Hi, I just d/l and compiled the sip stack on mac and was wondering if anyone had an xcode project setup? if not, that is my goal, and will share it. thanks, brad pjsip-request at lists.pjsip.org wrote: > Send pjsip mailing list submissions to > pjsip at lists.pjsip.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org > or, via email, send a message with subject or body 'help' to > pjsip-request at lists.pjsip.org > > You can reach the person managing the list at > pjsip-owner at lists.pjsip.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of pjsip digest..." > > > Today's Topics: > > 1. VAD matter (Markus Vechiorno) > 2. Re: asterisk support any kind of codec and can translate it > freely (Benny Prijono) > 3. Re: asterisk support any kind of codec and can translate it > freely (Olivier DERVYN) > 4. Re: Audio Stuttering in SVN? (Nanang Izzuddin) > 5. Re: bug in pjisp (Nanang Izzuddin) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Tue, 8 Apr 2008 10:27:51 +0200 > From: Markus Vechiorno <finalpfc@xxxxxxxxxxx> > Subject: VAD matter > To: lista pjsip.org <pjsip at lists.pjsip.org> > Message-ID: <BAY111-W227F922DD6D2933E5F46A6D6F20 at phx.gbl> > Content-Type: text/plain; charset="iso-8859-1" > > > Hi, > > I need to know when a silence is detected. > > What I need is to do some actions when a silence is detected. Imagine this situation: > I am running pjsua and I command Q. Before, I had included some functionality dor this command, but I need this functionallity to start when the silence is detected. > > I have thought that maybe pjsip have a simple solution for this matter (a trigger or smth similar). > > Any idea? > > Thanks!!! > _________________________________________________________________ > MSN Video. > http://video.msn.com/?mkt=es-es > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/attachments/20080408/b9bba68f/attachment-0001.html > > ------------------------------ > > Message: 2 > Date: Tue, 8 Apr 2008 09:32:20 +0100 > From: "Benny Prijono" <bennylp@xxxxxxxxx> > Subject: Re: asterisk support any kind of codec and can > translate it freely > To: pjsip <pjsip at lists.pjsip.org> > Message-ID: > <1879720d0804080132g613d9826ud619ac040c66b54c at mail.gmail.com> > Content-Type: text/plain; charset=ISO-8859-1 > > 2008/4/8 hlzhangxt <hlzhangxt at 163.com>: >> >> >> Asterisk can support any kind of codec and translate them in asterisk >> server. >> >> And I test with eyebeam, it works fine, and eyebeam always show the codec >> used is g729, no changing. >> >> In then function play_cb there is comments: >> >> /* We're risking accessing the port without holding any mutex. >> * It's possible that port is disconnected then destroyed while >> * we're trying to access it. >> * But in the name of performance, we'll try this approach until >> * someone complains when it crashes. >> */ >> >> So i think if you add some mutex, maybe it will work. >> > > Maybe, or maybe not. The code has been like that for ages and it is > fine, as there is a certain step to follow to disconnect the port from > the sound port to avoid the race condition. Sorry I don't know what > are you trying to say here. What's the problem? > > -benny > > > > ------------------------------ > > Message: 3 > Date: Tue, 8 Apr 2008 10:34:00 +0200 > From: "Olivier DERVYN"<der@xxxxxxxxxxx> > Subject: Re: asterisk support any kind of codec and can > translate it freely > To: pjsip at lists.pjsip.org > Message-ID: <200804080834.m388Y0un030521 at me-ml2.teamlog.fr> > > Bonjour. > Je suis absent jusqu'au 11avril 2008 inclus. > Pendant cette p?riode, merci de contacter > joseph.faure at teamlog.com si n?cessaire. > _________ > Hello. > I'm on leave until April 11 included. > For this period, please contact joseph.faure at teamlog.com > > Cordialement/Regards, > Olivier DERVYN > > > > > ------------------------------ > > Message: 4 > Date: Tue, 8 Apr 2008 16:25:18 +0700 > From: "Nanang Izzuddin" <nanang@xxxxxxxxx> > Subject: Re: Audio Stuttering in SVN? > To: "pjsip list" <pjsip at lists.pjsip.org> > Message-ID: > <c7f43120804080225l4495293bha87406840543677 at mail.gmail.com> > Content-Type: text/plain; charset="iso-8859-1" > > Hi Norman, > > Was the calls done locally in LAN, which potentially has no problem with > jitter? > Furthermore, if you have time, please try as Benny said, > use older revision (r 1886 or older) for jitter buffer (jbuf.h & c) but use > latest revision for the rest of the codes. > > Regards, > nanang > > > On 04/04/2008, Norman Franke <norman at myasd.com> wrote: >> Unfortunately, that's going to be hard. I can let our folks use this >> normally since it makes some calls impossible to hear, but not all. I can >> try to reproduce it on my end, though. As someone mentioned, it may have >> something to do with loud audio. Not sure. I'll see what I can find out. >> >> Norman Franke >> Answering Service for Directors, Inc. >> www.myasd.com >> >> On Apr 1, 2008, at 5:03 PM, Nanang Izzuddin wrote: >> >> I could not reproduce and still unsure which part is causing the stutter. >> Could you send the log file with log level 5 of the stuttering pjsua? >> >> Regards, >> nanang >> >> >> On 01/04/2008, Norman Franke <norman at myasd.com> wrote: >> >> This is under OS X 10.4.11 compiled with: >> >> ./aconfigure --disable-ssl CFLAGS='-O2 -DNDEBUG' >> >> And with these options: >> >> >> APPCFG.media_cfg.clock_rate = 16000; >> APPCFG.media_cfg.no_vad = PJ_TRUE; // Disable silence detection >> APPCFG.media_cfg.ec_tail_len = 0; >> APPCFG.media_cfg.quality = 3; >> >> Otherwise, everything else is default. It doesn't always happen, only >> about >> 15% of the time. >> >> >> Norman Franke >> Answering Service for Directors, Inc. >> www.myasd.com >> >> On Apr 1, 2008, at 1:19 AM, Nanang Izzuddin wrote: >> >> Hi Norman, >> >> There are major changes (mostly enhancements) have been done recently >> in the pjmedia, so far everything looks fine in our test machines, and >> we are actually waiting for feedbacks. >> To make it easier to reproduce, could you specify the media related >> command line/configuration of pjsua? >> >> Thanks, >> nanang >> >> >> On 01/04/2008, Norman Franke <norman at myasd.com> wrote: >> I've been noticing what appears to be like a stuttering or picket fencing >> of incoming audio with the latest svn branch. Did something change in >> early >> or mid March? I've reverted to 06-Mar-2008 and that seems fine (which is >> after the splitcomb changes, which I also use.) >> >> >> Norman Franke >> Answering Service for Directors, Inc. >> www.myasd.com >> >> _______________________________________________ >> Visit our blog: http://blog.pjsip.org >> >> pjsip mailing list >> pjsip at lists.pjsip.org >> http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org >> >> >> >> >> _______________________________________________ >> Visit our blog: http://blog.pjsip.org >> >> pjsip mailing list >> pjsip at lists.pjsip.org >> http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org >> >> _______________________________________________ >> Visit our blog: http://blog.pjsip.org >> >> pjsip mailing list >> pjsip at lists.pjsip.org >> http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org >> >> >> >> _______________________________________________ >> Visit our blog: http://blog.pjsip.org >> >> pjsip mailing list >> pjsip at lists.pjsip.org >> http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org >> >> >> >> _______________________________________________ >> Visit our blog: http://blog.pjsip.org >> >> pjsip mailing list >> pjsip at lists.pjsip.org >> http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org >> >> > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/attachments/20080408/d9bc96b0/attachment-0001.html > > ------------------------------ > > Message: 5 > Date: Tue, 8 Apr 2008 17:04:47 +0700 > From: "Nanang Izzuddin" <nanang@xxxxxxxxx> > Subject: Re: bug in pjisp > To: "pjsip list" <pjsip at lists.pjsip.org> > Message-ID: > <c7f43120804080304x3362d13br4f2f22bb1be040b6 at mail.gmail.com> > Content-Type: text/plain; charset="gb2312" > > It seems the packets are coming but rejected by any component of pjmedia > stream. > (which most suspected would be codec parser or jitter buffer) > > There is a scenario, empty jitter buffer + restarting RTP timestamp, > which may causing jitter buffer ignoring packets, this only happens on the > older version > of jitter buffer and it seems you are not using the latest SVN. > So please update your source from the latest SVN. > > Just to make sure that the problem is not in G.729 implementation, > please also try to call using PCMU codec. > > Thanks, > nanang > > > On 08/04/2008, hlzhangxt <hlzhangxt at 163.com> wrote: >> >> >> >> This is the log file, I hold B three times, the last time the problem >> happend. The first twice normal. >> >> You can establish asterisk, and try it. >> >> >> ------------------------------ >> 500???????????????????? <http://popme.163.com/link/004199_0407_8566.html> >> _______________________________________________ >> Visit our blog: http://blog.pjsip.org >> >> pjsip mailing list >> pjsip at lists.pjsip.org >> http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org >> >> >> > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/attachments/20080408/bee5bd0b/attachment.html > > ------------------------------ > > _______________________________________________ > pjsip mailing list > pjsip at lists.pjsip.org > http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org > > > End of pjsip Digest, Vol 8, Issue 24 > ************************************ >
