How to improve the stability of WebRTC, Asterisk?

Question

Asterisk

Colleagues, welcome.

JsSIP + Asterisk

I'm asking a question for my colleagues who are responsible for Asterisk and telephony in general.
How to improve the stability of WebRTC, Asterisk?

The problem is that some users of our system do not hear the interlocutor (no sound).
Basically, the problem is observed in the countries of the former CIS.

A VPN helps sometimes, but not for long.

Is it worth thinking about raising an asterisk separately in these countries?
Or what to do in general?

Answer 1

Basically, the problem is observed in the countries of the former CIS.

mobile providers can block both webrtc and many other things, since these are p2p protocols and it’s difficult for them to make / buy smart dpi machines.
Also, by default, opera can block webrtc traffic for security reasons (this can be turned on) part or only incoming
Correctly said, white ip and without NAT can help a lot.
ps what do you mean vpn helps but not really?