[Asterisk-Users] How to set the jitter buffer

[Rich Adamson]

Andres,

Thanks, I applied the changes and simply dialed the * demo site. Seems to have
totally corrected the problem, although hitting that site is probably not all
that great of a test. 

Is this something that will changed in cvs / stable?

What's the change actually doing (I'm not good at reading C code)?

Rich


> >. As a result, both
> >ends of the c7960 -> iax connection hear choppy audio and audio drop outs.
> >I'm trying to use ethereal decodes to identify the issue, however its rather
> >tough to correlate the audio problems to exact packets within a trace of
> >thousands of packets. (My hearing verses finger response time is not as
> >quick as packet sniffers.)
> >
> >  
> >
> Rich,
> 
> If you are still testing out the Cisco phones give the following rtp.c 
> modification a try.   It basically has the "Timestamp" carryover stuff 
> commented out.  Asterisk thus generates evenly spaced out timestamps.  
> (also note the 2560 change).  My hunch is this will fix your Cisco 
> issues.  Let me know please.
> 
> Andres
> 
> 
>                 /* Re-calculate last TS */
>                 rtp->lastts = rtp->lastts + ms * 8;
>   /*              if (!f->delivery.tv_sec && !f->delivery.tv_usec) { */
>                         /* If this isn't an absolute delivery time, 
> Check if it is close to our prediction,
>                            and if so, go with our prediction */
>                         if (abs(rtp->lastts - pred) < 2560)
>                                 rtp->lastts = pred;
>                         else
>                                 ast_log(LOG_DEBUG, "Difference is %d, ms 
> is %d\n", abs(rtp->lastts - pred), ms);
> /*                }*/
> 
> >Rich
> >
> >
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> >  
> >
> 
> 
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