pjsip Digest, Vol 30, Issue 39

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Hello,

You can disable ssl support too with this option : --disable-ssl

Samuel

Le 26/02/10 08:59, tech guy a ?crit :
> 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 
> <mailto: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
>     <mailto:pjsip-request at lists.pjsip.org>> wrote:
>
>         Send pjsip mailing list submissions to
>         pjsip at lists.pjsip.org <mailto:pjsip at lists.pjsip.org>
>
>         To subscribe or unsubscribe via the World Wide Web, visit
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>         or, via email, send a message with subject or body 'help' to
>         pjsip-request at lists.pjsip.org
>         <mailto:pjsip-request at lists.pjsip.org>
>
>         You can reach the person managing the list at
>         pjsip-owner at lists.pjsip.org <mailto: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
>         <mailto:ricardo2500 at hotmail.com>>
>         To: <pjsip at lists.pjsip.org <mailto: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 <http://www.vivirmessenger.com>
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>
>         ------------------------------
>
>         Message: 2
>         Date: Thu, 25 Feb 2010 16:07:17 +0700
>         From: raksa chao <chao.raksa@xxxxxxxxx
>         <mailto:chao.raksa at gmail.com>>
>         To: pjsip at lists.pjsip.org <mailto:pjsip at lists.pjsip.org>
>         Subject: I wanna know about Multi call?
>         Message-ID:
>         <1f8c401002250107w55f59d6fg5784e6b4b6d84e01 at mail.gmail.com
>         <mailto: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,,
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>
>         ------------------------------
>
>         Message: 3
>         Date: Thu, 25 Feb 2010 15:43:10 +0530
>         From: tech guy <techguy244@xxxxxxxxx
>         <mailto:techguy244 at gmail.com>>
>         To: pjsip at lists.pjsip.org <mailto:pjsip at lists.pjsip.org>
>         Subject: Calling issue on iPhone
>         Message-ID:
>         <951e50ce1002250213h1ed21b5eh3cd18d706d6adb6 at mail.gmail.com
>         <mailto: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...
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>
>         ------------------------------
>
>         Message: 4
>         Date: Thu, 25 Feb 2010 16:40:31 +0530
>         From: Premalatha Kuppan <premalatha@xxxxxxxxxxxx
>         <mailto:premalatha at ngintech.com>>
>         To: pjsip list <pjsip at lists.pjsip.org
>         <mailto:pjsip at lists.pjsip.org>>
>         Subject:  Service Based Features like CW, CF
>         Message-ID:
>         <b7d14e61002250310k6676621ci7636175d73c8c9c0 at mail.gmail.com
>         <mailto: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
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>
>         ------------------------------
>
>         Message: 5
>         Date: Thu, 25 Feb 2010 13:45:54 +0200
>         From: Kabil Akp?nar <kabilakpinar@xxxxxxxxx
>         <mailto:kabilakpinar at gmail.com>>
>         To: pjsip at lists.pjsip.org <mailto:pjsip at lists.pjsip.org>
>         Subject: Please Help, carrying TCP data with pjnath
>         session
>         Message-ID:
>         <2e31597e1002250345u3197a608k2cfced650546f18f at mail.gmail.com
>         <mailto: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
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>
>         ------------------------------
>
>         Message: 6
>         Date: Thu, 25 Feb 2010 13:32:55 +0100 (CET)
>         From: "samuel.vinson" <samuelv@xxxxxxxxxxx
>         <mailto:samuelv at laposte.net>>
>         To: pjsip list <pjsip at lists.pjsip.org
>         <mailto: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 <mailto: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 <http://www.laposte.net>
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>
>         ------------------------------
>
>         Message: 7
>         Date: Thu, 25 Feb 2010 18:10:05 +0100
>         From: Fabio Cherchi <fabio.cherchi@xxxxxxxx
>         <mailto:fabio.cherchi at yahoo.it>>
>         To: pjsip list <pjsip at lists.pjsip.org
>         <mailto:pjsip at lists.pjsip.org>>
>         Subject: Re: jitter buffeer - reimplementation
>         Message-ID: <4B86AEED.6040701 at yahoo.it
>         <mailto: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>
>         >
>         <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 <mailto:pjsip at lists.pjsip.org>
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>         >
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>         ------------------------------
>
>         _______________________________________________
>         pjsip mailing list
>         pjsip at lists.pjsip.org <mailto:pjsip at lists.pjsip.org>
>         http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
>
>
>         End of pjsip Digest, Vol 30, Issue 39
>         *************************************
>
>
>
>
> _______________________________________________
> Visit our blog: http://blog.pjsip.org
>
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