Tunnelled Media and Call Degradation

WebRTC is an innovative video technology that is portable and does not require the installation of specialised applications to provide video conference capabilities through the web browser.

There are some important points to be aware of for the best possible experience.

Like other real time video technologies it is recommend that you choose a suitable setting for bandwidth according to your internet connection speeds, it also recommend that you use a wired network connection.

With WebRTC a common cause for degraded or poor video and audio quality is the utilisation of TCP port tunnelling. Call degradation may also include high latency on the updating of shared content. Port restriction and forced tunnelling is common in many web proxy environments.

With WebRTC during the call set up the browser will attempt to send the media using UDP over ephemeral port ranges (=10,000), however if these ports are administratively blocked for web users then the real time media (video and audio) will be forced to tunnel over TCP port 443.


 In the scenario where video is tunnelled there is a greater likelihood of packet loss, jitter and/or latency and resulting call degradation. The preferred method for transport of real time media is via UDP due to the transport efficiency and lower processing overheads.

To assist in establishing if your environment supports UDP please see the Network Test at the following WebRTC test site: https://test.webrtc.org/

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