[Asterisk-Users] How to set the jitter buffer

[Brian Cuthie]

So I've read the bug report and, I must admit, I'm totally unclear on why
one would want to pass timestamps end-to-end through a system that was doing
media conversions.

My experience, in case it helps, is that the recent CVS builds (4/9) don't
work well at all with my SIP phone, while backing out the RTP.c mods results
in a huge improvements. I'm using a 7960 and an IAX2 connection through
VoicePulse.

-brian 

> -----Original Message-----
> From: asterisk-users-admin at lists.digium.com 
> [mailto:asterisk-users-admin at lists.digium.com] On Behalf Of Andres
> Sent: Sunday, April 11, 2004 12:55 PM
> To: asterisk-users at lists.digium.com
> Subject: Re: [Asterisk-Users] How to set the jitter buffer
> 
> 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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> >>> 
> >>>
> >>>      
> >>>
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> >
> >---------------End of Original Message-----------------
> >
> >
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> >
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> 
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