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.
