oRTP "pulsing" issues

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

 

 

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

 

 

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

 

 

 ;

 

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Mike, try to set jitter buffer to 60 ms and not to 20 as you do...


Thanks
Vadim