RTT indicates the round trip time (latency) observed. The higher the RTT, the lower the overall media quality of the session is. testRTC shows RTT information at the top ribbon results across the various solutions available for testing and monitoring:
The RTT values shown indicate the outgoing round trip time in milliseconds and are split between audio and video channels. The data is averaged throughout the duration of the session.
When the round trip time value is above a certain threshold value, it will be marked in red.
The default configuration is set to 250 milliseconds. This threshold can be changed through our support.
Things you should know about RTT
Here are a few things to remember in the back of your mind about round trip time:
- Delay, latency and RTT. All these come to describe the time it takes to receive media packets. Delay and latency are one-way and round trip time is two-way in nature (and what is measured by WebRTC)
- RTT is measured only on outgoing channels. The receiver of the outgoing media sends feedback that is then used to measure the round trip time
- The lower the round trip the better the media quality will be – remember that what we are aiming for is real-time and conversational in nature
- The round trip time measured is the network one – it isn’t glass-to-glass. Additional delays caused by the camera and display processing are left out of the measurements done here
- High RTT may indicate any of the following:
- Connecting to a peer located remotely
- Connecting to a media server located remotely (when a closer one may be available)
- Using proxies or VPNs that route traffic through a remote server
- delays in the local network