Most humans don’t notice audio delays of less than 150 ms, so this is the well-accepted one-way maximum latency in the voice environment. This value is very difficult to achieve in current videoconferencing environments, and, because the audio has to sync with the video, whichever is slower will govern the user’s experience.
Cisco says their system creates less than 200 ms of latency between two adjacent systems, with no network in between. Most humans notice delays above 250 ms, so this leaves only about 50 milliseconds of latency budget for the networks connecting the rooms.
Unfortunately, 50 ms barely gets you across the continental U.S. Table 2 shows the network latency between New York and various cities, calculated with a conservative model I created a few years ago based on a large dataset from an international network service provider. By conservative, I mean that some improvement to these latencies is possible through careful network design, although the speed of light will always contribute a significant portion of these values.
Network latency does not affect the quality of the video images or the sound, but it does affect the interactive nature of the conference. People can learn to adjust to these delays, as they did with satellite-based long distance calls, but it takes practice, it’s annoying, it’s tiring and it breaks the illusion of being in the same room.
It’s clear in Table 2 that delays will be noticeable on telepresence sessions between New York and most U.S. cities west of the Mississippi. Even lengthier delays will occur between New York and the cities in the Asia Pacific region. Telepresence vendors need to improve the quality of experience with innovations that reduce the latency added by the video equipment.
Jitter is important because packets arriving late can miss their scheduled play window. The video image is being reproduced in real time, and thus the data must arrive on schedule so that it can be used as a part of the video construction. A packet arriving late cannot be used.
Most video and audio systems have a jitter buffer, which delays the audio or video reconstruction by some amount of time, like starting an event 10 minutes late to allow tardy patrons the opportunity to arrive and be seated. Typical jitter buffers are 50 milliseconds long, allowing packets to be up to 50 ms late and still be used.
Of course, the jitter buffer delay contributes to the latency of the system and to the overall latency of the end-to-end connection. System designers must trade off the quality of the network (e.g., its ability to deliver packets with low jitter) against the total latency of the video system.
The jitter buffers of traditional videoconferencing systems range from 50 to 100 milliseconds. VOIP phones often have dynamic jitter buffers which contract when the network quality is good and expand as the quality of the network degrades. Some can expand to 100 or 200 milliseconds.
In contrast, the newer telepresence systems, with their emphasis on providing as close to a “real” conferencing environment as possible, are designed with smaller jitter buffers and thus require the network to cause less jitter. Cisco recommends that the network jitter remain below 10 milliseconds for telepresence system support. This is a significant challenge in a converged, global network.
