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IP Phone Adaptive Jitter Buffer

Could I get some details on how the adaptive jitter buffer algorithm works?

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Re: IP Phone Adaptive Jitter Buffer


The playout-delay command allows you to select a jitter buffer mode (fixed or adaptive) and specify certain values used by the DSP algorithms to adjust the size of the jitter buffer. For any voice call, the algorithms read time stamps in the Real-Time Transport Protocol (RTP) headers of a sample of packets to determine the amount of delay the jitter buffer applies to an average packet; that is, as if there is no jitter at all in the network. This is called the average delay.

When you configure the playout-delay mode adaptive option, the DSP algorithms in the codec take samples throughout the voice call and adjust the value of the average delay as network jitter conditions change . The size of the jitter buffer (and the amount of delay applied) is adjusted upward or downward as needed to ensure smooth

transmission of voice frames to the codec, within the minimum and maximum limits you configure. The algorithms are designed to reduce the amount of delay slowly and increase delay quickly during adjustment, so that voice quality is achieved at the risk of larger delay times.

for more info about delay:

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