[Asterisk-Dev] ***TESTING NEEDED*** JITTER BUFFER FOR SIP

[Zoa]

The current version is with a simple fixed size buffer again. (to first
test the stability of the RTP implementation) When this part proves to
be stable, we go back to the adaptive jitter buffer.

There is a manual setting in sip.conf you can set, by default it uses a
300ms buffer.

zoa


Matt wrote:

>I also tested this with several phones over a wide-area broadband
>wireless network where packetloss and delayed packets happen and it
>definately helped!!!
>
>One thing that would be nice would be to know if it is automatically
>adjusting for jitter?  Or is there a manual setting you should be
>setting someplace for like low/medium/high jitter buffer?  But overall
>it seemed to work very well!
>
>
>On 8/25/05, Brian West <brian.west at mac.com> wrote:
>
>
>>Ok we tested the new jitter buffer today from a very distant network
>>with about 20-55% packet loss and lots of jitter you could at least
>>understand the person that the far end... granted I think it was
>>better than SprintPC :P
>>
>>/b
>>
>>On Aug 25, 2005, at 1:10 PM, Olle E. Johansson wrote:
>>
>>
>>
>>>http://bugs.digium.com/view.php?id=3854
>>>
>>>If you really care about a SIP jitter buffer, make sure you help us
>>>testing this yesterday or at least as soon as possible!
>>>
>>>Thank you very much for your help in this matter! If we can get enough
>>>positive test reports quickly, this new jitter buffer will be a great
>>>addition to Asterisk!
>>>
>>>/O
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