Peter Svensson wrote:
On Tue, 8 Feb 2005, Roy Sigurd Karlsbakk wrote:
is this in HEAD yet?how can I tune SIP jitter? is it possible today in asterisk?I assume you are asking for how to alleviate the effects of jitter on the
RTP audio streams initated by SIP? Asterisk currently only has a jitter
buffer for IAX, not for RTP streams. There are pland for the next
generation jitter buffer code to hook into RTP as well.
There is an entry on the bug tracker that touches on this topic.
See http://bugs.digium.com/bug_view_advanced_page.php?bug_id=0002532
There isn't even any code for SIP yet. However the iax integration works wonders for a link with just a bit of packet loss and jitter. Voice conversations are nice and crisp and without the pops associated with lost packets or growth of the jitter buffer.
Glad it's working for you, Peter..
I don't want to mention names, but somebody has been working on integration into the RTP stack and chan_sip, based on what I wrote in the bug, and some IRC conversations..
-SteveK
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