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[Linphone-developers] RE: Linphone-developers Digest, Vol 62, Issue 13


From: József Békés
Subject: [Linphone-developers] RE: Linphone-developers Digest, Vol 62, Issue 13
Date: Wed, 30 Apr 2008 17:14:10 +0000

Hello Guys

This is the first time I am writing to such a list. Please let me know if I should ask my question somewhere else.

I am writing a client for the libjingle code that uses the mediastreamer for accessing the soundcard. I think it uses mediastreamer, and not mediastreamer2.

I am not 100 percent sure that this is so on every configuration, but on mine libjingle requires explicit access to the soundcard (i.e. I cannot listen to music while using libjingle). I have tried skype on the same configuration, and this is not the case for skype.

My question is if I could upgrade the code to use mediastreamer2 instead of mediastreamer, would this problem be solved? Do you have any other ideas how I could improve the situation?

Thank you for the answers.

Best Regards,
Jozsi
 

> Date: Wed, 30 Apr 2008 12:00:21 -0400
> From: address@hidden
> Subject: Linphone-developers Digest, Vol 62, Issue 13
> To: address@hidden
>
> Send Linphone-developers mailing list submissions to
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> To subscribe or unsubscribe via the World Wide Web, visit
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> When replying, please edit your Subject line so it is more specific
> than "Re: Contents of Linphone-developers digest..."
>
>
> Today's Topics:
>
> 1. RE: oRTP "pulsing" issues (Mike)
>
>
> ----------------------------------------------------------------------
>
> Message: 1
> Date: Tue, 29 Apr 2008 13:18:13 -0400
> From: "Mike" <address@hidden>
> Subject: RE: [Linphone-developers] oRTP "pulsing" issues
> To: "'Vadim Lebedev'" <address@hidden>
> Cc: address@hidden
> Message-ID: <address@hidden>
> Content-Type: text/plain; charset="us-ascii"
>
> 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
>
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>
> End of Linphone-developers Digest, Vol 62, Issue 13
> ***************************************************


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