Rich Adamson wrote:

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?



This is the way it was **before** :)

We suggested to Mark that he should leave it like this instead of trying to "carry over" the Timestamp which is way more complicated. So far I don't think he liked our suggestion. Maybe if enough people complain he will revert to the old way. Look at this bug for our discussion on the subject:
http://bugs.digium.com/bug_view_page.php?bug_id=0001260



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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