Improving Asterisk call quality with SIP jitter buffers

I had a customer let me know that they had improved their call quality from WiFi and 3G connections by turning on the Asterisk jitter buffers for SIP connections. If you have any extensions where connection quality is intermittent it could be worth trying.

This can by done with the FreePBX SIP Settings module or by adding the following lines to –

/etc/asterisk/sip_general_custom.conf

jbenable=yes
jbimpl=adaptive

 

After changing you should restart Asterisk or ‘sip reload’ from the console.

If you are just using Asterisk the change would go in the [general] section of sip.conf

If you try this option and notice any difference to call quality please post a comment below. Thanks!

5 thoughts on “Improving Asterisk call quality with SIP jitter buffers

  1. Pingback: Link: Improving Asterisk call quality with SIP jitter buffers » TechNotes

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  3. Christina

    I’m trying this out and going to see what happens. Alot of my clients complained that they were sounding as if they they were under water/every other word cutting out.

  4. Maurice Naftalin

    This helped me a lot! I had the jitter buffer turned on already, but changing the implementation to adaptive has made all the difference. I’m using voice recognition, so quality is a real issue.

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