scheme is working another way round.
Polycom VVX -> (TLS/SRTP) Asterisk -> Some other endpoint. If Polycom phone is call originator, hold/unhold works as expected.
On investigating this scenario, I've found, that in the first case, Polycom phone on unhold change RTP payload of Opus from initial 107 sent by Asterisk originally (and accepted in 200 OK by Polycom with same 107 payload) to 121, and Asterisk also changes payload to 121 and seems Polycom is not accepting it anymore.
This problem is not replicated with using PCMA/PCMU codecs.
I've attached 2 pcap's with this case. owa stands for one-way-audio, 2wa - for two-way-audio.