[Asterisk-Users] How to set the jitter buffer

Andres andres at telesip.net
Sun Apr 11 09:07:43 MST 2004


>. 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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