[Asterisk-Dev] Please help test jitterbuffer/PLC patches.
Jonne Kodu
jkodu at hotmail.com
Wed Mar 16 06:44:35 MST 2005
Hi,
Is the new jitterbuffer channel indepentent already?
If not, can anybody estimate when it will work with chan_sip also?
/J
>From: Steve Kann <stevek at stevek.com>
>Reply-To: Asterisk Developers Mailing List <asterisk-dev at lists.digium.com>
>To: Asterisk Developers Mailing List <asterisk-dev at lists.digium.com>
>Subject: [Asterisk-Dev] Please help test jitterbuffer/PLC patches.
>Date: Tue, 15 Mar 2005 12:51:38 -0500
>
>
>
>Hi,
>
> The Jitterbuffer and PLC patches, plus IAX2 integration posted on
>mantis at http://bugs.digium.com/bug_view_advanced_page.php?bug_id=0002532
>and the "trunktimestamp" patch posted at
>http://bugs.digium.com/bug_view_advanced_page.php?bug_id=0003400 are now
>ready for testing and review.
>
> The new jitterbuffer offers several new features, and gives us a
>foundation for even more improvement. It presently includes the following
>features that may be of interest:
>
>1) Packet Loss Concealment: It detects lost packets, and attempts to
>reconstruct them, concealing their loss. With this functionality, you can
>hardly notice 10% packet loss situations.
>
>2) Greatly improved diagnostics, including IAX2 "receiver reports", where
>you can get good details of the network situation with "iax2 show
>netstats". (A manager-specific interface to this will be added soon -- it
>would also be nice to integrate this with the new enhanced CDR system, so
>these metrics can be stored for post-mortem analysis).
>
>3) Some testing capabilities with a new command "iax2 test losspct
><percentage>" which simulates a particular rate of random incoming packet
>loss.
>
>4) Enhanced tuning for jitter vs loss.
>
>5) Automatic bypass of the JB for VoIP <-> VoIP bridging (configurable).
>
>
>TODO:
>
>Since the jitterbuffer (any jitterbuffer, really) requires getting sensible
>timestamps, it is important that IAX2 clients output timestamps correctly.
>In particular, discontinuities in timestamps will lead to audio loss or
>large delays. There's some cleanup in chan_iax2 in particular that can
>address some of the causes of this.
>
>Need to re-examine if IAX2 bridge optimization is doing the right things.
>
>
>STATUS:
>This code is presently in production on a handful of production boxes with
>pretty good results. If we can get some more reports on it, we can
>hopefully get this into CVS. Also useful would be for people to test this
>with the jitterbuffer (and PLC) turned off.
>
>PATCHES:
>You really ought to read the bug reports before you apply these, but here's
>the patches in question, in (one possible) order they will apply most
>cleanly:
>
>
>[IAX2 trunk timestamps]
>http://bugs.digium.com/file_download.php?file_id=5124&type=bug
>
>[Add PLC to codecs]
>http://bugs.digium.com/file_download.php?file_id=5155&type=bug
>
>[Build the JB with asterisk]
>http://bugs.digium.com/file_download.php?file_id=5137&type=bug
>
>[Integrate JB with IAX2]
>http://bugs.digium.com/file_download.php?file_id=5138&type=bug
>
>[Add new "channel properties" flag to identify VoIP technologies]
>http://bugs.digium.com/file_download.php?file_id=5139&type=bug
>
>[Optionally bypass IAX2 jitterbuffer when we're bridged to a VoIP channel]
>http://bugs.digium.com/file_download.php?file_id=5140&type=bug
>
>
>-SteveK
>
>
>
>
>
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