Sometimes there is no audio and video. I am using pjsip.
Far too little information to be useful. Lack of media is often the result of firewall or NAT misconfiguration.
Asterisk server is behind the NAT. My pjsip configuration file:
pjsip.txt (2.4 KB)
You are missing signalling and media addresses, and local networks.
Asterisk is a client, when making outgoing calls.
Attempt 3 to send STUN request to ‘x.x.x.x’ timed out. How to resolve this errror. In the firewall port 3478 is opened.
I’ve given
local_net=10.0.0.0/255.0.0.0
external_media_address=
external_signaling_address=
still no audio and video.
Do we need a TURN server when using asterisk?
No, and generally if properly configured (in the case of WebRTC/ICE you configure things in rtp.conf) you don’t need STUN configured on the Asterisk side either.
Do we need to allow port 5060 in firewall? It is allowed in vm. UDP Ports 10000-11000 opened in firewall both inbound and outbound. 443 is also allowed. Registration is successful. But there is no audio and video on both sides.
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