[Asterisk-Users] Important :: Please support the development of a new Jitterbuffer for SIP

[Olle E. Johansson]

Steve Kann has developed a new jitterbuffer for IAX2, that hopefully 
will be integrated into Asterisk v1.1 soon, to be part of the 1.2 stable 
relase.

Zoa and his bulgarian team is porting this buffer to SIP/RTP, but needs 
support in the form of funding in order to take the time to test this 
out and complete it in time.

Please paypal your contribution to sponsor at astertest.com today. Every 
little dollar is worth quite a lot!

I fully trust that Joachim (Zoa) and his team will complete this in a 
good way and look forward to improved sound quality in the SIP channel.
Read more here: http://www.astertest.com/forum/viewtopic.php?t=13

Thank you for your contribution!

/Olle

If you're going to VON in San José, meet me, Joachim and other Asterisk
developers in the Asterisk Pavillion!