Thanks again for replying in the on_call_media_state i commended out the second call (the one with reversed reversed parameters) but i still get a full duplex communication, is there anything else that i should do to make it half-duplex? pjsua_conf_connect (ci.conf_slot, 0); /*pjsua_conf_connect (0, ci.conf_slot);*/ Thanks in advance On Fri, Oct 16, 2009 at 7:00 PM, <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. Re: Replacing the audio backend in pjsua (Samuel Vinson) > 2. Audio problem: peer is missing. (Thiago Rondon) > 3. How to Compile Pjsip for Android (buntee b) > 4. changing symbian pjsip from full-duplex to half-duplex > (hlabishi kobo) > 5. Re: changing symbian pjsip from full-duplex to half-duplex > (Srivatsan Deenadayalan) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Thu, 15 Oct 2009 21:11:51 +0200 > From: Samuel Vinson <samuelv@xxxxxxxxxxx> > Subject: Re: Replacing the audio backend in pjsua > To: Shayne O'Neill <shayne.oneill at gmail.com> > Cc: pjsip list <pjsip at lists.pjsip.org> > Message-ID: <4AD773F7.8020305 at laposte.net> > Content-Type: text/plain; charset=ISO-8859-1; format=flowed > > Hello, > > I have begun to port on the new audio API. > I need to make some test, before release a new version, and after to use > the HW/SW codec of iPhone. > > Samuel > > Shayne O'Neill a ?crit : > > > > > > Sorry for the double mail > > > > As an alternative, is there a good template driver that a new iphone > > driver can be built from. Like a stub with all the callbacks , or > > something like that. I might have some time next week I could have try > > at at it. > > I'm not a great coder (samuels a better coder than I , likely) but I > > could at least get a head start on it. > > > > Note that this would still not solve the problem for 'oddball' > > platforms with custom old-school audio drivers. > > > > Shayne. > > > > On 15/10/2009, at 12:44 AM, samuel.vinson wrote: > > > >> > >> Hello, > >> > >> I posted a patch here to resolve your problem, few weeks ago. > >> Because in 1.4 branch, the legacy disapeared :-( > >> > >> Benny could you integrate this patch or fixe the problem, pls. > >> > >> Regards > >> > >> Samuel > >> > >> > >> > Message du 14/10/09 17:21 > >> > De : "Dan Arrhenius" > >> > A : "pjsip list" > >> > Copie ? : > >> > Objet : Re: [pjsip] Replacing the audio backend in pjsua > >> > > >> > > >> > It didn't work for me to define > >> PJMEDIA_AUDIO_DEV_HAS_LEGACY_DEVICE. I should probably > >> > make it clear that I'm using pjsua-lib, so I don't initialize the > >> audio directly in my code. > >> > > >> > In audiodev.c there is support for maximum 16(MAX_DRIVERS) audio > >> device factories, but > >> > they are added and initialized statically, and in my case no driver > >> at all is added :-( > >> > Might I suggest the ability to dynamically add audio device > >> factories, for example > >> > 'pjmedia_aud_subsys_add_driver(...)'. > >> > > >> > Best regards, > >> > Dan > >> > > >> > > >> > Benny Prijono wrote: > >> > > On Wed, Oct 14, 2009 at 5:45 PM, Dan Arrhenius wrote: > >> > >> Hello, > >> > >> I've been working with pjproject 1.0.x and want to upgrade to > >> the latest > >> > >> version. > >> > >> How can I replace the audio back-end in pjsua with my own using > >> the new > >> > >> audio subsystem? With the old version I configured pjproject with > >> > >> '--enable-ext-sound' and supplied rules to build the audio > >> back-end in > >> > >> user.mak. > >> > >> > >> > >> As I understand it all available audio back-ends are hard-coded in > >> > >> audiodev.c (PORTAUDIO, WMME, SYMB_VAS, SYMB_APS, and SYMB_MDA), > >> and there is > >> > >> no way of dynamically add a new audio driver. Or am I missing > >> something? > >> > >> Do I have to modify audiodev.c to get my own audio back-end in > >> pjsua? I want > >> > >> to modify as little code in pjproject as possible to ease > >> maintenance. > >> > >> > >> > >> > >> > > > >> > > In http://trac.pjsip.org/repos/wiki/Audio_Dev_API there is a > >> guide on > >> > > how to access legacy device using the new API (see under > >> > > PJMEDIA_AUDIO_DEV_HAS_LEGACY_DEVICE). I have not tested it with > >> > > --enable-ext-sound, but it's supposed to work. :) > >> > > > >> > > Cheers > >> > > Benny > >> > > > >> > > _______________________________________________ > >> > > 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 > >> > > >> > > >> > >> > >> > >> Gratuite, garantie ? vie et d?j? utilis?e par des millions > >> d'internautes... > >> vous aussi, pour votre adresse e-mail, choisissez laposte.net. > >> > >> Laposte.net, bien + qu'une messagerie > >> > >> _______________________________________________ > >> 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 > > > > =================================== > > Shayne O'Neill Development > > Mobile, Web and Business process integration. > > shayne.oneill at gmail.com 0400247091 > > Ask me about how Alfresco can help your business grow. > > > > > > > > > > > ------------------------------ > > Message: 2 > Date: Fri, 16 Oct 2009 02:12:19 -0300 > From: Thiago Rondon <thiago@xxxxxxxxxxxx> > Subject: Audio problem: peer is missing. > To: pjsip list <pjsip at lists.pjsip.org> > Message-ID: <4AD800B3.8040806 at aware.com.br> > Content-Type: text/plain; charset=ISO-8859-1; format=flowed > > > Hi, > > I have a problem, for call telephone numbers it's ok, but when I make > call to users to make a P2P connection, I have one problem.. > > [CONFIRMED] To: > sip:thiago at sip.domaincom;tag=ca6ac557c0f0496091cbad383cef2bdf > Call time: 00h:00m:14s, 1st res in 3110 ms, conn in 3110ms > SRTP status: Not active Crypto-suite: (null) > #0 iLBC @8KHz, sendrecv, peer=- > RX pt=117, stat last update: 00h:00m:00.141s ago > total 1pkt 0B (40B +IP hdr) @avg=0bps/21bps > pkt loss=0 (0.0%), discrd=0 (0.0%), dup=0 (0.0%), reord=0 (0.0%) > (msec) min avg max last dev > loss period: 0.000 0.000 0.000 0.000 0.000 > jitter : 0.000 0.000 0.000 0.000 0.000 > TX pt=117, ptime=90ms, stat last update: never > total 164pkt 24.6KB (31.1KB +IP hdr) @avg 13.2Kbps/16.8Kbps > pkt loss=0 (0.0%), dup=0 (0.0%), reorder=0 (0.0%) > (msec) min avg max last dev > loss period: 0.000 0.000 0.000 0.000 0.000 > jitter : 0.000 0.000 0.000 0.000 0.000 > RTT msec : 0.000 0.000 0.000 0.000 0.000 > > Look, the peer=- is empty, why ? > > I connect each other, but I doesnt listen nothing, maybe because of this > peer. > > I look at wireshark, I doesn't have problem with NAT. > > Thanks! > > > > > ------------------------------ > > Message: 3 > Date: Fri, 16 Oct 2009 13:49:21 +0530 > From: buntee b <b.buntee@xxxxxxxxx> > Subject: How to Compile Pjsip for Android > To: pjsip at lists.pjsip.org > Message-ID: > <caaffa760910160119j3b39d1eu59bfcc220c6d99e8 at mail.gmail.com> > Content-Type: text/plain; charset="iso-8859-1" > > Hi All > > I would like to employ the Pjsip on Android platform , is it possible?..... > if yes then please suggest me > the process.... how to complile pjsip for Android? > > > Regards > Buntee > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: < > http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/attachments/20091016/4644e649/attachment-0001.html > > > > ------------------------------ > > Message: 4 > Date: Fri, 16 Oct 2009 11:53:36 +0200 > From: hlabishi kobo <hlabishik@xxxxxxxxx> > Subject: changing symbian pjsip from full-duplex to > half-duplex > To: pjsip at lists.pjsip.org > Message-ID: > <b87c2d4c0910160253g616cd208ie136ea1a946e300f at mail.gmail.com> > Content-Type: text/plain; charset="iso-8859-1" > > hi > can someone help me, how can i change symbian pjsip from full-duplex to > half-duplex > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: < > http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/attachments/20091016/fcf0baf5/attachment-0001.html > > > > ------------------------------ > > Message: 5 > Date: Fri, 16 Oct 2009 19:19:14 +0530 > From: Srivatsan Deenadayalan <srivatsan@xxxxxxxxxxx> > Subject: Re: changing symbian pjsip from full-duplex to > half-duplex > To: pjsip list <pjsip at lists.pjsip.org> > Message-ID: <4AD879DA.1060601 at ongobiz.com> > Content-Type: text/plain; charset="iso-8859-1"; Format="flowed" > > Hi, > > Call back method *on_call_media_state *is the place you can manage full > / half duplex audio, > > pjsua_conf_connect (ci.conf_slot, 0); > pjsua_conf_connect (0, ci.conf_slot); > > First a half duplex is established by connecting remote port with your > port and again a half duplex is established by connecting your port with > remote port, in turn creates a full duplex audio. > > Hope this helps... > > Thanks, > Srivatsan D > > > hlabishi kobo wrote: > > hi > > can someone help me, how can i change symbian pjsip from full-duplex > > to half-duplex > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > 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/20091016/cf36dec9/attachment-0001.html > > > > ------------------------------ > > _______________________________________________ > pjsip mailing list > pjsip at lists.pjsip.org > http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org > > > End of pjsip Digest, Vol 26, Issue 32 > ************************************* > -------------- next part -------------- An HTML attachment was scrubbed... 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