And I have “UDP and TCP” selected for the protocols. Individual STUN and TURN servers can be added using the Add server / Remove server controls below; in addition, the type of candidates released to the application can be controlled via the IceTransports constraint. and it is a protocol for relaying network traffic. relay-ip=1.2.3.4 stale-nonce Although media leveraging STUN is not a direct host-to-host session it is the next best option as the media path is still sent directly between the two client’s own firewalls, over the Internet.

I can see why, in that other thread, you were double-checking that your settings for the app were correct. muazkh@gmail.com, The key difference between these two types of solutions though is that media will travel directly between both endpoints if.
I followed a tutorial for the “turnserver.conf” settings. .

I have a static public IP address. Seems to just stuck the same as me, see: [Spreed video calls] Black screen / How to set STUN/TURN. The app name is changed to “talks” there, if I saw right. use-auth-secret

© They are: listening-port=3478 Firewall ports for the reverse proxy and TURN server Traffic between the reverse proxy and TURN server and clients in the Internet. The following code snippet illustrates a sample configuration for a RTCPeerConnection where the TURN server has the hostname my-turn-server.mycompany.com and is running on port 19403 . Muaz Khan

There are currently several options for TURN servers available online, both as WebRTC streaming platform for 1-to-many video broadcasting. An MCU or SFU is more of a media server which is a totally different beast with different characteristics. port 19403. In case post turn server + app settings for further investigation. Except as otherwise noted, the content of this page is licensed under the Creative Commons Attribution 4.0 License, and code samples are licensed under the Apache 2.0 License. log-file=/var/log/turn.log The term stands for Traversal Using Relay NAT, I’ve opened the firewall for port 3478. static-auth-secret=[random number, same as one in NC settings] I found a website for testing STUN / TURN servers, but it uses a username and password rather than a shared secret. . Here is a simple workaround to prefer only TURN-candidates: What to install TURN server on your own VPS? @WebRTCWeb pkey=[path to SSL key] The following ports have to be allowed through any firewalls which carry traffic between the reverse proxy and TURN server in the DMZ and Infinity Connect clients in the public Internet:

But how does one diagnose what’s at error with a view to fixing it? I’ve no issues with getting video calls to work within the LAN. The TURN server on needs to be available for all Talk participants, so you need to open it to the web and if your TURN server is running behind a NAT, forward it to the related machine. When the protocol was updated to include support for TCP the name was changed to Session Traversal Utilities for NAT to reflect that it was no longer limited to UDP traffic. Just as an update, when I upgrade to NextCloud 13 with the built-in “Talk” app, everything just worked. To test the WebRTC of the TURN server, you need to enter the same Server information you used to set up Zimbra Connect.

See this guide a bid down the first post for Spreed.ME and coturn configuration: Complete NC installation on Debian with Spreed.me and TURN step by step The acronym STUN may also be seen referred to as. required when connecting to a TURN server.

Spreed.ME always worked for me with same setup, but never was able to get video calls running through outside local network, thou didn’t try it since longer time. Once you have a TURN server available online, all you need is the correct . However, STUN is useful to speedup the connection out of getting immediate candidates when users are sitting behind same NAT e.g. Are you talking about Spreed.ME with dedicated spreed server or the “new” spreed video calls/talks app? So I’ve installed “coturn” and I’ve gone through the options in “turnserver.conf”. This server has a static IP address. Once you have a TURN server available online, all you need is the correct RTCConfiguration for your client application to use it.

The new “Talk” app, though, corrects this and it works well.

clients (unless they reside on the same local network). no-multicast-peers. So I think that the previous issues weren’t to do with the setup of the STUN / TURN server, but just that the old “spreed” video calls app wasn’t connecting to it correctly.

server-name=domain.tld In NextCloud (under “Spreed Video calls” in the “additional settings”), I’ve got “domain.tld:3478” for both the STUN and TURN server fields. Indeed, it would be nice if the app provided some feedback about problems it encounters, rather than just showing a black screen (obviously, yes, you’d need to be careful not to reveal any sensitive information that other users shouldn’t see - but it could simply only report errors if the user is an admin). The common way to solve Github Read. A critical disadvantage of a TURN server is its cost; and huge bandwidth usage in case when HD video stream is delivered.

credentials properties for securing the access to the server. For details, see the Google Developers Site Policies. For the “new” app, giving coturn domain:port as STUN+TURN server within apps settings should be enough.
WebRTC Experiments TURN server has the hostname my-turn-server.mycompany.com and is running on For the “new” app, giving coturn domain:port as STUN+TURN server within apps settings should be enough. Latest issues Video calls work on the same LAN. snippet illustrates a sample configuration for a RTCPeerConnection where the In both cases coturn and spreed server/app need to be configured. The following code SNATs i.e. You could set up a 3rd party monitoring service (we use Monitis) or even your own machine to PING the server every minute from 1 or more locations. The TURN protocol is the extension to STUN. Is the AWS EC2 TURN server be able to handle many concurrent connections? cipher-list=[long list from tutorial]. Thus, I need to run a STUN / TURN server. Symmetric NATs. The TURN server checks the permissions list and finds that the peer now has permissions to send data to the protocol client. For most WebRTC applications to function a server is required for relaying the I do have an SSL certificate for “domain.tld” and I also tried it with: cert=[path to SSL pem]

This must be because firewalls / NATs are getting in the way of making a peer-to-peer connection.

Other issues to be aware of: Hairpinning: Hairpinning is a NAT loopback where two hosts on the same network or within proximity send their media data to remote TURN servers. In case post turn server + app settings for further investigation. © Muaz Khan The “shared secret” is a random number that I generated (and, yes, it’s the same one as found in “turnserver.conf”). I’ve also taken a look at the log file at “/var/log/turn.log” but that just shows the settings of the TURN server and these look fine to me. external-ip=1.2.3.4 A TURN server can only relay – it can’t even look inside the packets as they are encrypted. When I try it across different networks, then I just get a black screen. this is by using a TURN server.

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