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 and applications.

There are however 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 capabilities, it also recommend that you use a wired network connection.

Beyond this, with WebRTC a common cause of degraded or poor video and audio quality is the utilisation of TCP port tunnelling. Call degradation may also include high latency causing shared content to take a long time to update.

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 (=10,000), however if this port are administratively blocked for web users then the real time media (video and audio) will be forced to tunnel over TCP port 443.

Screen_Shot_2019-09-24_at_10.04.11_am.png

 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/

Was this article helpful?
1 out of 1 found this helpful
Have more questions? Submit a request

Comments

0 comments

Please sign in to leave a comment.