Thanks for your reply
Yes all calls performed through my own Asterisk server.
All SIP agents are registered on the Asterisk server directly.
Between all SIP agents, like Microsip or Linphone - all calls are successful.
The problem is only with this Android application.
I create APK from source github code and since this is flutter, I can create app for Google Chrome or Windows or IOS.
After creation, i must input
websocket address like
wss://my server ip:8089/ws
SIP URI like
9004@my server ip
Auth user like
auth pass like
and register on the server. I can see the registration in the server logs, all right. And after that I call from Linphone to Google Chrome app with packetization-mode=0 or packetization-mode=1 or without this parameter at all, no matter - everything OK - audio/video OK.
But, after creation this same app on Android, and doing the same steps as above, everything OK only when parameter packetization-mode=1.
If some SIP agent doesn’t have this parameter or packetization-mode=0 ( Like my intercom, for example, which can be connected to an asterisk and which use`s a pure codec H264 without any parameters and cannot be changed)
(piece of SDP packet from my Intercom)
In this case - call is dropped immediately.
I need to understand where the problem could be, in the Asterisk server or in the application itself
externip=my external IP
[ 8002 ] (basic,phones)
[ 9004 ] (basic,webrtc)