[Asterisk-Users] Jittery Linksys/Sipura meetme conference fixed

[Rana Dutt]

In a previous message I described how Linksys 942 phone users who dialed in
to a meetme conference at their site heard severe jitter. This was also
experienced with Sipura SPA-2002 ATAs. Users of other IP phones like Polycom
and Snom had no such problem. Also, the Linksys and SPA users had no
problems with regular phone calls, just the meetme conference.

This problem was finally fixed by going in to the settings for the Linksys
phone and setting the RTP frame length to 0.020 and disabling the jitter
buffer adjustment. The same fix also worked for the SPA. Hope this helps
others who have experienced a similar problem.

Rana Dutt
Softel Solutions
www.softelinc.com
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060326/0749075d/attachment.htm