Hi everyone!
I have such a infrastructure
Gs wave lite ----- asterisk 1.8 PBX ----- asterisk 13.38 Gateway ----- VoiP provider -------- UAS
When I calling to UAS, PBX plays some announcement to UAS, then Gs wave lite can talking.
Astersik 1.8 PBX starts Early Media and Gs wave lite starts RTP session before it can talk to UAS.
When Announcment ends - Gawe lite received 200 OK and changes SSRC in RTP stream. It causes some failures on Astrerisk 13.38 Gateway and drops few rtp in several seconds (due to ssrc change on Gateway too).
When Im using Yealink VoIP Table Phone, new SSRC is not generated.
So my question is why Gs wave lite changes ssrc in RTP stream when starts talking? Does any way exists to avoid such a behaviour?
Thx for reply. HOPE you will help me.