turnserver | C# .net TURN server -

I have a web app that plays video broadcasted using WebRTC. I'm using a COTURN simple implementation (as described here: https://gabrieltanner.org/blog/turn-server). My turnserver.conf looks like this:

According to RFC5766, it seems that TURN server should have UDP port 49152 - 65535 open for relay.

In all cases, the server SHOULD only allocate ports from the range 49152 - 65535 (the Dynamic and/or Private Port range [Port-Numbers]),

However, some of the TURN servers, such as Coturn, allow you to specify even smaller set of UDP ports for relay.

Because Docker consumes substantial amount of CPU/RAM resources when large port mappings need to be created between a container and a host, I am interested in using smaller set of ports to mitigate high resource usage when I am deploying dockerized TURN server.

I want to know the implications of using smaller set of UDP ports, such as from 49500-50000 (Using 500 UDP ports open) vs 49152-65535 (Using ~16400 UDP ports).

What are the downside of using smaller set of ports for relay? If there is no downside, why does TURN recommend using so many UDP ports for relay?

I am making a WebRTC application which is working fine on wifi. But when I open the app on my phone via 4G the webRTC Livestream won't start. It seems something is going wrong by making the iceConnectionState.

For the Iceserver I made a turn server with coturn on my Nginx server with the following settings:

I have a task to create an audio chat room between 2 users.

I used app example: Peer connection: audio only

I've changed the code to TypeScript, and it works:

Stackblitz

But I can't make it work between two users (not in one tab; I open 2 firefox windows). I have only very basic knowledge of Angular. To be honest, I'm not an Angular developer and normally I write desktop apps, but I need to develop this app with Angular.

I need to send an offer event to a communication platform(to a room with another user - it's always 2 users in the room). It needs a type: offer and sdp.

Other user responds with an answer event to the same platform, to the same room. It also needs sdp, and type set to answer.

We use matrix.org. Here's VOIP specs

I have already implemented sending these events to a communication platform, I read events from the room, and if there's a call offer event, I can answer with an answer event, but I can't make it work.

I've tried to make it work, so I had to work on last weekend, but there's always some issue like DOMException: "Cannot set remote answer in state stable" etc.

Of course I don's ask for a working code(I know it's a work for few days), but I would appreciate any advice to help me implement it, make it easier and possible for someone with very basic knowledge of Angular and webRTC.

Oh, and I'm not sure why, but this stackblitz example doesn't work in Chrome for me, but it works in firefox.

//EDIT

I'm still trying to find a solution. I try to do something like this now:

I have a web and standalone applications that use WebRTC. I have my own coturn server on Amazon EC2. I configure everything based on that article (maybe it was a mistake?). I test my apps with this stun/turn server and everything works just fine for Chrome, Edge, Safari, but does not work for Mozilla Firefox (Offer and Answer were generated). I received the following error in a console:

I have my own server with ubuntu 19.

NOTE: I need a stun/turn server for a video chat web application. It works already if both devices are in the same network.

I have successfully installed and configured coturn. I can successfully start the turnserver by running: turnserver or nohup turnserver & in the terminal.

NOTE: I have an SSH connection to my server.

I also ensured that the necessary ports are open and checked them with netstat -tulpn.

I have been also restarting coturn a few times while testing sudo service coturn restart.

I use these configurations in the turnserver.conf:

RFC 5389 allows one of two forms of authentication for STUN servers:

I have two clients communicating over webrtc. (Client A writen in js, Client B in Python with aiortc). Now it happens that Client A wants to connect from a mobile Network thus it requires a turn-relay connection.

I have already setup a turn server which seems to do his job. But only approx 50% of the connections succeed now. I already found out when they succeed and when they fail:

SDP relay information in case of success:
Offer Client A