Jitter Buffer

RTP packets sent over the IP network are subject to random variation in delays, out-of-sequence arrival, and a risk to be dropped. These artifacts decreases audio quality, and the jitter buffer is used to mitigate this. However, while the jitter buffer can improve audio quality it does this to the cost of increased voice delays. Long delays, especially in combination with echoes at far end (for example, caused by 2 to 4 wire hybrids in analogue lines) makes echoes more noticeable and disturbing. Although it might be tried to minimize delays in echo situations (if not the source of the echo could be removed) it must be understood that affecting voice quality due to dropped packets, which have negative impact on the echo canceler. See Echo Canceler (EC).

In MGU3, the jitter buffer can be configured in adaptive or non-adaptive mode, and there are configuration parameters to adjust for actual network conditions.

The configuration of the jitter buffer will be a trade off between audio quality and delays. By default, the jitter buffer in MGU3 is adaptive with settings for a fairly “normal” network, to preserve audio quality over minimizing delays. For a very delay sensitive installation, where audio quality could be negotiated and/or network is very good, re-configuration might be considered.

Although primarily the jitter buffer is for adapting to artifacts caused by network, also VoIP endpoints (phones, gateways, proxies, and so on) is part of the network and can cause these. For example, soft SIP clients with no dedicated HW (e.g. DSP) for VoIP media will have substantially more jitter in outgoing RTP packets than a HW dito. This can cause the jitter buffer to increase and thus to increase the delays even further. In those, and similar scenarios it might appear that the delay through MGU3 is longer than expected.