Latency is one of three key video conferencing performance metrics. The other two are packet loss and jitter.
Latency is the equivalent of not having a person's spoken words match with the movement of their mouth. If you do not prioritize video conference traffic as a whole, latency can cause the effect of a loss of lip synchronization. Audio packets tend to be small (480 bytes or less), while video packets are typically large (800-1500 bytes). Intermediate routers may prioritize the two packet sizes differently, creating differing transit times so the audio and video packets become out of sync.
A typical rule of thumb for latency is < 300 msec round trip between endpoints before users in an interactive call start to notice a delay between the speaker and the receipt of their words by the far end participants. Increased latency can be tolerated, but it will eventually result in awkwardness during the call and a potential hindrance to the free flow of communication as parties talk over each other at > 400 msec. The lesser the interactivity, the more tolerant the latency specification can be, such as in a training situation that is largely one way, even a satellite link might work sufficiently.*
*we'd like to thank TechRepublic for their helpful summary
Latency troubleshooting tips
- Try lowering speeds (i.e. Call Rate or Bandwidth)
- Test when no one or fewer people are in the office
- This may be a network congestion problem. Is the packet loss at the video conferencing unit or on the customer's network stats?
- Lost packets on the video not showing in the network stats usually point to a jitter issue. Packets arriving outside of the jitter buffer will be considered as lost to the video system