Hi Samuel, By mistake There is a wrong path of config_site.h file which i mentioned. SRTP is disabled in pjlib/include/pj/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 Thanks On Fri, Feb 26, 2010 at 12:54 PM, tech guy <techguy244 at gmail.com> wrote: > > 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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