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Tackling a 3-headed beast – Latency, Jitter, and Packet Loss – for VoIP and Video

VoIP monitoring and troubleshooting are now table stakes for any respectable network analysis system, and video monitoring is quickly becoming so as well as enterprise networks get bogged down under the weight of streaming video, often for extracurricular reasons. Fortunately VoIP and video over IP share an important characteristic – both are real-time protocols with unique network performance requirements. When your quest is problem-free VoIP and Video experience for your end users, you must tackle and slay Latency, Jitter, and Packet Loss.

Latency
You’re probably already familiar with latency. Ever watched the news and listened to a story being broadcasted from the other side of the world? As the signals get bounced from satellite to satellite and, ultimately, to your TV’s receiver, the delay might cause the news anchor and correspondent to interact out of sync. A person who is listening expects to hear the other’s voice in certain amount of time; when this doesn’t happen people start talking over each other. This can happen on your network as well. These delays cause the conversation to come off as unnatural and callers feel like they must “push to talk” or say, “over” to control the conversation and let the other person know that they have finished their thought.

The bright side is that packet-based analysis can measure latency, letting you know whether or not excessive latency is an issue on your network. This way, a network administrator can identify the problem in real-time and solve it before latency issues get worse. How long is too long? That’s somewhat dependent on each situation, but certainly latencies of 100 msec or longer are cause for concern, especially if they are common. In extreme cases, prolonged latencies can cause serious degradation to VoIP calls, including missing words or phrases.

Jitter
Jitter doesn’t have the same simple, real world example as latency, but its effects are just as real. Jitter causes static and other audio anomalies, like stuttering, uneven audio and abnormal speech rhythm, in VoIP calls. Jitter is caused when the data packets that make up the VoIP call are not delivered at regular intervals to the receiver. Regular delivery of IP packets is required for the final digital to analog conversion at the receiver to work correctly. A typical receiver expects packets to be delivered every 20 msec, no more and no less. When the packets start to deviate from this expected delivery sequence, jitter happens.

Jitter can be even more detrimental in multimedia systems. With jitter, videos become jerky or irregular and very difficult to watch. If jitter levels become too high, packet loss can result, with a resulting loss of data.

Packet Loss
And speaking of packet loss, it is just as the name implies. Some packets, meaning some critical media data, never make it to the receiver. Packet loss causes missing sounds, syllables, words or phrases. DSP algorithms may compensate for up to about 30 ms of missing data, but anything more and the algorithms can’t compensate for the data loss.

Real-time protocols like VoIP and video are much more susceptible to packet loss than traditional network data, since there is very little cushion to wait for missing data or to put out-of-order data back into the right sequence. After about 150 msec or so, any data that is missing or out of order is essentially lost forever, since there is no way to properly reconstruct and maintain the real-time data stream, and this creates gaps (packet loss) in the data.

Gaps of more than 30 msec are noticeable to listeners. An average person speaks at a rate of 200 words per minute. That translates to about 3.33 words/sec = 300 ms per word. For G.711, we would need to lose 15 consecutive RTP packets to lose a whole word. Dropping 15 packets/sec for G.711 would be a loss rate of 30%, but losing only a few packets can still be very noticeable. As a general rule of thumb, loss of more than 2 consecutive packets will be heard. Loss rates > 2% will have a strong impact on quality. Losses of 5-10% make calls all but intolerable. Another good rule of thumb is that bursty periods of packet loss are worse than more dispersed loss.

These problems are very common and as more companies move their communication systems onto digital networks, they will only happen more frequently. It is important to have network monitoring and troubleshooting solutions that provide full visibility into all the types of data streaming on your network.

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