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  1. testRTC
  2. qualityRTC: Network Testing
  3. Widget Results

TURN CONNECTIVITY: What we measure

TURN connectivity ensures that communication can still occur even when direct connections are blocked or not possible due to network configurations. The results in the widget indicate the time it takes a connection to be established between the machine being tested and the TURN server. This is particularly important for real-time communication applications like voice calls, video conferencing, and online gaming, where low latency and high reliability are crucial.

Think of TURN connectivity like this

Imagine you want to talk to a friend, but there's a big wall between you two, and you can't go around it. TURN connectivity is like having a helpful friend who can pass messages between you. So, instead of talking directly, you send your message to the helpful friend, who then passes it to your friend behind the wall. Similarly, your friend sends their message to the helpful friend, who then passes it to you. This way, even though you can't talk directly, you can still communicate through the helpful friend, which makes sure your messages get through despite the obstacle. Mind you - that helpful friend can’t really see your messages - he just passes them around, so the privacy of your communication is maintained. In the world of networks, this helpful friend is called a TURN server, and it helps devices communicate when they can't connect directly to each other due to things like firewalls or network setups.

Widget results



The values for UDP, TCP, and TLS which are presented in the widget indicate the time it has taken for a connection to be established between the machine being tested and the TURN server. This number includes the time taken to;

  • Gather the candidate
  • Connect to it
  • Negotiate any necessary certificates
Note: If any of these connectivity checks fail, the numeric round trip time will be replaced by a Red X mark.

Data we collect and share

WebRTC media traffic takes place over UDP as much as possible to reduce the latency and improve media quality. When direct UDP connections aren’t available, we resort to the use of TURN servers and with these servers we can connect WebRTC sessions over UDP, TCP or TLS – as needed for the given scenario.
UDP User Datagram Protocol (UDP) is a communications protocol for time-sensitive applications like gaming, playing videos, or Domain Name System (DNS) lookups. UDP results in speedier communication because it does not spend time forming a firm connection with the destination before transferring the data.
TCP Transmission Control Protocol (TCP) is a standard for delivering data and messages over networks. It's a common protocol used in digital network communications.
TLS

Transport Layer Security (TLS) encrypts data sent over the Internet to ensure that eavesdroppers and hackers are unable to see what you transmit which is particularly useful for private and sensitive information such as passwords, credit card numbers, and personal correspondence.

For the purpose of TURN, TLS isn’t used for encryption (since WebRTC already encrypts the data between the peers) but rather to try and get around firewalls which only allow for TLS encrypted communications to pass through.

Things to notice

  • If UDP is blocked and marked with a red X, this means that your WebRTC sessions either don’t get connected at all or that their quality will be degraded by the use of either TCP or TLS. It is highly recommended that the network you use be open for UDP traffic and configured properly to be reachable for live media exchange.
  • The number is no indication of latency or roundtrip – only on the initial connection time. For actual network latencies it is best to look at other tests conducted for their RTT (round trip time) measurements.

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