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Tackling ‘Choppy Audio’ or ‘Voice Breaking’ with Jitter Buffer Settings in Your IP PBX

Using a jitter buffer can help VoIP conversations sound better and avoid choppy audio. The way jitter buffer technology works is by slightly delaying both your incoming and outgoing voice packets. You may eliminate packet loss issues by employing this technique.

The end-to-end delay brought on by the jitter buffer and the packet loss is traded off. End-to-end mouth-to-ear latency, including jitter buffer, should be as short as feasible or at least between 150 ms and 250 ms for a comfortable VOIP conversation.

How to enable Jitter Buffer via IP PBX server?

Here is a working example:

Jitter buffer=enabled

Force Jitter Buffer=yes

Implementation=Adaptive

Jitter Buffer logging=Disable

Jitter Buffer Size =300 jbresynchthreshold=500

Jitter buffer is usually put in place after the following are taken care of:

  • Good connection quality. You can use websites such as https://www.fusionconnect.com/speed-test-plus to test your connection.
  • Use a cabled connection on the client side.
  • Prioritize packets.
  • Minimize unnecessary bandwidth usage.

Some open source VOIP software are Issabel, Asterisk, etc. These programs are frequently handled using FreePBX, another open-source program. When you first use VoIP, the experience won’t always be ideal, but problems with the service often have straightforward remedies. Apps4rent can assist you to improve your VoIP experience if you’re struggling with far more chronic and complicated VoIP troubles.

About the Author
Apps4Rent Author George Dockrell
George Dockrell writes practical, solution-focused content for Apps4Rent. With a strong grasp of cloud platforms and business applications, he simplifies complex topics like application hosting, hosted Exchange, QuickBooks hosting, SharePoint hosting, and desktop virtualization into clear, actionable insights. His work helps businesses navigate hosting solutions, integrations, and service management with confidence.

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