Hi all experts I am struggling with an simplex audio issue when using SRTP. Normal call setup is working fine. Remote re-invite (from SBC/server) works ok. (HOLD/UNHOLD) However, if I let the call have a duration of 20 minutes, and then receiving a reinvite it causes simplex audio. The uplink srtp stream is not accepted by the SBC/server. Testing with other clients, the scenario works fine. I will try to understand the srtp code, but I do wonder if someone knows how the key exchange mechanism works. Do PJSIP have any mechanism that changes behaviour after 20 minutes? Does the client change the upstream key up on reinvite? Is this a known issue? cheers, lars