[Asterisk-Users] RC1 problem? (Conversation over two IAX2 streams
= nasty, gappy audio)
steve at daviesfam.org
steve at daviesfam.org
Fri Aug 6 14:36:09 MST 2004
On Fri, 6 Aug 2004, Andrew Kohlsmith wrote:
> I've been having 'gappy' audio problems with nufone for about a week now but I
> think I've nailed it down.
>
> Setup:
> office* - iax2 - colo* - iax2 - nufone
> What I believe is happening is some kind of bug with regard to having a
> conversation go over two IAX2 streams.
>
> A call to Nufone will show "normal" lag and jitter values between colo* and
> Nufone, but the jitter values between office* and colo* seem to be unsigned
> but negative:
>
> Peer Username ID (Lo/Rem) Seq (Tx/Rx) Lag Jitter JitBuf
> Format
> 192.168.2.2 benphone 16385/16387 00178/00177 00006ms 6356992ms
> 0805ms GSM
> 66.225.202.72 benshaw 16386/00025 00169/00171 00051ms 0012ms 0026ms
> GSM
>
> As a result the jitter buffer grows and grows and grows... and we get
> horrendous audio. jitterbuffer=off doesn't seem to make a difference.
OK...
jitterbuffer=off isn't valid. Did you mean =no? If you do turn the
jitterbuffer off then it will stop being used - frames will be delivered
immediately they arrive. But the jitterbuffer code will still make and
report its measurements. This is confusing, I must admit...
I have done lots of testing on a setup just like yours with good results.
Except without trunking (but you said trunking makes no difference)
I'd like you to send me a captured debug log from "colo" machine for a
call with this problem.
Here's what to do:
in logger.conf make sure you have a line like so:
debug => notice,warning,error,debug,verbose
stop asterisk.
delete or rename existing debug file ("rm /var/log/asterisk/debug")
start asterisk with lots of debugging (eg "/usr/sbin/asterisk -vvvvvv -g -dddddd -c")
at CLI> prompt, enter "iax2 debug"
now initiate your call from "office" via nufone.
once your done with the call, quit asterisk, gzip up the debug file and
email it to me.
Let me see what is happening.
Regards,
Steve
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