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VoIP Troubleshooting 101 – Latency & Jitter March 29, 2010

Posted by TelUS Consulting Services in VoIP catagory.


Latency (or delay) is the time that it takes a packet to make its way through a network
end-to-end. In telephony terms, latency is the measure of time it takes the talker’s voice to reach the listener’s ear. Large latency values do not necessarily degrade the sound quality of a phone call, but the result can be a lack of synchronization between the speakers, such that there are hesitations in the speaker’ interactions. Generally, it is accepted that the end-to-end latency should be less than 150 ms for toll quality phone calls. To ensure that the latency budget remains below 150 ms, administrators need to take into account the following primary causes of latency. When designing a multiservice network, the total delay that a signal or packet exhibits is a summation of all the latency contributors.

One source of latency is the time it takes for the endpoints to create the packets used in voice services. These “packetization” delays are caused by the amount of time it takes to fill a packet with data. Generally, the larger the packet size, the greater the amount of time it takes to fill it. Packetization delay is governed by the CODEC standard being used. This problem also exists on the receiving side because the media gateway must remove and further process the packet data. If the packets are kept small, this amount of delay, in both directions, is usually quite small, depending on the hardware / software implementation of the media gateways. All considerations being equal, nominal operation of any media gateway unit should not exceed 30 ms.

Another source of latency is the delay it takes to serialize the digital data onto the physical links of the interconnecting equipment. This delay is inversely proportional to the link speed. In other words, the faster the media, the lower the latency. This value is somewhat dependent on the link technology used and its access method. For example, it takes 125 microseconds to place one byte on a 64-Kb circuit. The same byte placed on an OC-3/STM-1 circuit takes 0.05 microseconds. Although this delay is
unavoidable (regardless of the bandwidth used), keeping the number of intervening links small and using high bandwidth interfaces reduces the overall latency.

Propagation delay is the time it takes an electrical (or photonic) signal to traverse the length of a conductor. The speed of these signals is always slower than that of the speed of light. There is always propagation delay; however, it only becomes an issue when the signal (or packet) travels a great distance. The accepted formula for calculating propagation delay is as follows.

Propagation delay = Circuit km / (299,300 km x .6)
Example: Calculation of one-way propagation delay of a 6,000 km fiber run
(discounting any signal repeaters in between)
0.0334 sec = 6000 km / (299,300 km x .6)

By this calculation, the latency contributed by just propagation delay would be 33.4 ms.

A queuing delay, which is a large source of latency, is the amount of time that a packet remains buffered in a network element while it awaits transmission. Network traffic loads result in variable queuing delays. The amount of buffer that a queue uses is usually a configurable parameter, with a smaller number being better for latency values. However, this delay is also based on the amount of traffic the element is
trying to pass through a given link, and therefore it increases with network load. Hence, you need to set aside adequate bandwidth and resources for voice traffic. If the queue used for voice traffic is not serviced fast enough and that queue is allowed to grow too large, the result is greater latency.

Packet forwarding delay is the time it takes a network device (router, switch, firewall, etc.) to buffer a packet and make the forwarding decision. Included in that decision could be which interface to forward the packet to, whether to drop or forward the packet against an Access Control List (ACL) or security policy, etc. Packet forwarding delay is variable and depends on the function and architecture of
the networking device. If a packet must be further buffered as a part of its processing, greater latency is incurred.


Jitter is the measure of time between when a packet is expected to arrive to when it actually arrives. In other words, with a constant packet transmission rate of every 20 ms, every packet would be expected to arrive at the destination exactly every 20 ms. This situation is not always the case.

The greatest culprit of jitter is queuing variations caused by dynamic changes in network traffic loads. Another cause is packets that might sometimes take a different equal-cost link that is not physically (or electrically) the same length as the other links. Media gateways have play-out buffers that buffer a packet stream, so that the reconstructed voice waveform is not affected by packet jitter. Play-out buffers can
minimize the effects of jitter, but cannot eliminate severe jitter. Although some amount of jitter is to be expected, severe jitter can cause voice quality issues because the media gateway might discard packets arriving out of order. In this condition, the media gateway could starve its play-out buffer and cause gaps in the reconstructed waveform.

our next installment of VoIP Troubleshooting will be on Bandwidth utilization

Joe Buck, N.C.E.

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