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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 > address@hidden > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.nongnu.org/mailman/listinfo/linphone-developers > or, via email, send a message with subject or body 'help' to > address@hidden > > You can reach the person managing the list at > address@hidden > > 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 > > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: http://lists.gnu.org/pipermail/linphone-developers/attachments/20080429/5b2d0fda/attachment.html > > ------------------------------ > > _______________________________________________ > Linphone-developers mailing list > address@hidden > http://lists.nongnu.org/mailman/listinfo/linphone-developers > > > End of Linphone-developers Digest, Vol 62, Issue 13 > *************************************************** When was the last time you saw your friends or family? 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