Thanks Vadim,
I tried
messing with this before, and now- I set:
rtp_session_enable_adaptive_jitter_compensation(s, 1);
rtp_session_set_jitter_compensation(s, 60);
and it didn’t
seem to make any difference in the audio.
The output
messages from JitterControl are
ortp-message-JitterControl:
slide=1.60888e+09,jitter=0,count=50
ortp-message-JitterControl:
slide=1.60888e+09,jitter=0,count=100
ortp-message-JitterControl:
slide=1.60888e+09,jitter=0,count=150
ortp-message-JitterControl:
slide=1.60888e+09,jitter=0,count=200
ortp-message-JitterControl:
slide=1.60888e+09,jitter=0,count=250
ortp-message-JitterControl:
slide=1.60888e+09,jitter=0,count=300
ortp-message-JitterControl:
slide=1.60888e+09,jitter=0,count=350
ortp-message-JitterControl:
slide=1.60888e+09,jitter=0,count=400
ortp-message-JitterControl:
slide=1.60888e+09,jitter=0,count=450
ortp-message-JitterControl:
slide=1.60888e+09,jitter=0,count=500
Not sure if
this helps any.
Mike
From: Vadim Lebedev
[mailto:address@hidden
Sent: Tuesday, April 29, 2008 1:03 PM
To: Mike
Subject: Re: [Linphone-developers] oRTP "pulsing" issues
Mike wrote:
Hello,
I’m using the oRTP
library (ortp-0.13.1) in a custom SIP application, and I’m having an
issue with the audio channel “pulsing”; it sounds like the latter
bits of the waveform are being dropped, or the volume decreases or something
like that.
It’s rhythmical, and
consistent, and seems to tick at a 20ms interval or so- and I can get it to
happen with a few different SIP devices (the x-lite soft phone, and a Snom 370
SIP phone, both registered through asterisk).
I’ve extracted out the
code I’m using in my app, into a simple stand alone RTP echo application
(it simply does a recv on the session, and sends the same data back to the same
session)- I extracted it to make sure it wasn’t something else in my code
before I went any further, and it seems to happen with just the oRTP code;
I’ve attached the app I’m using. Everything is configured to only
use u-law (payload type 0).
I wrote another simple UDP
proxy, that simply recv’s packets from a given port, and forwards them
(without modification) back to a given ip/port- when I run that app against the
phone, the audio sounds perfect (as I would expect it to sound through the oRTP
library); this uses the same process through asterisk.
Can anybody see any issues
with my implementation of the oRTP library? Does this sound like any known
issue with the oRTP library?
Thanks,
Mike
_______________________________________________
Linphone-developers mailing list
address@hidden
http://lists.nongnu.org/mailman/listinfo/linphone-developers
Mike,
try to set jitter buffer to 60 ms and not to 20 as you do...
Thanks
Vadim