Hi bharat
kindly find the attachment
Regards
vivek
On Tue, May 12, 2009 at 11:57 AM, Bharat Yadav <
bharat.yadav at axisconvergence.com> wrote:
> Hi
>
> Please post your .mmp and config_site_sample.h
>
> Thanks
>
> Bharat
>
>
>
>
>
> *From:* pjsip-bounces at lists.pjsip.org [mailto:
> pjsip-bounces at lists.pjsip.org] *On Behalf Of *vivek shrivastava
> *Sent:* Tuesday, May 12, 2009 10:21 AM
>
> *To:* pjsip list
> *Subject:* Re: [pjsip] APS in PjSIP 1.1 APS Direct
>
>
>
> Hello bharat ,
>
> i am using Symbian_ua_gui and implemented the 2 files provided by you
> previously to enable
> G729 and desable the rest of codecs
>
>
> regards
> vivek
>
> On Tue, May 12, 2009 at 9:03 AM, Bharat Yadav <
> bharat.yadav at axisconvergence.com> wrote:
>
> Hi
>
> Please let me know which one u are using Symbian_ua or Symbian_ua_gui if
> you are using Symbian_ua then use only which one u need I mean donnt use
> software codec G711 just use APS G711 A/U and G729 because many sip
> providers and also most of TELECOM companies using on priority order of
> >G729>G711, actually we are also running sip based VoIP solutions, so please
> use G711A/U and G729 only, by disabling unwanted codec(s) there is no need
> to enable disable codec(s) by key pressing it will negotiate Codec on the
> behalf of SIP Server Settings because no one want to transcode codec(s) from
> GSM,SPEEX and AMR to G729 or G711, At last if u r using Symbian_ua_gui then
> no need to make any change in code just compile code with G711 and G729
> codec and all will be manage internally by Symbian_ua_gui for codec related
> concerns.
>
> Thanks
>
> Bharat
>
>
>
> *From:* pjsip-bounces at lists.pjsip.org [mailto:
> pjsip-bounces at lists.pjsip.org] *On Behalf Of *Srivatsan Deenadayalan
> *Sent:* Monday, May 11, 2009 9:34 PM
>
>
> *To:* pjsip list
> *Subject:* Re: [pjsip] APS in PjSIP 1.1 APS Direct
>
>
>
> Hi,
>
> Yes i have used all the codecs Supported by APS and PjSIP Software codecs.
>
> Here is the list of Codec i have used:
>
> 1. GSM
> 2. PCMA,
> 3. PCMU
> 4. SPEEX
> 5. AMR
> 6. G729
>
> How your enabling codes ? You need to use respective keypress events for
> enabling / disabling.
>
> Below segment from ua.cpp class shows how to enable and disable codec's.
> But prior to enabling any of the codes, the code registration have to be
> made. Don't worry all the codec registration will be done by PjSIP media
> itself.
>
> static void PrintMainMenu()
> {
> const char *menu =
> "\n\n"
> "Main Menu:\n"
> " *d Enable/disable codecs*\n"
> " m Call " SIP_DST_URI "\n"
> " a Answer call\n"
> " g Hangup all calls\n"
> " t Toggle audio route\n"
> #if !defined(PJMEDIA_CONF_USE_SWITCH_BOARD) ||
> PJMEDIA_CONF_USE_SWITCH_BOARD==0
> " j Toggle loopback audio\n"
> #endif
> "up/dn Increase/decrease output volume\n"
> " s Subscribe " SIP_DST_URI "\n"
> " S Unsubscribe presence\n"
> " o Set account online\n"
> " O Set account offline\n"
> " w Quit\n";
>
> PJ_LOG(3, (THIS_FILE, menu));
> }
>
> static void PrintCodecMenu()
> {
> const char *menu =
> "\n\n"
> "Codec Menu:\n"
> " a Enable all codecs\n"
> #if PJMEDIA_HAS_PASSTHROUGH_CODEC_AMR
> " d Enable only AMR\n"
> #endif
> #if PJMEDIA_HAS_PASSTHROUGH_CODEC_G729
> " g Enable only G.729\n"
> #endif
> #if PJMEDIA_HAS_PASSTHROUGH_CODEC_ILBC
> " j Enable only iLBC\n"
> #endif
> " m Enable only Speex\n"
> " p Enable only GSM\n"
> " t Enable only PCMU\n"
> " w Enable only PCMA\n";
>
> PJ_LOG(3, (THIS_FILE, menu));
> }
>
>
>
>
> vivek shrivastava wrote:
>
> hello Bharat ,
>
>
>
> I had followed your process for enabling G729 and while calling it is
> giveing Dialog
>
> *media type missing.*
>
> * *
> previously as by default g711 application was using and i was able to
> call . what else can i try ??
>
> * *Srivatsan did u managed to make the call for g729 . please guide me if
> possible
>
>
>
> Regards
>
>
>
>
>
>
>
>
>
>
>
>
> On 5/4/09, *Srivatsan Deenadayalan* <srivatsan at ongobiz.com> wrote:
>
> Thanks Bharat.. At last i am able to build and run PjSIP 1.1 APS Direct
> branch. The problem is with IDE (Carbide), Carbide 2.0 produces errors which
> i mentioned earlier. But same compiles in Carbide 1.2. I have tested all the
> enabling / disabling codes and works fine. If possible can u pls let me know
> which IDE you use ?
>
> Thanks once again for your patience in replying... :-)
>
> Bharat Yadav wrote:
>
> Hi
>
> Disable all codecs except your required mean G729 and G711A/U into
> /svn2668/pjmedia/include/pjmedia-codec/config.h, I amd using svn2668 and
> working fine. And this too
> /svn2668/pjmedia/include/pjmedia-audiodev/config.h I am including both file
> with their changed name like pjmedia-codec-config.h and
> pjmedia-audiodev-config.h .
>
> Letme know.
>
> Thanks
>
>
>
>
>
> *From:* pjsip-bounces at lists.pjsip.org [
> mailto:pjsip-bounces at lists.pjsip.org <pjsip-bounces at lists.pjsip.org>] *On
> Behalf Of *Srivatsan Deenadayalan
> *Sent:* Monday, May 04, 2009 6:21 PM
>
>
> *To:* pjsip list
> *Subject:* Re: [pjsip] APS in PjSIP 1.1 APS Direct
>
>
>
> Hi,
>
> Not succeeded yet.. I have just downloaded latest trunk version and
> compiled by changing .mmp file and including #define. But still the error
> remains the same. Have any one succeeded with latest trunk version APS
> Direct ?
>
> Bharat and Spider pls give a try with trunk version and let me know how it
> works.
>
> *Error :*
>
>
>
> pjsua_media.c:(.text+0x200): undefined reference to
>
> `pjmedia_codec_passthrough_init'
>
> \Symbian\9.2\S60_3rd_FP1\EPOC32\RELEASE\ARMV5\UREL\pjsua_lib.lib(pjsua_media.o):
>
> In function `pjsua_media_subsys_destroy':
>
> pjsua_media.c:(.text+0xb58): undefined reference to
>
> `pjmedia_codec_passthrough_deinit'
>
> \Symbian\9.2\S60_3rd_FP1\EPOC32\RELEASE\ARMV5\UREL\pjmedia_audiodev.lib(symb_aps_dev.o):
>
> In function `CPjAudioEngine::~CPjAudioEngine()':
>
> symb_aps_dev.cpp:(.text+0x80c): undefined reference to
>
> `RAPSSession::Close()'
>
> \Symbian\9.2\S60_3rd_FP1\EPOC32\RELEASE\ARMV5\UREL\pjmedia_audiodev.lib(symb_aps_dev.o):
>
> In function `CPjAudioEngine::~CPjAudioEngine()':
>
> symb_aps_dev.cpp:(.text+0x9a0): undefined reference to
>
> `RAPSSession::Close()'
>
> \Symbian\9.2\S60_3rd_FP1\EPOC32\RELEASE\ARMV5\UREL\pjmedia_audiodev.lib(symb_aps_dev.o):
>
> In function `CPjAudioEngine::~CPjAudioEngine()':
>
> symb_aps_dev.cpp:(.text+0xb30): undefined reference to
>
> `RAPSSession::Close()'
>
> \Symbian\9.2\S60_3rd_FP1\EPOC32\RELEASE\ARMV5\UREL\pjmedia_audiodev.lib(symb_aps_dev.o):
>
> In function `CPjAudioEngine::InitPlayL()':
>
> symb_aps_dev.cpp:(.text+0xc30): undefined reference to
>
> `RAPSSession::InitializePlayer(TAPSInitSettings&)'
>
> \Symbian\9.2\S60_3rd_FP1\EPOC32\RELEASE\ARMV5\UREL\pjmedia_audiodev.lib(symb_aps_dev.o):
>
> In function `CPjAudioEngine::InitRecL()':
>
> symb_aps_dev.cpp:(.text+0xd8c): undefined reference to
>
> `RAPSSession::InitializeRecorder(TAPSInitSettings&)'
>
> \Symbian\9.2\S60_3rd_FP1\EPOC32\RELEASE\ARMV5\UREL\pjmedia_audiodev.lib(symb_aps_dev.o):
>
> In function `CPjAudioEngine::Stop()':
>
> symb_aps_dev.cpp:(.text+0xfb0): undefined reference to `RAPSSession::Stop()'
>
> \Symbian\9.2\S60_3rd_FP1\EPOC32\RELEASE\ARMV5\UREL\pjmedia_audiodev.lib(symb_aps_dev.o):
>
> In function `CPjAudioEngine::ConstructL()':
>
>
>
> symb_aps_dev.cpp:(.text+0x106c): undefined reference to
>
> `RAPSSession::Connect()'
>
> symb_aps_dev.cpp:(.text+0x1098): undefined reference to
>
> `RAPSSession::SetCng(int)'
>
> symb_aps_dev.cpp:(.text+0x10a4): undefined reference to
>
> `RAPSSession::SetVadMode(int)'
>
> symb_aps_dev.cpp:(.text+0x10b0): undefined reference to
>
>
>
> `RAPSSession::SetPlc(int)'
>
> symb_aps_dev.cpp:(.text+0x10bc): undefined reference to
>
> `RAPSSession::SetEncoderMode(TAPSCodecMode)'
>
> symb_aps_dev.cpp:(.text+0x10c8): undefined reference to
>
> `RAPSSession::SetDecoderMode(TAPSCodecMode)'
>
> symb_aps_dev.cpp:(.text+0x10d4): undefined reference to
>
> `RAPSSession::ActivateLoudspeaker(int)'
>
> symb_aps_dev.cpp:(.text+0x10ec): undefined reference to
>
> `RAPSSession::Write()'
>
> symb_aps_dev.cpp:(.text+0x1118): undefined reference to
>
> `RAPSSession::Read()'
>
>
>
>
> spider wrote:
>
> On Mon, May 4, 2009 at 3:31 AM, Bharat Yadav
>
> <bharat.yadav at axisconvergence.com> <bharat.yadav at axisconvergence.com> wrote:
>
>
>
> Hi
>
>
>
> Ok you have add some plugins extensions from FP1 realease from here
>
> http://www.forum.nokia.com/info/sw.nokia.com/id/4ff42a22-7099-4cc9-91bf-5e66166bd28d/S60_3rd_SDK_FP1_API_Plug-In_Pack.html
>
>
>
> And extract it then select AudioProxyServer_v2.43.zip and MMFDevSoundAPI.zip
>
> and extract it to C:\Symbian\9.2\S60_3rd_FP1 then try to compile your
>
> project. Hope it should help you. (note: extract on
>
> C:\Symbian\9.2\S60_3rd_FP1 may ask for overwrite accept it with yes.)
>
>
>
> Thanks
>
>
>
> Bharat
>
>
>
> From: pjsip-bounces@xxxxxxxxxxxxxxx [mailto:pjsip-bounces at lists.pjsip.org <pjsip-bounces at lists.pjsip.org>]
>
> On Behalf Of Srivatsan Deenadayalan
>
> Sent: Monday, May 04, 2009 3:41 PM
>
>
>
> To: pjsip list
>
> Subject: Re: APS in PjSIP 1.1 APS Direct
>
>
>
>
>
>
>
> Thanks Bharat, but my problem not solved yet.
>
>
>
> 1. I am using PjSIP symbian console based app.
>
> 2. I have changed symbian_ua.mmp as given below,
>
> #define SND_HAS_APS 1
>
> #define SND_HAS_VAS 0
>
> #define SND_HAS_MDA 0
>
> 3. Included config_site.h file pjlib\include\pj
>
> 4. Added #define PJ_CONFIG_NOKIA_APS_DIRECT in config_site_sample.h found
>
> under pjlib\include\pj
>
> 5. Build - Phone Release GCCE,
>
>
>
> What should config_site.h file should have ? it should be empty ? i have
>
> added only #include <pj/config_site_sample.h> in config_site.h, anything
>
> else to be added ?
>
>
>
> I am using S60 Fp1 SDK with required plugin added and Carbide 2.0.
>
>
>
> Error :
>
>
>
> pjsua_media.c:(.text+0x200): undefined reference to
>
> `pjmedia_codec_passthrough_init'
>
> \Symbian\9.2\S60_3rd_FP1\EPOC32\RELEASE\ARMV5\UREL\pjsua_lib.lib(pjsua_media.o):
>
> In function `pjsua_media_subsys_destroy':
>
> pjsua_media.c:(.text+0xb58): undefined reference to
>
> `pjmedia_codec_passthrough_deinit'
>
> \Symbian\9.2\S60_3rd_FP1\EPOC32\RELEASE\ARMV5\UREL\pjmedia_audiodev.lib(symb_aps_dev.o):
>
> In function `CPjAudioEngine::~CPjAudioEngine()':
>
> symb_aps_dev.cpp:(.text+0x80c): undefined reference to
>
> `RAPSSession::Close()'
>
> \Symbian\9.2\S60_3rd_FP1\EPOC32\RELEASE\ARMV5\UREL\pjmedia_audiodev.lib(symb_aps_dev.o):
>
> In function `CPjAudioEngine::~CPjAudioEngine()':
>
> symb_aps_dev.cpp:(.text+0x9a0): undefined reference to
>
> `RAPSSession::Close()'
>
> \Symbian\9.2\S60_3rd_FP1\EPOC32\RELEASE\ARMV5\UREL\pjmedia_audiodev.lib(symb_aps_dev.o):
>
> In function `CPjAudioEngine::~CPjAudioEngine()':
>
> symb_aps_dev.cpp:(.text+0xb30): undefined reference to
>
> `RAPSSession::Close()'
>
> \Symbian\9.2\S60_3rd_FP1\EPOC32\RELEASE\ARMV5\UREL\pjmedia_audiodev.lib(symb_aps_dev.o):
>
> In function `CPjAudioEngine::InitPlayL()':
>
> symb_aps_dev.cpp:(.text+0xc30): undefined reference to
>
> `RAPSSession::InitializePlayer(TAPSInitSettings&)'
>
> \Symbian\9.2\S60_3rd_FP1\EPOC32\RELEASE\ARMV5\UREL\pjmedia_audiodev.lib(symb_aps_dev.o):
>
> In function `CPjAudioEngine::InitRecL()':
>
> symb_aps_dev.cpp:(.text+0xd8c): undefined reference to
>
> `RAPSSession::InitializeRecorder(TAPSInitSettings&)'
>
> \Symbian\9.2\S60_3rd_FP1\EPOC32\RELEASE\ARMV5\UREL\pjmedia_audiodev.lib(symb_aps_dev.o):
>
> In function `CPjAudioEngine::Stop()':
>
> symb_aps_dev.cpp:(.text+0xfb0): undefined reference to `RAPSSession::Stop()'
>
> \Symbian\9.2\S60_3rd_FP1\EPOC32\RELEASE\ARMV5\UREL\pjmedia_audiodev.lib(symb_aps_dev.o):
>
> In function `CPjAudioEngine::ConstructL()':
>
>
>
> symb_aps_dev.cpp:(.text+0x106c): undefined reference to
>
> `RAPSSession::Connect()'
>
> symb_aps_dev.cpp:(.text+0x1098): undefined reference to
>
> `RAPSSession::SetCng(int)'
>
> symb_aps_dev.cpp:(.text+0x10a4): undefined reference to
>
> `RAPSSession::SetVadMode(int)'
>
> symb_aps_dev.cpp:(.text+0x10b0): undefined reference to
>
>
>
> `RAPSSession::SetPlc(int)'
>
> symb_aps_dev.cpp:(.text+0x10bc): undefined reference to
>
> `RAPSSession::SetEncoderMode(TAPSCodecMode)'
>
> symb_aps_dev.cpp:(.text+0x10c8): undefined reference to
>
> `RAPSSession::SetDecoderMode(TAPSCodecMode)'
>
> symb_aps_dev.cpp:(.text+0x10d4): undefined reference to
>
> `RAPSSession::ActivateLoudspeaker(int)'
>
> symb_aps_dev.cpp:(.text+0x10ec): undefined reference to
>
> `RAPSSession::Write()'
>
> symb_aps_dev.cpp:(.text+0x1118): undefined reference to
>
> `RAPSSession::Read()'
>
>
>
>
>
>
>
> Bharat Yadav wrote:
>
>
>
>
>
> Hi
>
>
>
> Check symbian_ua and follow their .mmp and next include config_site.h and
>
>
>
> define #define PJ_CONFIG_NOKIA_APS_DIRECT in config_site_sample.h located in
>
>
>
> your_project\pjlib\include\pj now your project APS Direct enabled.
>
>
>
> Thanks
>
>
>
> Bharat
>
>
>
>
>
>
>
> -----Original Message-----
>
>
>
> From: pjsip-bounces@xxxxxxxxxxxxxxx [mailto:pjsip-bounces at lists.pjsip.org <pjsip-bounces at lists.pjsip.org>]
>
>
>
> On Behalf Of Srivatsan Deenadayalan
>
>
>
> Sent: Monday, May 04, 2009 2:43 PM
>
>
>
> To: pjsip list
>
>
>
> Subject: Re: APS in PjSIP 1.1 APS Direct
>
>
>
>
>
>
>
> UPDATE:
>
>
>
>
>
>
>
> My question is regarding Symbian OS
>
>
>
>
>
>
>
> Question :
>
>
>
>
>
>
>
>
>
> How to enable APS in PjSIP 1.1 APS direct ? Can anyone please provide
>
>
>
> the details of Macros and others to be set to activate APS. I have tried
>
>
>
> by enabling #define SND_HAS_APS 1 in .mmp file but nothing happens. Is
>
>
>
> their any thing else to be set to use APS in PjSIP 1.1 and trunk versions ?
>
>
>
>
>
>
>
> Srivatsan Deenadayalan wrote:
>
>
>
>
>
>
>
> Hi All,
>
>
>
>
>
>
>
> How to enable APS in PjSIP 1.1 APS direct ? Can anyone please provide
>
>
>
> the details of Macros and others to be set to activate APS. I have
>
>
>
>
>
> tried by enabling #define SND_HAS_APS 1 in .mmp file but nothing
>
>
>
> happens. Is their any thing else to be set to use APS in PjSIP 1.1 and
>
>
>
> trunk versions ?
>
>
>
>
>
>
>
>
>
> Pls help me...
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
> --
>
>
>
> Thanks,
>
>
>
> Srivatsan.D,
>
>
>
>
>
>
>
> _______________________________________________
>
> 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
>
>
>
>
>
>
>
>
>
> try to press up/down arrow after call connected to switch audio to airpiece
>
>
>
> _______________________________________________
>
> 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
>
>
>
>
>
> --
>
> Thanks,
>
> Srivatsan.D,
>
> ------------------------------
>
>
>
>
>
>
>
> _______________________________________________
>
> 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
>
>
>
> --
>
> Thanks,
>
> Srivatsan.D,
>
>
>
>
>
>
>
>
> _______________________________________________
> 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
>
>
>
>
>
> --
>
> Thanks,
>
> Srivatsan.D
>
>
> _______________________________________________
> 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
>
>
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