Hi Samuel, i checked SRTP settings. SRTP is manually disabled in pjmedia/include/pjmedia/config_site.h. // SRTP has not been ported to iPhone yet. // SRTP = secure real-time transport protocol. #undef PJMEDIA_HAS_SRTP #define PJMEDIA_HAS_SRTP 0 is there any other solution ? Thanks On Thu, Feb 25, 2010 at 10:40 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. Ring Tone in symbian_ua (Ricardo Antonio Ponce Gomez) > 2. I wanna know about Multi call? (raksa chao) > 3. Calling issue on iPhone (tech guy) > 4. Service Based Features like CW, CF (Premalatha Kuppan) > 5. Please Help, carrying TCP data with pjnath session (Kabil Akp?nar) > 6. Re: Calling issue on iPhone (samuel.vinson) > 7. Re: jitter buffeer - reimplementation (Fabio Cherchi) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Wed, 24 Feb 2010 16:15:13 -0600 > From: Ricardo Antonio Ponce Gomez <ricardo2500@xxxxxxxxxxx> > To: <pjsip at lists.pjsip.org> > Subject: Ring Tone in symbian_ua > Message-ID: <COL122-W24781A8E58DC12DCACAB38A8410 at phx.gbl> > Content-Type: text/plain; charset="iso-8859-1" > > > Hi all.I'm trying to play a ring tone when an incomming call is there. I'm > using the CMdaAudioPlayerUtility API and it just work fine the first time, i > can hear a .wav audio file in the loudspeaker, but after I finish with this > call, and there's a second incomming call my ring tone it's gone. I'm dong > something like this: > _LIT(tono,"ring.wav");CPlayer *player = CPlayer::NewL(tono); > //on_incoming_call callback....pjsua_call_answer(call_id, 180, NULL, NULL); > player->Play();.... > //on_call_state callback......if (ci.state == PJSIP_INV_STATE_DISCONNECTED) > { if (call_id == g_call_id){ g_call_id = -1; > player->Stop(); }..... > Do you think i'm missing something? Is there any other better option to do > this? I'm using VAS. > Thanks & Regards > _________________________________________________________________ > Prefiero un d?a sin coche que sin Messenger > www.vivirmessenger.com > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: < > http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/attachments/20100224/36238dc3/attachment-0001.html > > > > ------------------------------ > > Message: 2 > Date: Thu, 25 Feb 2010 16:07:17 +0700 > From: raksa chao <chao.raksa@xxxxxxxxx> > To: pjsip at lists.pjsip.org > Subject: I wanna know about Multi call? > Message-ID: > <1f8c401002250107w55f59d6fg5784e6b4b6d84e01 at mail.gmail.com> > Content-Type: text/plain; charset="iso-8859-1" > > Hi all! > I wonder when i make call using sipek i can only 4 line at the same > time. Can i make call more line 10 and more at the same time? Do i need to > change something in PJSIP or SIPEK? > > Please I need your help. > > Thanks,, > Raksa,, > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: < > http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/attachments/20100225/c4b73d3a/attachment-0001.html > > > > ------------------------------ > > Message: 3 > Date: Thu, 25 Feb 2010 15:43:10 +0530 > From: tech guy <techguy244@xxxxxxxxx> > To: pjsip at lists.pjsip.org > Subject: Calling issue on iPhone > Message-ID: > <951e50ce1002250213h1ed21b5eh3cd18d706d6adb6 at mail.gmail.com> > Content-Type: text/plain; charset="iso-8859-1" > > Hi All > > I have compiled the pjsip for iPhone. I got registered properly but when I > try to make a call, I m getting the following error: > > ipodsound.c pjmedia_snd_stream_start. 22:47:53.441 ipodsound.c > pjmedia_snd_stream_start : play back starting... 22:47:53.668 ipodsound.c > pjmedia_snd_stream_start : play back started 22:47:53.668 ipodsound.c > pjmedia_snd_stream_start : capture starting... 22:47:54.593 ipodsound.c > pjmedia_snd_stream_start : capture started... 22:47:54.593 ipodsound.c > pjmedia_snd_stream_get_info. 22:47:54.593 ipodsound.c > pjmedia_snd_stream_get_info. 22:47:54.593 ipodsound.c > pjmedia_snd_get_dev_info 0. 22:47:54.594 pjsua_call.c Making call with acc > #2 to (destination number) 22:47:54.595 pjsua_call.c Error initializing > media channel: Require secure session/transport (PJSIP_ESESSIONINSECURE) > [status=171142] 2010-02-24 22:47:55.088 SIP regid=2 dial (destination > number) 22:47:58.931 sound_port.c EC suspended because of inactivity > > > Can someone please describe what exactly the problem is and how to resolve > this. > Thanks in advance... > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: < > http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/attachments/20100225/a155c938/attachment-0001.html > > > > ------------------------------ > > Message: 4 > Date: Thu, 25 Feb 2010 16:40:31 +0530 > From: Premalatha Kuppan <premalatha@xxxxxxxxxxxx> > To: pjsip list <pjsip at lists.pjsip.org> > Subject: Service Based Features like CW, CF > Message-ID: > <b7d14e61002250310k6676621ci7636175d73c8c9c0 at mail.gmail.com> > Content-Type: text/plain; charset="iso-8859-1" > > Hi, > > Can PJSIP support service based features like Call waiting, call forwarding > ? > > Thanks, > Prem > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: < > http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/attachments/20100225/9bafd798/attachment-0001.html > > > > ------------------------------ > > Message: 5 > Date: Thu, 25 Feb 2010 13:45:54 +0200 > From: Kabil Akp?nar <kabilakpinar@xxxxxxxxx> > To: pjsip at lists.pjsip.org > Subject: Please Help, carrying TCP data with pjnath session > Message-ID: > <2e31597e1002250345u3197a608k2cfced650546f18f at mail.gmail.com> > Content-Type: text/plain; charset="utf-8" > > Hi All; > > What is your suggestion for carrying TCP data on tunnel opened by pjnath, > for example with ICE? > As it states on the documentation, any transport can be implemented on the > session layer, but normally any opened tunnel is based on UDP(almost). > > Waiting for your responses... > > Kabil Akp?nar > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: < > http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/attachments/20100225/0ed80e0f/attachment-0001.html > > > > ------------------------------ > > Message: 6 > Date: Thu, 25 Feb 2010 13:32:55 +0100 (CET) > From: "samuel.vinson" <samuelv@xxxxxxxxxxx> > To: pjsip list <pjsip at lists.pjsip.org> > Subject: Re: Calling issue on iPhone > Message-ID: <11698353.129.1267101175900.JavaMail.www at wwinf8401> > Content-Type: text/plain; charset="utf-8" > > Hello, > > It seems it's not an iPhone issue, because error message is: > "pjsua_call.c Error initializing media channel: Require secure > session/transport (PJSIP_ESESSIONINSECURE) [status=171142]" > > I believe you compile with srtp feature. You should investigate in this > part. > > Regards > > Samuel > > > > Message du 25/02/10 11:13 > > De : "tech guy" > > A : pjsip at lists.pjsip.org > > Copie ? : > > Objet : [pjsip] Calling issue on iPhone > > > > Hi All ? I have compiled the pjsip for iPhone. I got registered properly > but when I try to make a call, I m?getting the following error: ? > ipodsound.c pjmedia_snd_stream_start. 22:47:53.441 ipodsound.c > pjmedia_snd_stream_start : play back starting... 22:47:53.668 ipodsound.c > pjmedia_snd_stream_start : play back started 22:47:53.668 ipodsound.c > pjmedia_snd_stream_start : capture starting... 22:47:54.593 ipodsound.c > pjmedia_snd_stream_start : capture started... 22:47:54.593 ipodsound.c > pjmedia_snd_stream_get_info. 22:47:54.593 ipodsound.c > pjmedia_snd_stream_get_info. 22:47:54.593 ipodsound.c > pjmedia_snd_get_dev_info 0. 22:47:54.594 pjsua_call.c Making call with acc > #2 to (destination number) 22:47:54.595 pjsua_call.c Error initializing > media channel: Require secure session/transport (PJSIP_ESESSIONINSECURE) > [status=171142] 2010-02-24 22:47:55.088 SIP regid=2 dial (destination > number) 22:47:58.931 sound_port.c EC suspended because of inactivity ? ? Can > someone please describe what exactly the problem is and how to resolve this. > Thanks in advance... ? > > > [ (pas de nom de fichier) (0.2 Ko) ] > > Une messagerie gratuite, garantie ? vie et des services en plus, ?a vous > tente ? > Je cr?e ma bo?te mail www.laposte.net > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: < > http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/attachments/20100225/85deae66/attachment-0001.html > > > > ------------------------------ > > Message: 7 > Date: Thu, 25 Feb 2010 18:10:05 +0100 > From: Fabio Cherchi <fabio.cherchi@xxxxxxxx> > To: pjsip list <pjsip at lists.pjsip.org> > Subject: Re: jitter buffeer - reimplementation > Message-ID: <4B86AEED.6040701 at yahoo.it> > Content-Type: text/plain; charset="iso-8859-1"; Format="flowed" > > Hi Nik, > > You said that jitter buffer actually implemented in pjsip has a very bad > response when the jitter is continuosly changing and I'm experiencing > the same issue. > I would like to know if you had an improvement by applying the algorithm > you were talking about. > > Thank you in advance, > Fabio > > > nir elkayam ha scritto: > > hi all, > > > > I plan of implementing new jitter buffer, as the sound when working > > with pjsip is not so good, and I think that some of the problem > > related to the jitter buffer. > > I am tring to gather some info about that. attach 2 article about the > > subject. > > I have read them, and now implementing the new jitter buffer. > > > > Is anyone else facing trouble with the sound when using pjsip? mainly > > in cellular network where the jitter can vary very much? > > also, if anyone how has input on the subject, please let start some > > discussion here to get a better voice quiality, > > > > Ramjee paper on jitter buffer: > > > http://citeseerx.ist.psu.edu/viewdoc/download?doi=10.1.1.89.2199&rep=rep1&type=pdf > > < > http://citeseerx.ist.psu.edu/viewdoc/download?doi=10.1.1.89.2199&rep=rep1&type=pdf > > > > > > thanks, > > nir > > > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > 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/20100225/0aa50e2b/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 30, Issue 39 > ************************************* > -------------- next part -------------- An HTML attachment was scrubbed... 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