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Frequently Asked Questions


These are very important concepts that developers must understand well to work with WebRTC. Here is a collection of all Kurento material talking about these acronyms:

When are STUN and TURN needed?

STUN (and possibly TURN) is needed for every WebRTC participant behind a NAT. All peers that try to connect from behind a NAT will need to auto-discover their own external IP address, and also open up ports for RTP data transmission, a process that is known as NAT Traversal. This is achieved by using a STUN server which must be deployed outside of the NAT.

The STUN server uses a single port for client connections (3478 by default), so this port should be opened up for the public in the server’s network configuration or Security Group. If using TURN relay, then the whole range of TURN ports (49152 to 65535 by default) should be opened up too, besides the client port. Depending on the features of the STUN/TURN server, these might be only UDP or both UDP and TCP ports. For example, Coturn uses both UDP and TCP in its default configuration.

If you are installing Kurento in a NAT environment (e.g. if your media server is behind a NAT firewall), you also need to configure an external STUN server, in /etc/kurento/modules/kurento/WebRtcEndpoint.conf.ini (check How to configure STUN/TURN in Kurento? for more details). Similarly, all browser clients that are behind a NAT need to use the STUN server through the iceServers field of the RTCPeerConnection constructor.


Kurento Media Server and its Application Server are running in a cloud machine without any NAT or port restriction on incoming connections, while a browser client runs from a possibly restricted NAT network that forbids incoming connections on any port that hasn’t been “opened” in advance

The browser client may communicate with the Application Server for signaling purposes, but at the end of the day the bulk of the audio/video RTP transmission is done between the WebRTC engines of the browser and KMS.

NAT client without STUN

In scenarios like this, the client is able to send data to KMS because its NAT will allow outgoing packets. However, KMS will not be able to send data to the client, because the client’s NAT is closed for incoming packets. This is solved by configuring the client to use a STUN server; this server will be used by the client’s browser to open the appropriate ports in its own NAT. After this operation, the client is now able to receive audio/video streams from KMS:

NAT client with STUN

This procedure is done by the ICE implementation of the client’s browser.

Note that you can also deploy KMS behind a NAT firewall, as long as KMS itself is also configured to use a STUN server.

Further reading:

How does TURN work?

This is a very simplified explanation of TURN; for the complete details on how it works, read the RFC 8656 (Traversal Using Relays around NAT (TURN)).

TURN separates two network segments that cannot connect directly (otherwise, STUN and direct connections would be used). In order to allow for maximum probabilities of successful connections, TURN servers such as Coturn will enable both UDP and TCP protocols by default.

  • When a WebRTC participant is behind a strict NAT or firewall that requires relay, it becomes a TURN client, contacting the TURN server on its client listening port (3478 by default, either UDP or TCP), and requesting a TURN relay transport.
    • The TURN server listens for client requests on both UDP and TCP ports, to maximize the chances that the client’s firewall will allow the connection.
    • The TURN relay transport, mentioned above, is a random port selected on the TURN port range of the TURN server. This range, again, can be either UDP or TCP, to maximize the chances that remote peers are also able to send RTP data to the server.
  • When a remote WebRTC peer wants to send RTP data to the TURN client, it doesn’t send to it directly, instead it sends data towards the corresponding TURN relay transport of the TURN server. Then the server will relay this data through its client port (3478) towards the actual TURN client.

How to install Coturn?

Coturn is a STUN server and TURN relay, supporting all features required for the ICE protocol and allowing to establish WebRTC connections from behind a NAT.

Coturn can be installed directly from the Ubuntu package repositories:

sudo apt-get update && sudo apt-get install --no-install-recommends --yes \

To configure it for WebRTC, follow these steps:

  1. Edit /etc/turnserver.conf.

    This example configuration is a good baseline; it will work for using Coturn with Kurento Media Server for WebRTC streams. However, you may want to change it according to your needs:

    # The external IP address of this server, if Coturn is behind a NAT.
    # It must be an IP address, not a domain name.
    # STUN listener port for UDP and TCP.
    # Default: 3478.
    # TURN lower and upper bounds of the UDP relay ports.
    # Default: 49152, 65535.
    # Uncomment to run server in 'normal' 'moderate' verbose mode.
    # Default: verbose mode OFF.
    # TURN fingerprints in messages.
    # TURN long-term credential mechanism.
    # TURN realm used for the long-term credential mechanism.
    # TURN static user account for long-term credential mechanism.
    # Set the log file name.
    # The log file can be reset sending a SIGHUP signal to the turnserver process.
    # Disable log file rollover and use log file name as-is.


    • The external-ip is necessary in cloud providers which use internal NATs, such as Amazon EC2 (AWS). Write your server’s public IP address, like, in the <CoturnIp> parameter. It must be an IP address, not a domain name.
    • Comment out all the TURN parameters if you only want Coturn acting as a STUN server.
    • The user parameter is the most basic form of authorization to use the TURN relay capabilities. Write your desired user name and password in the fields <TurnUser> and <TurnPassword>.
    • Other parameters can be tuned as needed. For more information, check the Coturn help pages:
  2. Edit the file /etc/default/coturn and set


    so the server starts automatically as a system service daemon.

  3. Follow with the next sections to test that Coturn is working, and then set it up as your STUN/TURN server in both Kurento Media Server and the WebRTC clients.

How to test my STUN/TURN server?

To test if your STUN/TURN server is functioning properly, open the Trickle ICE test page. In that page, follow these steps:

  1. Remove any server that might be filled in already by default.

  2. Fill in your STUN/TURN server details.

    • To only test STUN (TURN relay will not be tested):

    • To test both STUN and TURN:


      … and also fill in the TURN username and TURN password.

  3. Click on Add Server. You should have only one entry in the list, with your server details.

  4. Click on Gather candidates. Verify that you get candidates of type srflx if you are testing STUN. Likewise, you should get candidates of type srflx and type relay if you are testing TURN.

    If you are missing any of the expected candidate types, your STUN/TURN server is not working well and WebRTC will fail. Check your server configuration, and your cloud provider’s network settings.

How to configure STUN/TURN in Kurento?

To configure a STUN server or TURN relay with Kurento Media Server, you may use either of two methods:

  1. Write the parameters into the file /etc/kurento/modules/kurento/WebRtcEndpoint.conf.ini. Do this if your settings are static and you know them beforehand.

    To only use STUN server (TURN relay will not be used):


    <StunServerIp> should be the public IP address of the STUN server. It must be an IP address, not a domain name. For example:


    To use both STUN server and TURN relay:


    <TurnServerIp> should be the public IP address of the TURN relay. It must be an IP address, not a domain name. For example:

  2. Use the API methods to set the parameters dynamically. Do this if your STUN server details are not known beforehand, or if your TURN credentials are generated on runtime:

    To only use STUN server (TURN relay will not be used):


    Kurento Client API docs: Java, JavaScript.

    To use both STUN server and TURN relay:


    Kurento Client API docs: Java, JavaScript.


You don’t need to configure both STUN and TURN, because TURN already includes STUN functionality.

The following ports should be open in the firewall or your cloud provider Security Group:

  • <CoturnPort> (Default: 3478) UDP & TCP, unless you disable either UDP or TCP in Coturn (for example, with no-tcp).
  • 49152 to 65535 UDP & TCP: As per RFC 8656, this port range will be used by a TURN relay to exchange media by default. These ports can be changed using Coturn’s min-port and max-port parameters. Again, you can disable using either TCP or UDP for the relay port range (for example, with no-tcp-relay).


Port ranges do NOT need to match between Coturn and Kurento Media Server.

If you happen to deploy both Coturn and KMS in the same machine, we recommend that their port ranges do not overlap.

When you are done, (re)start both Coturn and Kurento servers:

sudo service coturn restart
sudo service kurento-media-server restart

How many Media Pipelines do I need for my Application?

A Pipeline is a top-level container that handles every resource that should be able to achieve any kind of interaction with each other. Media Elements can only communicate when they are part of the same Pipeline. Different Pipelines in the server are independent, so they do not share audio, video, data or events.

99% times, this translates to using 1 Pipeline object for each “room”-like videoconference. It doesn’t matter if there is 1 single presenter and N viewers (“one-to-many”), or if there are N participants Skype-style (“many-to-many”), all of them are managed by the same Pipeline. So, most actual real-world applications would only ever create 1 Pipeline, because that’s good enough for most needs.

A good heuristic is that you will need one Pipeline per each set of communicating partners in a channel, and one Endpoint in this Pipeline per audio/video streams exchanged with a participant.

How many Endpoints do I need?

Your application will need to create at least one Endpoint for each media stream flowing to (or from) each participant. You might actually need more, if the streams are to be recorded or if streams are being duplicated for other purposes.

Which participant corresponds to which Endpoint?

The Kurento API offers no way to get application-level semantic attributes stored in a Media Element. However, the application developer can maintain a HashMap or equivalent data structure, storing the Endpoint identifiers (which are plain strings) to whatever application information is desired, such as the names of the participants.