Steve-U,
This sip jitterbuffer stuff is still for free, no one has to contribute
anything, but any help financially, with code or testing is greatly
appreciated.
Everything is GPL and code is disclaimed to digium.
We spent the last 2 months researching + working on a sip jitter buffer,
first without using the iax stuff but after long talks with Steve Kann
we started all over with his recommendations to have something more
generic, in the mean time we are trying to improve the PLC and jitter
buffer code as well as testing the stability.
The changes made to chan_sip are only minor but the time used to make
them and test them are not.
Zoa.
(Your PLC code rocks so dont take it back please :)
Steve Underwood wrote:
Olle E. Johansson wrote:
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 [EMAIL PROTECTED] 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
The hard work of building the thing was done for free, and now someone
brings out the begging bowl for the relatively minor activity or
porting into to another home. Frankly, that sucks. Can I have my PLC
code back, please?
Regards,
Steve
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