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How to use the Jitter Buffer and how to handle latency?

This article is a brief introduction about using the jitter buffer and handling latency in relation with Ozeki VoIP SIP SDK. After reading through this page you will be fully familiar with all the essential terms concerning the jitter buffer and latency handling and what you will need for creating your own solution using Ozeki VoIP SIP SDK.

What is Jitter Buffer? How does it work?

In voice over IP (VoIP), a jitter buffer is a shared data area where voice packets can be collected, stored, and sent to the voice processor in evenly spaced intervals. Variations in packet arrival time, called jitter, can occur because of network congestion, timing drift, or route changes. The jitter buffer, which is located at the receiving end of the voice connection, intentionally delays the arriving packets so that the end user experiences a clear connection with very little sound distortion.

There are two kinds of Jitter Buffers:

  • Static Jitter Buffer: hardware-based and is configured by the manufacturer.
  • Dynamic Jitter Buffer: software-based and can be configured by the network administrator to adapt to changes in the network's delay.

How to use Jitter Buffer in C#?

Ozeki VoIP SIP SDK contains dynamic jitter buffer technology. This technique is automatically used in the SDK, so you do not need and cannot turn it on or off, all your VoIP solutions will use this feature without any further method call or implementation.

Training guides, simple examples

If you would like to visit the detailed developer documentations, you can learn much more from the Ozeki VoIP SIP SDK Training chapter.


If you have any questions or need assistance, please contact us at info@voip-sip-sdk.com

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