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How to test Co-Turn TURN server configurations?

Updated: Jul 10, 2023

A TURN server is a need for any WebRTC application when a user behind a strict firewall needs to connect to your application. Co-Turn is a very popular open-source TURN server that can be used with any WebRTC application for bypassing firewalls. It can be installed on a Linux server(on-premise / cloud ) and need to be properly configured to work with your WebRTC application. As the scope of this post, we are going to focus only on how to test your TURN server to find out if all the configurations are done properly or not as the configuration of a TURN server deserves a separate post.

The most popular tool available today to test a TURN server is known as the WebRTC trickle ICE testing tool. The link to the site is at the bottom of this post. Here is what it looks like.

As shown, we can add our STUN/TURN server credentials using the add server button. Once the server is added, we can select the server and gather all possible ICE candidates using the gather candidates button. Here is what it looks like for when a generate candidates for working a TURN server.

As shown, there are 3 kinds of candidates being generated from one candidate gathering request. They are

  • host candidates

  • srflx candidates

  • relay candidates

host candidates: Host candidates are those candidates which can be used to connect a WebRTC call within the same network, i.e. both the peers joining the call are connected to one network router / switch.

srflx candidates: srflx candidates are those candidates which can be used to connect a WebRTC call when both the users are not in the same network but on different networks may be even in 2 different continents connected by the Internet.

relay candidates: relay candidates are those candidates which can be used to connect a WebRTC call when both users are not in the same network and either one user is or both the users are, behind a firewall. When a user is behind a firewall, he/she is not reachable directly from the Internet. That's why the firewall has been put in place so that a network can be isolated from the Internet and interaction with the Internet can be controlled by a network administrator in a desired manner. The primary reason is to keep the evil eyes out of the network.But this network arrangement poses a problem to a WebRTC app as it can't reach to a user. In order to solve this, a technique is used where a TURN server is put in place to let both the users use it as a relay server to relay their media through the server rather than directly connecting with each other, in order to bypass the firewall. If this sounds a bit confusing, you can read more about firewalls and trickle ICE to gain a better understanding.

There can be 3 kinds of relay candidates. Any one of these candidate types can be chosen to establish the connection.

  • relay-udp

  • relay-tcp

  • relay-tls

relay-udp: The most common kind of relay candidates and the easiest to connect one. When the firewall is not very strict and allows udp connection to happen, then this type is chosen.

relay-tcp: This is the next best available option when the relay-udp candidate type is not possible as udp ports are blocked by the firewall. In this case the relay-tcp candidate type is used.

relay-tls: This is the next best available option as the firewall has not only blocked the udp ports but also blocked unsecured connections over tcp. Only a secured connection with a valid authentication mechanism like ssl certificates can pass through this firewall. In this case, the relay-tls candidate type is used which has ssl certificate based authentication mechanism already enabled at the TURN server end.

To test which kind of relay candidate a TURN server is using, we can use the Firefox browser with a special setting flag switched on. Here is how it looks like.

As shown, there is a flag in Firefox to tell the browser to use only relay type connections for any kind of WebRTC application used in the browser. By default, it is set to false which means only relay type connection is not enforced for WebRTC calls. This flag can be switched on to enforce the relay only candidate generation by clicking on the button provided at the right most side of the flag. This is how it will look like after switching it on.

Any WebRTC application, using Firefox after this flag is switched on will use relay type candidates only. Here is how the candidate logs of Firefox looks like while a WebRTCcall is running after this flag is switched on.

As it can be seen in the above image, Firefox has chosen relay-tls candidate type in this case as the only turn server credentials provided for this example is enabled with ssl certificate based authentication. More than one turn server credentials can also be provided so that the TURN server can take a decision to choose the correct relay candidate type based on the users firewall configuration. An ideal iceServers parameter for establishing a WebRTCconnection should have a STUN server and multiple turn server entries different purposes. It should something like this.

As it can be seen there are 4 entries in this case. The explanation for each entry is as below.

Entry 1: For generating srflx candidates using a STUN server when both the users are not behind a firewall. This is the most common and most used method to to connect a WebRTC call. A STUN server is a must for connecting a WebRTC call. Also there are no costs related maintaining a STUN server in terms of data transfer costs except for the cost of the server itself.

Entry 2: For generating relay candidates as there is a firewall in place for either one or both the users of the WebRTC application. Here the turn server is reachable on port 80 because most of the firewalls allow access on port 80 which also is used by http.

Entry 3: For generating relay candidates as there is a firewall in place and all access are blocked by it except authentication based access. For this a ssl certificate based authentication is used to pass through this firewall restriction. This firewall is stricter than the previous one. Hence port 443 is used which also is used by https.

Entry 4: For generating relay candidates of type tcp as there is a strictest possible firewall in place which not only needs a authentication based access but also has blocked udp ports.

This is even a stricter firewall than the previous one. This type of candidate should be able to pass through al kind of firewalls as All most all firewalls allow port 443 access for normal internet access to websites and without opening port 443, they won't work!

While working with one of our customers, we realized that testing the TURN server settings one by one for all possible candidate types is critical for any application for reducing call failures due to strict firewall policies. Except for trickle ICE, there are no reliable tools available today to test TURN servers primarily from the perspective of what kind of relay candidates(udp/tcp/tls) this turn server supports. Therefore, you can use our open-source p2p application to test it in your local network or deploy the application to test it over the internet. We will try to have a hosted version of the same p2p application in some time.

How does it work?

Step 1: Download this repo to your local machine and follow clearly mentioned instructions in the read me to set it up. Don't forget to provide your TURN server credentials you wish to test as mentioned in the read me.

Step 2: Once the setup is successful, open Firefox, change the settings as mentioned above switch on the relay-only flag.

Step 3: Copy and paste the generated link either on a different browser on the same device or on a different device in the same network. You can share the link with your friend as well to hang out with him while testing if you choose to host the application in a cloud provider like DigitalOcean.

Step 4: Once the other user joins the link, press the call button to start the video chat.

Step 5: Open the about:webrtc in a different tab in Firefox, clear the history using the Clear History button so that the statistics are available for the current call only and open the show details and then click on the show details link to view the detailed call statistics including the ice candidate details. Here you should be able to view the relay candidate type(udp/tcp/tls) along with all other relevant details.

Hope this helps you in testing your turn server for misconfigurations. Feel free to drop us an email at for any questions/doubts/concerns related to the above post or overall TURN servers/ WebRTC.



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