Thanks

Hi Benny, I am trying to add voice-age g729 codec with PJSIP for symbian. but i am getting some difficulties while following the gsm.c file. 1. while creating g729 codec private data what should i declare in replace of // struct gsm_state *encoder; // struct gsm_state *decoder; in "l16.c" its only use the frame size .. 2. in Generate default attribute. i have changed the value to this. is it ok? attr->info.clock_rate = 8000; attr->info.channel_cnt = 1; attr->info.avg_bps = 8000; attr->info.max_bps = 8000; attr->info.pcm_bits_per_sample = 16; attr->info.frm_ptime = 10; 3. In gsm.c there is some hard coding in *gsm_codec_parse(* -- what will be the value in replace of "33" here 4. frame_size and sample_per_frame are also not clear to me.. Sorry for lot of questions and may be those are lot easier to ask.. but really i am not good in codec.. waiting for reply. Thanks S. M. Nazmul Hasan Opu Software Engineer R & D Application Dhaka, Bangladesh Mob: +880 1712 901 764

      Thanks

Well.. your attribute settings seems fine. For the 33 it will be 20. And in place of 320, it will be 160.. hope this will give u a jump start.. Rawshan Iajdani From: pjsip-bounces@lists.pjsip.org [mailto:pjsip-bounces@lists.pjsip.org] On Behalf Of S. M. Nazmul Hasan (Opu) Sent: Wednesday, October 29, 2008 6:12 PM To: pjsip list Subject: [pjsip] g729 codec Hi Benny, I am trying to add voice-age g729 codec with PJSIP for symbian. but i am getting some difficulties while following the gsm.c file. 1. while creating g729 codec private data what should i declare in replace of // struct gsm_state *encoder; // struct gsm_state *decoder; in "l16.c" its only use the frame size .. 2. in Generate default attribute. i have changed the value to this. is it ok? attr->info.clock_rate = 8000; attr->info.channel_cnt = 1; attr->info.avg_bps = 8000; attr->info.max_bps = 8000; attr->info.pcm_bits_per_sample = 16; attr->info.frm_ptime = 10; 3. In gsm.c there is some hard coding in gsm_codec_parse( -- what will be the value in replace of "33" here 4. frame_size and sample_per_frame are also not clear to me.. Sorry for lot of questions and may be those are lot easier to ask.. but really i am not good in codec.. waiting for reply. Thanks S. M. Nazmul Hasan Opu Software Engineer R & D Application Dhaka, Bangladesh Mob: +880 1712 901 764

        pj_str_t codec_id = pj_str("g729");
        pjmedia_codec_mgr_set_codec_priority(
            pjmedia_endpt_get_codec_mgr(pjsua_var.med_endpt),
            &codec_id, PJMEDIA_CODEC_PRIO_HIGHEST);

       Thanks

Thanks Lajdani. My jumped start is running well. I am using Brekeke sip server to check the active session. and i am getting the using codec (payload) is always *PCMU/8000*. even after changing the g729 for highest priority. pj_str_t codec_id = pj_str("g729"); pjmedia_codec_mgr_set_codec_priority( pjmedia_endpt_get_codec_mgr(pjsua_var.med_endpt), &codec_id, PJMEDIA_CODEC_PRIO_HIGHEST); when i changed it to *PJMEDIA_CODEC_PRIO_LOWEST *the result was same. even after disabling all the codecs and after deleting all g711 files and related fields it showed the payload is *PCMU/8000*. How can i easily disable all the codecs except g729. Thanks On Thu, Oct 30, 2008 at 3:43 AM, Rawshan Iajdani <iajdani@provati.com>wrote: > Well.. your attribute settings seems fine. For the 33 it will be 20. And > in place of 320, it will be 160.. hope this will give u a jump start.. > > > > > > > > Rawshan Iajdani > > > > > > *From:* pjsip-bounces@lists.pjsip.org [mailto: > pjsip-bounces@lists.pjsip.org] *On Behalf Of *S. M. Nazmul Hasan (Opu) > *Sent:* Wednesday, October 29, 2008 6:12 PM > *To:* pjsip list > *Subject:* [pjsip] g729 codec > > > > > Hi Benny, > > I am trying to add voice-age g729 codec with PJSIP for symbian. but i am > getting some difficulties while following the gsm.c file. > > 1. while creating g729 codec private data > what should i declare in replace of > // struct gsm_state *encoder; > // struct gsm_state *decoder; > > in "l16.c" its only use the frame size .. > > 2. in Generate default attribute. > > i have changed the value to this. is it ok? > attr->info.clock_rate = 8000; > attr->info.channel_cnt = 1; > attr->info.avg_bps = 8000; > attr->info.max_bps = 8000; > attr->info.pcm_bits_per_sample = 16; > attr->info.frm_ptime = 10; > > 3. In gsm.c there is some hard coding in *gsm_codec_parse(* > -- what will be the value in replace of "33" here > > 4. frame_size and sample_per_frame are also not clear to me.. > > Sorry for lot of questions and may be those are lot easier to ask.. but > really i am not good in codec.. > > waiting for reply. > > Thanks > > S. M. Nazmul Hasan Opu > Software Engineer > R & D Application > Dhaka, Bangladesh > Mob: +880 1712 901 764 > > _______________________________________________ > Visit our blog: http://blog.pjsip.org > > pjsip mailing list > pjsip@lists.pjsip.org > http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org > > -- S. M. Nazmul Hasan Opu Software Engineer R & D Application Dhaka, Bangladesh Mob: +880 1712 901 764

        pj_str_t codec_id = pj_str("g729");
        pjmedia_codec_mgr_set_codec_priority( 
            pjmedia_endpt_get_codec_mgr(pjsua_var.med_endpt),
            &codec_id, PJMEDIA_CODEC_PRIO_HIGHEST);

      Thanks

You should be able to do that in pjlib/include/pj/config_site.h and pjmedia/include/pjmedia-codec/config.h file. But always remember config_site.h settings will overwrite any settings done in config.h file. Good luck. From: pjsip-bounces@lists.pjsip.org [mailto:pjsip-bounces@lists.pjsip.org] On Behalf Of S. M. Nazmul Hasan (Opu) Sent: Saturday, November 01, 2008 6:01 PM To: pjsip list Subject: Re: [pjsip] g729 codec Thanks Lajdani. My jumped start is running well. I am using Brekeke sip server to check the active session. and i am getting the using codec (payload) is always PCMU/8000. even after changing the g729 for highest priority. pj_str_t codec_id = pj_str("g729"); pjmedia_codec_mgr_set_codec_priority( pjmedia_endpt_get_codec_mgr(pjsua_var.med_endpt), &codec_id, PJMEDIA_CODEC_PRIO_HIGHEST); when i changed it to PJMEDIA_CODEC_PRIO_LOWEST the result was same. even after disabling all the codecs and after deleting all g711 files and related fields it showed the payload is PCMU/8000. How can i easily disable all the codecs except g729. Thanks On Thu, Oct 30, 2008 at 3:43 AM, Rawshan Iajdani <iajdani@provati.com> wrote: Well.. your attribute settings seems fine. For the 33 it will be 20. And in place of 320, it will be 160.. hope this will give u a jump start.. Rawshan Iajdani From: pjsip-bounces@lists.pjsip.org [mailto:pjsip-bounces@lists.pjsip.org] On Behalf Of S. M. Nazmul Hasan (Opu) Sent: Wednesday, October 29, 2008 6:12 PM To: pjsip list Subject: [pjsip] g729 codec Hi Benny, I am trying to add voice-age g729 codec with PJSIP for symbian. but i am getting some difficulties while following the gsm.c file. 1. while creating g729 codec private data what should i declare in replace of // struct gsm_state *encoder; // struct gsm_state *decoder; in "l16.c" its only use the frame size .. 2. in Generate default attribute. i have changed the value to this. is it ok? attr->info.clock_rate = 8000; attr->info.channel_cnt = 1; attr->info.avg_bps = 8000; attr->info.max_bps = 8000; attr->info.pcm_bits_per_sample = 16; attr->info.frm_ptime = 10; 3. In gsm.c there is some hard coding in gsm_codec_parse( -- what will be the value in replace of "33" here 4. frame_size and sample_per_frame are also not clear to me.. Sorry for lot of questions and may be those are lot easier to ask.. but really i am not good in codec.. waiting for reply. Thanks S. M. Nazmul Hasan Opu Software Engineer R & D Application Dhaka, Bangladesh Mob: +880 1712 901 764 _______________________________________________ Visit our blog: http://blog.pjsip.org pjsip mailing list pjsip@lists.pjsip.org http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org -- S. M. Nazmul Hasan Opu Software Engineer R & D Application Dhaka, Bangladesh Mob: +880 1712 901 764

         pj_str_t codec_id = pj_str("g729");
         pjmedia_codec_mgr_set_codec_priority(
             pjmedia_endpt_get_codec_mgr(pjsua_var.med_endpt),
             &codec_id, PJMEDIA_CODEC_PRIO_HIGHEST);

       Thanks

Dear Rawshan, Can you please check the file if it is ok to add voiceage g729 in pjsip for symbian. i am trying this for several days but i can get it working for me. Help from anyone would be great appretiable. Thanks On Mon, Nov 3, 2008 at 3:17 PM, Rawshan Iajdani <iajdani@provati.com> wrote: > You should be able to do that in pjlib/include/pj/config_site.h and > pjmedia/include/pjmedia-codec/config.h file. But always remember > config_site.h settings will overwrite any settings done in config.h file. > Good luck… > > > > > > > > > > *From:* pjsip-bounces@lists.pjsip.org [mailto: > pjsip-bounces@lists.pjsip.org] *On Behalf Of *S. M. Nazmul Hasan (Opu) > *Sent:* Saturday, November 01, 2008 6:01 PM > *To:* pjsip list > *Subject:* Re: [pjsip] g729 codec > > > > Thanks Lajdani. My jumped start is running well. > > I am using Brekeke sip server to check the active session. and i am > getting the using codec (payload) is always *PCMU/8000*. even after > changing the g729 for highest priority. > > pj_str_t codec_id = pj_str("g729"); > pjmedia_codec_mgr_set_codec_priority( > pjmedia_endpt_get_codec_mgr(pjsua_var.med_endpt), > &codec_id, PJMEDIA_CODEC_PRIO_HIGHEST); > > when i changed it to *PJMEDIA_CODEC_PRIO_LOWEST *the result was same. > even after disabling all the codecs and after deleting all g711 files and > related fields it showed the payload is *PCMU/8000*. > > How can i easily disable all the codecs except g729. > > Thanks > > > > On Thu, Oct 30, 2008 at 3:43 AM, Rawshan Iajdani <iajdani@provati.com> > wrote: > > Well.. your attribute settings seems fine. For the 33 it will be 20. And in > place of 320, it will be 160.. hope this will give u a jump start.. > > > > > > > > Rawshan Iajdani > > > > > > *From:* pjsip-bounces@lists.pjsip.org [mailto: > pjsip-bounces@lists.pjsip.org] *On Behalf Of *S. M. Nazmul Hasan (Opu) > *Sent:* Wednesday, October 29, 2008 6:12 PM > *To:* pjsip list > *Subject:* [pjsip] g729 codec > > > > > Hi Benny, > > I am trying to add voice-age g729 codec with PJSIP for symbian. but i am > getting some difficulties while following the gsm.c file. > > 1. while creating g729 codec private data > what should i declare in replace of > // struct gsm_state *encoder; > // struct gsm_state *decoder; > > in "l16.c" its only use the frame size .. > > 2. in Generate default attribute. > > i have changed the value to this. is it ok? > attr->info.clock_rate = 8000; > attr->info.channel_cnt = 1; > attr->info.avg_bps = 8000; > attr->info.max_bps = 8000; > attr->info.pcm_bits_per_sample = 16; > attr->info.frm_ptime = 10; > > 3. In gsm.c there is some hard coding in *gsm_codec_parse(* > -- what will be the value in replace of "33" here > > 4. frame_size and sample_per_frame are also not clear to me.. > > Sorry for lot of questions and may be those are lot easier to ask.. but > really i am not good in codec.. > > waiting for reply. > > Thanks > > S. M. Nazmul Hasan Opu > Software Engineer > R & D Application > Dhaka, Bangladesh > Mob: +880 1712 901 764 > > > _______________________________________________ > Visit our blog: http://blog.pjsip.org > > pjsip mailing list > pjsip@lists.pjsip.org > http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org > > > > > -- > S. M. Nazmul Hasan Opu > Software Engineer > R & D Application > Dhaka, Bangladesh > Mob: +880 1712 901 764 > > _______________________________________________ > Visit our blog: http://blog.pjsip.org > > pjsip mailing list > pjsip@lists.pjsip.org > http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org > > -- S. M. Nazmul Hasan Opu Software Engineer R & D Application Dhaka, Bangladesh Mob: +880 1712 901 764

         pj_str_t codec_id = pj_str("g729");
         pjmedia_codec_mgr_set_codec_priority(
             pjmedia_endpt_get_codec_mgr(pjsua_var.med_endpt),
             &codec_id, PJMEDIA_CODEC_PRIO_HIGHEST);

       Thanks

Hi, Just in case, please make sure that your codec is registered to the pjmedia endpoint, as you can see in pjsua_media.c there is pjmedia_codec_speex_init() for registering Speex. And please remember to unregister it. Regards, nanang On Sat, Nov 1, 2008 at 7:01 PM, S. M. Nazmul Hasan (Opu) <apus29@gmail.com> wrote: > Thanks Lajdani. My jumped start is running well. > > I am using Brekeke sip server to check the active session. and i am getting > the using codec (payload) is always PCMU/8000. even after changing the g729 > for highest priority. > > pj_str_t codec_id = pj_str("g729"); > pjmedia_codec_mgr_set_codec_priority( > pjmedia_endpt_get_codec_mgr(pjsua_var.med_endpt), > &codec_id, PJMEDIA_CODEC_PRIO_HIGHEST); > > when i changed it to PJMEDIA_CODEC_PRIO_LOWEST the result was same. even > after disabling all the codecs and after deleting all g711 files and related > fields it showed the payload is PCMU/8000. > > How can i easily disable all the codecs except g729. > > Thanks > > > > > On Thu, Oct 30, 2008 at 3:43 AM, Rawshan Iajdani <iajdani@provati.com> > wrote: >> >> Well.. your attribute settings seems fine. For the 33 it will be 20. And >> in place of 320, it will be 160.. hope this will give u a jump start.. >> >> >> >> >> >> >> >> Rawshan Iajdani >> >> >> >> >> >> From: pjsip-bounces@lists.pjsip.org [mailto:pjsip-bounces@lists.pjsip.org] >> On Behalf Of S. M. Nazmul Hasan (Opu) >> Sent: Wednesday, October 29, 2008 6:12 PM >> To: pjsip list >> Subject: [pjsip] g729 codec >> >> >> >> Hi Benny, >> >> I am trying to add voice-age g729 codec with PJSIP for symbian. but i am >> getting some difficulties while following the gsm.c file. >> >> 1. while creating g729 codec private data >> what should i declare in replace of >> // struct gsm_state *encoder; >> // struct gsm_state *decoder; >> >> in "l16.c" its only use the frame size .. >> >> 2. in Generate default attribute. >> >> i have changed the value to this. is it ok? >> attr->info.clock_rate = 8000; >> attr->info.channel_cnt = 1; >> attr->info.avg_bps = 8000; >> attr->info.max_bps = 8000; >> attr->info.pcm_bits_per_sample = 16; >> attr->info.frm_ptime = 10; >> >> 3. In gsm.c there is some hard coding in gsm_codec_parse( >> -- what will be the value in replace of "33" here >> >> 4. frame_size and sample_per_frame are also not clear to me.. >> >> Sorry for lot of questions and may be those are lot easier to ask.. but >> really i am not good in codec.. >> >> waiting for reply. >> >> Thanks >> >> S. M. Nazmul Hasan Opu >> Software Engineer >> R & D Application >> Dhaka, Bangladesh >> Mob: +880 1712 901 764 >> >> _______________________________________________ >> Visit our blog: http://blog.pjsip.org >> >> pjsip mailing list >> pjsip@lists.pjsip.org >> http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org >> > > > > -- > S. M. Nazmul Hasan Opu > Software Engineer > R & D Application > Dhaka, Bangladesh > Mob: +880 1712 901 764 > > _______________________________________________ > Visit our blog: http://blog.pjsip.org > > pjsip mailing list > pjsip@lists.pjsip.org > http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org > >

         pj_str_t codec_id = pj_str("g729");
         pjmedia_codec_mgr_set_codec_priority(
             pjmedia_endpt_get_codec_mgr(pjsua_var.med_endpt),
             &codec_id, PJMEDIA_CODEC_PRIO_HIGHEST);

       Thanks

Hi, Just did a quick skimming, things I can found so far: 1. For 8000kbps and 10ms frame time, the encoded frame size should be 10 bytes, not 20 bytes. 2. this line: pcm_in += 160; should be: pcm_in += 80; Numbers in a codec wrapper are very important, so please check and recheck carefully. Regards, nanang On Tue, Nov 4, 2008 at 8:42 PM, S. M. Nazmul Hasan (Opu) <apus29@gmail.com> wrote: > Dear Rawshan, > > Can you please check the file if it is ok to add voiceage g729 in pjsip for > symbian. i am trying this for several days but i can get it working for me. > > Help from anyone would be great appretiable. > > Thanks > > On Mon, Nov 3, 2008 at 3:17 PM, Rawshan Iajdani <iajdani@provati.com> wrote: >> >> You should be able to do that in pjlib/include/pj/config_site.h and >> pjmedia/include/pjmedia-codec/config.h file. But always remember >> config_site.h settings will overwrite any settings done in config.h file. >> Good luck… >> >> >> >> >> >> >> >> >> >> From: pjsip-bounces@lists.pjsip.org [mailto:pjsip-bounces@lists.pjsip.org] >> On Behalf Of S. M. Nazmul Hasan (Opu) >> Sent: Saturday, November 01, 2008 6:01 PM >> To: pjsip list >> Subject: Re: [pjsip] g729 codec >> >> >> >> Thanks Lajdani. My jumped start is running well. >> >> I am using Brekeke sip server to check the active session. and i am >> getting the using codec (payload) is always PCMU/8000. even after changing >> the g729 for highest priority. >> >> pj_str_t codec_id = pj_str("g729"); >> pjmedia_codec_mgr_set_codec_priority( >> pjmedia_endpt_get_codec_mgr(pjsua_var.med_endpt), >> &codec_id, PJMEDIA_CODEC_PRIO_HIGHEST); >> >> when i changed it to PJMEDIA_CODEC_PRIO_LOWEST the result was same. even >> after disabling all the codecs and after deleting all g711 files and related >> fields it showed the payload is PCMU/8000. >> >> How can i easily disable all the codecs except g729. >> >> Thanks >> >> >> >> On Thu, Oct 30, 2008 at 3:43 AM, Rawshan Iajdani <iajdani@provati.com> >> wrote: >> >> Well.. your attribute settings seems fine. For the 33 it will be 20. And >> in place of 320, it will be 160.. hope this will give u a jump start.. >> >> >> >> >> >> >> >> Rawshan Iajdani >> >> >> >> >> >> From: pjsip-bounces@lists.pjsip.org [mailto:pjsip-bounces@lists.pjsip.org] >> On Behalf Of S. M. Nazmul Hasan (Opu) >> Sent: Wednesday, October 29, 2008 6:12 PM >> To: pjsip list >> Subject: [pjsip] g729 codec >> >> >> >> Hi Benny, >> >> I am trying to add voice-age g729 codec with PJSIP for symbian. but i am >> getting some difficulties while following the gsm.c file. >> >> 1. while creating g729 codec private data >> what should i declare in replace of >> // struct gsm_state *encoder; >> // struct gsm_state *decoder; >> >> in "l16.c" its only use the frame size .. >> >> 2. in Generate default attribute. >> >> i have changed the value to this. is it ok? >> attr->info.clock_rate = 8000; >> attr->info.channel_cnt = 1; >> attr->info.avg_bps = 8000; >> attr->info.max_bps = 8000; >> attr->info.pcm_bits_per_sample = 16; >> attr->info.frm_ptime = 10; >> >> 3. In gsm.c there is some hard coding in gsm_codec_parse( >> -- what will be the value in replace of "33" here >> >> 4. frame_size and sample_per_frame are also not clear to me.. >> >> Sorry for lot of questions and may be those are lot easier to ask.. but >> really i am not good in codec.. >> >> waiting for reply. >> >> Thanks >> >> S. M. Nazmul Hasan Opu >> Software Engineer >> R & D Application >> Dhaka, Bangladesh >> Mob: +880 1712 901 764 >> >> _______________________________________________ >> Visit our blog: http://blog.pjsip.org >> >> pjsip mailing list >> pjsip@lists.pjsip.org >> http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org >> >> >> -- >> S. M. Nazmul Hasan Opu >> Software Engineer >> R & D Application >> Dhaka, Bangladesh >> Mob: +880 1712 901 764 >> >> _______________________________________________ >> Visit our blog: http://blog.pjsip.org >> >> pjsip mailing list >> pjsip@lists.pjsip.org >> http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org >> > > > > -- > S. M. Nazmul Hasan Opu > Software Engineer > R & D Application > Dhaka, Bangladesh > Mob: +880 1712 901 764 > > _______________________________________________ > Visit our blog: http://blog.pjsip.org > > pjsip mailing list > pjsip@lists.pjsip.org > http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org > >

         pj_str_t codec_id = pj_str("g729");
         pjmedia_codec_mgr_set_codec_priority(
             pjmedia_endpt_get_codec_mgr(pjsua_var.med_endpt),
             &codec_id, PJMEDIA_CODEC_PRIO_HIGHEST);

       Thanks

Nanag, I think so Nanang, may be its not registering because i am getting response 488(Not acceptable here). But i did it register and unregister in pjsua_media.c. Can you please see these files. Thanks Opu On Tue, Nov 4, 2008 at 10:18 PM, Nanang Izzuddin <nanang@pjsip.org> wrote: > Hi, > > Just did a quick skimming, things I can found so far: > 1. For 8000kbps and 10ms frame time, the encoded frame size should be > 10 bytes, not 20 bytes. > 2. this line: > pcm_in += 160; > should be: > pcm_in += 80; > > Numbers in a codec wrapper are very important, so please check and > recheck carefully. > > Regards, > nanang > > > On Tue, Nov 4, 2008 at 8:42 PM, S. M. Nazmul Hasan (Opu) > <apus29@gmail.com> wrote: > > Dear Rawshan, > > > > Can you please check the file if it is ok to add voiceage g729 in pjsip > for > > symbian. i am trying this for several days but i can get it working for > me. > > > > Help from anyone would be great appretiable. > > > > Thanks > > > > On Mon, Nov 3, 2008 at 3:17 PM, Rawshan Iajdani <iajdani@provati.com> > wrote: > >> > >> You should be able to do that in pjlib/include/pj/config_site.h and > >> pjmedia/include/pjmedia-codec/config.h file. But always remember > >> config_site.h settings will overwrite any settings done in config.h > file. > >> Good luck… > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> From: pjsip-bounces@lists.pjsip.org [mailto: > pjsip-bounces@lists.pjsip.org] > >> On Behalf Of S. M. Nazmul Hasan (Opu) > >> Sent: Saturday, November 01, 2008 6:01 PM > >> To: pjsip list > >> Subject: Re: [pjsip] g729 codec > >> > >> > >> > >> Thanks Lajdani. My jumped start is running well. > >> > >> I am using Brekeke sip server to check the active session. and i am > >> getting the using codec (payload) is always PCMU/8000. even after > changing > >> the g729 for highest priority. > >> > >> pj_str_t codec_id = pj_str("g729"); > >> pjmedia_codec_mgr_set_codec_priority( > >> pjmedia_endpt_get_codec_mgr(pjsua_var.med_endpt), > >> &codec_id, PJMEDIA_CODEC_PRIO_HIGHEST); > >> > >> when i changed it to PJMEDIA_CODEC_PRIO_LOWEST the result was same. > even > >> after disabling all the codecs and after deleting all g711 files and > related > >> fields it showed the payload is PCMU/8000. > >> > >> How can i easily disable all the codecs except g729. > >> > >> Thanks > >> > >> > >> > >> On Thu, Oct 30, 2008 at 3:43 AM, Rawshan Iajdani <iajdani@provati.com> > >> wrote: > >> > >> Well.. your attribute settings seems fine. For the 33 it will be 20. And > >> in place of 320, it will be 160.. hope this will give u a jump start.. > >> > >> > >> > >> > >> > >> > >> > >> Rawshan Iajdani > >> > >> > >> > >> > >> > >> From: pjsip-bounces@lists.pjsip.org [mailto: > pjsip-bounces@lists.pjsip.org] > >> On Behalf Of S. M. Nazmul Hasan (Opu) > >> Sent: Wednesday, October 29, 2008 6:12 PM > >> To: pjsip list > >> Subject: [pjsip] g729 codec > >> > >> > >> > >> Hi Benny, > >> > >> I am trying to add voice-age g729 codec with PJSIP for symbian. but i am > >> getting some difficulties while following the gsm.c file. > >> > >> 1. while creating g729 codec private data > >> what should i declare in replace of > >> // struct gsm_state *encoder; > >> // struct gsm_state *decoder; > >> > >> in "l16.c" its only use the frame size .. > >> > >> 2. in Generate default attribute. > >> > >> i have changed the value to this. is it ok? > >> attr->info.clock_rate = 8000; > >> attr->info.channel_cnt = 1; > >> attr->info.avg_bps = 8000; > >> attr->info.max_bps = 8000; > >> attr->info.pcm_bits_per_sample = 16; > >> attr->info.frm_ptime = 10; > >> > >> 3. In gsm.c there is some hard coding in gsm_codec_parse( > >> -- what will be the value in replace of "33" here > >> > >> 4. frame_size and sample_per_frame are also not clear to me.. > >> > >> Sorry for lot of questions and may be those are lot easier to ask.. but > >> really i am not good in codec.. > >> > >> waiting for reply. > >> > >> Thanks > >> > >> S. M. Nazmul Hasan Opu > >> Software Engineer > >> R & D Application > >> Dhaka, Bangladesh > >> Mob: +880 1712 901 764 > >> > >> _______________________________________________ > >> Visit our blog: http://blog.pjsip.org > >> > >> pjsip mailing list > >> pjsip@lists.pjsip.org > >> http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org > >> > >> > >> -- > >> S. M. Nazmul Hasan Opu > >> Software Engineer > >> R & D Application > >> Dhaka, Bangladesh > >> Mob: +880 1712 901 764 > >> > >> _______________________________________________ > >> Visit our blog: http://blog.pjsip.org > >> > >> pjsip mailing list > >> pjsip@lists.pjsip.org > >> http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org > >> > > > > > > > > -- > > S. M. Nazmul Hasan Opu > > Software Engineer > > R & D Application > > Dhaka, Bangladesh > > Mob: +880 1712 901 764 > > > > _______________________________________________ > > Visit our blog: http://blog.pjsip.org > > > > pjsip mailing list > > pjsip@lists.pjsip.org > > http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org > > > > > > _______________________________________________ > Visit our blog: http://blog.pjsip.org > > pjsip mailing list > pjsip@lists.pjsip.org > http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org > -- S. M. Nazmul Hasan Opu Software Engineer R & D Application Dhaka, Bangladesh Mob: +880 1712 901 764

         pj_str_t codec_id = pj_str("g729");
         pjmedia_codec_mgr_set_codec_priority(
             pjmedia_endpt_get_codec_mgr(pjsua_var.med_endpt),
             &codec_id, PJMEDIA_CODEC_PRIO_HIGHEST);

       Thanks

Hello Opu, I have a query to you... Is there available any VA G729 codec for symbian platform? regards, On Wed, Nov 5, 2008 at 7:39 AM, S. M. Nazmul Hasan (Opu) <apus29@gmail.com>wrote: > Nanag, > > I think so Nanang, may be its not registering because i am getting response > 488(Not acceptable here). But i did it register and unregister in > pjsua_media.c. > > Can you please see these files. > > Thanks > > Opu > > > On Tue, Nov 4, 2008 at 10:18 PM, Nanang Izzuddin <nanang@pjsip.org> wrote: > >> Hi, >> >> Just did a quick skimming, things I can found so far: >> 1. For 8000kbps and 10ms frame time, the encoded frame size should be >> 10 bytes, not 20 bytes. >> 2. this line: >> pcm_in += 160; >> should be: >> pcm_in += 80; >> >> Numbers in a codec wrapper are very important, so please check and >> recheck carefully. >> >> Regards, >> nanang >> >> >> On Tue, Nov 4, 2008 at 8:42 PM, S. M. Nazmul Hasan (Opu) >> <apus29@gmail.com> wrote: >> > Dear Rawshan, >> > >> > Can you please check the file if it is ok to add voiceage g729 in pjsip >> for >> > symbian. i am trying this for several days but i can get it working for >> me. >> > >> > Help from anyone would be great appretiable. >> > >> > Thanks >> > >> > On Mon, Nov 3, 2008 at 3:17 PM, Rawshan Iajdani <iajdani@provati.com> >> wrote: >> >> >> >> You should be able to do that in pjlib/include/pj/config_site.h and >> >> pjmedia/include/pjmedia-codec/config.h file. But always remember >> >> config_site.h settings will overwrite any settings done in config.h >> file. >> >> Good luck… >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> From: pjsip-bounces@lists.pjsip.org [mailto: >> pjsip-bounces@lists.pjsip.org] >> >> On Behalf Of S. M. Nazmul Hasan (Opu) >> >> Sent: Saturday, November 01, 2008 6:01 PM >> >> To: pjsip list >> >> Subject: Re: [pjsip] g729 codec >> >> >> >> >> >> >> >> Thanks Lajdani. My jumped start is running well. >> >> >> >> I am using Brekeke sip server to check the active session. and i am >> >> getting the using codec (payload) is always PCMU/8000. even after >> changing >> >> the g729 for highest priority. >> >> >> >> pj_str_t codec_id = pj_str("g729"); >> >> pjmedia_codec_mgr_set_codec_priority( >> >> pjmedia_endpt_get_codec_mgr(pjsua_var.med_endpt), >> >> &codec_id, PJMEDIA_CODEC_PRIO_HIGHEST); >> >> >> >> when i changed it to PJMEDIA_CODEC_PRIO_LOWEST the result was same. >> even >> >> after disabling all the codecs and after deleting all g711 files and >> related >> >> fields it showed the payload is PCMU/8000. >> >> >> >> How can i easily disable all the codecs except g729. >> >> >> >> Thanks >> >> >> >> >> >> >> >> On Thu, Oct 30, 2008 at 3:43 AM, Rawshan Iajdani <iajdani@provati.com> >> >> wrote: >> >> >> >> Well.. your attribute settings seems fine. For the 33 it will be 20. >> And >> >> in place of 320, it will be 160.. hope this will give u a jump start.. >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> Rawshan Iajdani >> >> >> >> >> >> >> >> >> >> >> >> From: pjsip-bounces@lists.pjsip.org [mailto: >> pjsip-bounces@lists.pjsip.org] >> >> On Behalf Of S. M. Nazmul Hasan (Opu) >> >> Sent: Wednesday, October 29, 2008 6:12 PM >> >> To: pjsip list >> >> Subject: [pjsip] g729 codec >> >> >> >> >> >> >> >> Hi Benny, >> >> >> >> I am trying to add voice-age g729 codec with PJSIP for symbian. but i >> am >> >> getting some difficulties while following the gsm.c file. >> >> >> >> 1. while creating g729 codec private data >> >> what should i declare in replace of >> >> // struct gsm_state *encoder; >> >> // struct gsm_state *decoder; >> >> >> >> in "l16.c" its only use the frame size .. >> >> >> >> 2. in Generate default attribute. >> >> >> >> i have changed the value to this. is it ok? >> >> attr->info.clock_rate = 8000; >> >> attr->info.channel_cnt = 1; >> >> attr->info.avg_bps = 8000; >> >> attr->info.max_bps = 8000; >> >> attr->info.pcm_bits_per_sample = 16; >> >> attr->info.frm_ptime = 10; >> >> >> >> 3. In gsm.c there is some hard coding in gsm_codec_parse( >> >> -- what will be the value in replace of "33" here >> >> >> >> 4. frame_size and sample_per_frame are also not clear to me.. >> >> >> >> Sorry for lot of questions and may be those are lot easier to ask.. but >> >> really i am not good in codec.. >> >> >> >> waiting for reply. >> >> >> >> Thanks >> >> >> >> S. M. Nazmul Hasan Opu >> >> Software Engineer >> >> R & D Application >> >> Dhaka, Bangladesh >> >> Mob: +880 1712 901 764 >> >> >> >> _______________________________________________ >> >> Visit our blog: http://blog.pjsip.org >> >> >> >> pjsip mailing list >> >> pjsip@lists.pjsip.org >> >> http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org >> >> >> >> >> >> -- >> >> S. M. Nazmul Hasan Opu >> >> Software Engineer >> >> R & D Application >> >> Dhaka, Bangladesh >> >> Mob: +880 1712 901 764 >> >> >> >> _______________________________________________ >> >> Visit our blog: http://blog.pjsip.org >> >> >> >> pjsip mailing list >> >> pjsip@lists.pjsip.org >> >> http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org >> >> >> > >> > >> > >> > -- >> > S. M. Nazmul Hasan Opu >> > Software Engineer >> > R & D Application >> > Dhaka, Bangladesh >> > Mob: +880 1712 901 764 >> > >> > _______________________________________________ >> > Visit our blog: http://blog.pjsip.org >> > >> > pjsip mailing list >> > pjsip@lists.pjsip.org >> > http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org >> > >> > >> >> _______________________________________________ >> Visit our blog: http://blog.pjsip.org >> >> pjsip mailing list >> pjsip@lists.pjsip.org >> http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org >> > > > > -- > S. M. Nazmul Hasan Opu > Software Engineer > R & D Application > Dhaka, Bangladesh > Mob: +880 1712 901 764 > > _______________________________________________ > Visit our blog: http://blog.pjsip.org > > pjsip mailing list > pjsip@lists.pjsip.org > http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org > > -- Salahuddin Ahmed

         pj_str_t codec_id = pj_str("g729");
         pjmedia_codec_mgr_set_codec_priority(
             pjmedia_endpt_get_codec_mgr(pjsua_var.med_endpt),
             &codec_id, PJMEDIA_CODEC_PRIO_HIGHEST);

       Thanks

you have to purchase it from them. On Wed, Nov 5, 2008 at 3:16 PM, Salahuddin Ahmed <bd.rubel@gmail.com> wrote: > Hello Opu, > I have a query to you... > Is there available any VA G729 codec for symbian platform? > > regards, > > > On Wed, Nov 5, 2008 at 7:39 AM, S. M. Nazmul Hasan (Opu) <apus29@gmail.com > > wrote: > >> Nanag, >> >> I think so Nanang, may be its not registering because i am getting >> response 488(Not acceptable here). But i did it register and unregister in >> pjsua_media.c. >> >> Can you please see these files. >> >> Thanks >> >> Opu >> >> >> On Tue, Nov 4, 2008 at 10:18 PM, Nanang Izzuddin <nanang@pjsip.org>wrote: >> >>> Hi, >>> >>> Just did a quick skimming, things I can found so far: >>> 1. For 8000kbps and 10ms frame time, the encoded frame size should be >>> 10 bytes, not 20 bytes. >>> 2. this line: >>> pcm_in += 160; >>> should be: >>> pcm_in += 80; >>> >>> Numbers in a codec wrapper are very important, so please check and >>> recheck carefully. >>> >>> Regards, >>> nanang >>> >>> >>> On Tue, Nov 4, 2008 at 8:42 PM, S. M. Nazmul Hasan (Opu) >>> <apus29@gmail.com> wrote: >>> > Dear Rawshan, >>> > >>> > Can you please check the file if it is ok to add voiceage g729 in pjsip >>> for >>> > symbian. i am trying this for several days but i can get it working for >>> me. >>> > >>> > Help from anyone would be great appretiable. >>> > >>> > Thanks >>> > >>> > On Mon, Nov 3, 2008 at 3:17 PM, Rawshan Iajdani <iajdani@provati.com> >>> wrote: >>> >> >>> >> You should be able to do that in pjlib/include/pj/config_site.h and >>> >> pjmedia/include/pjmedia-codec/config.h file. But always remember >>> >> config_site.h settings will overwrite any settings done in config.h >>> file. >>> >> Good luck… >>> >> >>> >> >>> >> >>> >> >>> >> >>> >> >>> >> >>> >> >>> >> >>> >> From: pjsip-bounces@lists.pjsip.org [mailto: >>> pjsip-bounces@lists.pjsip.org] >>> >> On Behalf Of S. M. Nazmul Hasan (Opu) >>> >> Sent: Saturday, November 01, 2008 6:01 PM >>> >> To: pjsip list >>> >> Subject: Re: [pjsip] g729 codec >>> >> >>> >> >>> >> >>> >> Thanks Lajdani. My jumped start is running well. >>> >> >>> >> I am using Brekeke sip server to check the active session. and i am >>> >> getting the using codec (payload) is always PCMU/8000. even after >>> changing >>> >> the g729 for highest priority. >>> >> >>> >> pj_str_t codec_id = pj_str("g729"); >>> >> pjmedia_codec_mgr_set_codec_priority( >>> >> pjmedia_endpt_get_codec_mgr(pjsua_var.med_endpt), >>> >> &codec_id, PJMEDIA_CODEC_PRIO_HIGHEST); >>> >> >>> >> when i changed it to PJMEDIA_CODEC_PRIO_LOWEST the result was same. >>> even >>> >> after disabling all the codecs and after deleting all g711 files and >>> related >>> >> fields it showed the payload is PCMU/8000. >>> >> >>> >> How can i easily disable all the codecs except g729. >>> >> >>> >> Thanks >>> >> >>> >> >>> >> >>> >> On Thu, Oct 30, 2008 at 3:43 AM, Rawshan Iajdani <iajdani@provati.com >>> > >>> >> wrote: >>> >> >>> >> Well.. your attribute settings seems fine. For the 33 it will be 20. >>> And >>> >> in place of 320, it will be 160.. hope this will give u a jump start.. >>> >> >>> >> >>> >> >>> >> >>> >> >>> >> >>> >> >>> >> Rawshan Iajdani >>> >> >>> >> >>> >> >>> >> >>> >> >>> >> From: pjsip-bounces@lists.pjsip.org [mailto: >>> pjsip-bounces@lists.pjsip.org] >>> >> On Behalf Of S. M. Nazmul Hasan (Opu) >>> >> Sent: Wednesday, October 29, 2008 6:12 PM >>> >> To: pjsip list >>> >> Subject: [pjsip] g729 codec >>> >> >>> >> >>> >> >>> >> Hi Benny, >>> >> >>> >> I am trying to add voice-age g729 codec with PJSIP for symbian. but i >>> am >>> >> getting some difficulties while following the gsm.c file. >>> >> >>> >> 1. while creating g729 codec private data >>> >> what should i declare in replace of >>> >> // struct gsm_state *encoder; >>> >> // struct gsm_state *decoder; >>> >> >>> >> in "l16.c" its only use the frame size .. >>> >> >>> >> 2. in Generate default attribute. >>> >> >>> >> i have changed the value to this. is it ok? >>> >> attr->info.clock_rate = 8000; >>> >> attr->info.channel_cnt = 1; >>> >> attr->info.avg_bps = 8000; >>> >> attr->info.max_bps = 8000; >>> >> attr->info.pcm_bits_per_sample = 16; >>> >> attr->info.frm_ptime = 10; >>> >> >>> >> 3. In gsm.c there is some hard coding in gsm_codec_parse( >>> >> -- what will be the value in replace of "33" here >>> >> >>> >> 4. frame_size and sample_per_frame are also not clear to me.. >>> >> >>> >> Sorry for lot of questions and may be those are lot easier to ask.. >>> but >>> >> really i am not good in codec.. >>> >> >>> >> waiting for reply. >>> >> >>> >> Thanks >>> >> >>> >> S. M. Nazmul Hasan Opu >>> >> Software Engineer >>> >> R & D Application >>> >> Dhaka, Bangladesh >>> >> Mob: +880 1712 901 764 >>> >> >>> >> _______________________________________________ >>> >> Visit our blog: http://blog.pjsip.org >>> >> >>> >> pjsip mailing list >>> >> pjsip@lists.pjsip.org >>> >> http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org >>> >> >>> >> >>> >> -- >>> >> S. M. Nazmul Hasan Opu >>> >> Software Engineer >>> >> R & D Application >>> >> Dhaka, Bangladesh >>> >> Mob: +880 1712 901 764 >>> >> >>> >> _______________________________________________ >>> >> Visit our blog: http://blog.pjsip.org >>> >> >>> >> pjsip mailing list >>> >> pjsip@lists.pjsip.org >>> >> http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org >>> >> >>> > >>> > >>> > >>> > -- >>> > S. M. Nazmul Hasan Opu >>> > Software Engineer >>> > R & D Application >>> > Dhaka, Bangladesh >>> > Mob: +880 1712 901 764 >>> > >>> > _______________________________________________ >>> > Visit our blog: http://blog.pjsip.org >>> > >>> > pjsip mailing list >>> > pjsip@lists.pjsip.org >>> > http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org >>> > >>> > >>> >>> _______________________________________________ >>> Visit our blog: http://blog.pjsip.org >>> >>> pjsip mailing list >>> pjsip@lists.pjsip.org >>> http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org >>> >> >> >> >> -- >> S. M. Nazmul Hasan Opu >> Software Engineer >> R & D Application >> Dhaka, Bangladesh >> Mob: +880 1712 901 764 >> >> _______________________________________________ >> Visit our blog: http://blog.pjsip.org >> >> pjsip mailing list >> pjsip@lists.pjsip.org >> http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org >> >> > > > -- > Salahuddin Ahmed > > _______________________________________________ > Visit our blog: http://blog.pjsip.org > > pjsip mailing list > pjsip@lists.pjsip.org > http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org > > -- S. M. Nazmul Hasan Opu Software Engineer R & D Application Dhaka, Bangladesh Mob: +880 1712 901 764

         pj_str_t codec_id = pj_str("g729");
         pjmedia_codec_mgr_set_codec_priority(
             pjmedia_endpt_get_codec_mgr(pjsua_var.med_endpt),
             &codec_id, PJMEDIA_CODEC_PRIO_HIGHEST);

       Thanks

Hi, Just in case, please make sure the other endpoint supports G729 too, e.g: make a call to PJSUA on desktop with IPP G729 enabled. And please check the pjsua log, see if G729 is in the INVITE SDP. If it is rejected, you may see the reason in the pjsua log. Regards, nanang On Wed, Nov 5, 2008 at 1:39 PM, S. M. Nazmul Hasan (Opu) <apus29@gmail.com> wrote: > Nanag, > > I think so Nanang, may be its not registering because i am getting response > 488(Not acceptable here). But i did it register and unregister in > pjsua_media.c. > > Can you please see these files. > > Thanks > > Opu > > On Tue, Nov 4, 2008 at 10:18 PM, Nanang Izzuddin <nanang@pjsip.org> wrote: >> >> Hi, >> >> Just did a quick skimming, things I can found so far: >> 1. For 8000kbps and 10ms frame time, the encoded frame size should be >> 10 bytes, not 20 bytes. >> 2. this line: >> pcm_in += 160; >> should be: >> pcm_in += 80; >> >> Numbers in a codec wrapper are very important, so please check and >> recheck carefully. >> >> Regards, >> nanang >> >> >> On Tue, Nov 4, 2008 at 8:42 PM, S. M. Nazmul Hasan (Opu) >> <apus29@gmail.com> wrote: >> > Dear Rawshan, >> > >> > Can you please check the file if it is ok to add voiceage g729 in pjsip >> > for >> > symbian. i am trying this for several days but i can get it working for >> > me. >> > >> > Help from anyone would be great appretiable. >> > >> > Thanks >> > >> > On Mon, Nov 3, 2008 at 3:17 PM, Rawshan Iajdani <iajdani@provati.com> >> > wrote: >> >> >> >> You should be able to do that in pjlib/include/pj/config_site.h and >> >> pjmedia/include/pjmedia-codec/config.h file. But always remember >> >> config_site.h settings will overwrite any settings done in config.h >> >> file. >> >> Good luck… >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> From: pjsip-bounces@lists.pjsip.org >> >> [mailto:pjsip-bounces@lists.pjsip.org] >> >> On Behalf Of S. M. Nazmul Hasan (Opu) >> >> Sent: Saturday, November 01, 2008 6:01 PM >> >> To: pjsip list >> >> Subject: Re: [pjsip] g729 codec >> >> >> >> >> >> >> >> Thanks Lajdani. My jumped start is running well. >> >> >> >> I am using Brekeke sip server to check the active session. and i am >> >> getting the using codec (payload) is always PCMU/8000. even after >> >> changing >> >> the g729 for highest priority. >> >> >> >> pj_str_t codec_id = pj_str("g729"); >> >> pjmedia_codec_mgr_set_codec_priority( >> >> pjmedia_endpt_get_codec_mgr(pjsua_var.med_endpt), >> >> &codec_id, PJMEDIA_CODEC_PRIO_HIGHEST); >> >> >> >> when i changed it to PJMEDIA_CODEC_PRIO_LOWEST the result was same. >> >> even >> >> after disabling all the codecs and after deleting all g711 files and >> >> related >> >> fields it showed the payload is PCMU/8000. >> >> >> >> How can i easily disable all the codecs except g729. >> >> >> >> Thanks >> >> >> >> >> >> >> >> On Thu, Oct 30, 2008 at 3:43 AM, Rawshan Iajdani <iajdani@provati.com> >> >> wrote: >> >> >> >> Well.. your attribute settings seems fine. For the 33 it will be 20. >> >> And >> >> in place of 320, it will be 160.. hope this will give u a jump start.. >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> Rawshan Iajdani >> >> >> >> >> >> >> >> >> >> >> >> From: pjsip-bounces@lists.pjsip.org >> >> [mailto:pjsip-bounces@lists.pjsip.org] >> >> On Behalf Of S. M. Nazmul Hasan (Opu) >> >> Sent: Wednesday, October 29, 2008 6:12 PM >> >> To: pjsip list >> >> Subject: [pjsip] g729 codec >> >> >> >> >> >> >> >> Hi Benny, >> >> >> >> I am trying to add voice-age g729 codec with PJSIP for symbian. but i >> >> am >> >> getting some difficulties while following the gsm.c file. >> >> >> >> 1. while creating g729 codec private data >> >> what should i declare in replace of >> >> // struct gsm_state *encoder; >> >> // struct gsm_state *decoder; >> >> >> >> in "l16.c" its only use the frame size .. >> >> >> >> 2. in Generate default attribute. >> >> >> >> i have changed the value to this. is it ok? >> >> attr->info.clock_rate = 8000; >> >> attr->info.channel_cnt = 1; >> >> attr->info.avg_bps = 8000; >> >> attr->info.max_bps = 8000; >> >> attr->info.pcm_bits_per_sample = 16; >> >> attr->info.frm_ptime = 10; >> >> >> >> 3. In gsm.c there is some hard coding in gsm_codec_parse( >> >> -- what will be the value in replace of "33" here >> >> >> >> 4. frame_size and sample_per_frame are also not clear to me.. >> >> >> >> Sorry for lot of questions and may be those are lot easier to ask.. but >> >> really i am not good in codec.. >> >> >> >> waiting for reply. >> >> >> >> Thanks >> >> >> >> S. M. Nazmul Hasan Opu >> >> Software Engineer >> >> R & D Application >> >> Dhaka, Bangladesh >> >> Mob: +880 1712 901 764 >> >> >> >> _______________________________________________ >> >> Visit our blog: http://blog.pjsip.org >> >> >> >> pjsip mailing list >> >> pjsip@lists.pjsip.org >> >> http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org >> >> >> >> >> >> -- >> >> S. M. Nazmul Hasan Opu >> >> Software Engineer >> >> R & D Application >> >> Dhaka, Bangladesh >> >> Mob: +880 1712 901 764 >> >> >> >> _______________________________________________ >> >> Visit our blog: http://blog.pjsip.org >> >> >> >> pjsip mailing list >> >> pjsip@lists.pjsip.org >> >> http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org >> >> >> > >> > >> > >> > -- >> > S. M. Nazmul Hasan Opu >> > Software Engineer >> > R & D Application >> > Dhaka, Bangladesh >> > Mob: +880 1712 901 764 >> > >> > _______________________________________________ >> > Visit our blog: http://blog.pjsip.org >> > >> > pjsip mailing list >> > pjsip@lists.pjsip.org >> > http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org >> > >> > >> >> _______________________________________________ >> Visit our blog: http://blog.pjsip.org >> >> pjsip mailing list >> pjsip@lists.pjsip.org >> http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org > > > > -- > S. M. Nazmul Hasan Opu > Software Engineer > R & D Application > Dhaka, Bangladesh > Mob: +880 1712 901 764 > > _______________________________________________ > Visit our blog: http://blog.pjsip.org > > pjsip mailing list > pjsip@lists.pjsip.org > http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org > >

         pj_str_t codec_id = pj_str("g729");
         pjmedia_codec_mgr_set_codec_priority(
             pjmedia_endpt_get_codec_mgr(pjsua_var.med_endpt),
             &codec_id, PJMEDIA_CODEC_PRIO_HIGHEST);

       Thanks

Hi Nanag, Yes the other endpoint supports g729. my tested environment was: server: AsteriskNow (which supports g729) client: pjsip (symbian_ua_gui) and Xlite Pro(which supports g729) 1. when i am using both endpoint as Xlite pro and fixing the codec as g729 only then both using g729 and call established fine. 2. But when i am trying with symbian_ua_gui with Xlite Pro fixing the codec to g729 then i am getting 488(Not acceptable here) response. Can you please tell me how can I get the log file for No. 2 test criteria. Thanks Opu On Wed, Nov 5, 2008 at 5:00 PM, Nanang Izzuddin <nanang@pjsip.org> wrote: > Hi, > > Just in case, please make sure the other endpoint supports G729 too, > e.g: make a call to PJSUA on desktop with IPP G729 enabled. And please > check the pjsua log, see if G729 is in the INVITE SDP. If it is > rejected, you may see the reason in the pjsua log. > > Regards, > nanang > > > On Wed, Nov 5, 2008 at 1:39 PM, S. M. Nazmul Hasan (Opu) > <apus29@gmail.com> wrote: > > Nanag, > > > > I think so Nanang, may be its not registering because i am getting > response > > 488(Not acceptable here). But i did it register and unregister in > > pjsua_media.c. > > > > Can you please see these files. > > > > Thanks > > > > Opu > > > > On Tue, Nov 4, 2008 at 10:18 PM, Nanang Izzuddin <nanang@pjsip.org> > wrote: > >> > >> Hi, > >> > >> Just did a quick skimming, things I can found so far: > >> 1. For 8000kbps and 10ms frame time, the encoded frame size should be > >> 10 bytes, not 20 bytes. > >> 2. this line: > >> pcm_in += 160; > >> should be: > >> pcm_in += 80; > >> > >> Numbers in a codec wrapper are very important, so please check and > >> recheck carefully. > >> > >> Regards, > >> nanang > >> > >> > >> On Tue, Nov 4, 2008 at 8:42 PM, S. M. Nazmul Hasan (Opu) > >> <apus29@gmail.com> wrote: > >> > Dear Rawshan, > >> > > >> > Can you please check the file if it is ok to add voiceage g729 in > pjsip > >> > for > >> > symbian. i am trying this for several days but i can get it working > for > >> > me. > >> > > >> > Help from anyone would be great appretiable. > >> > > >> > Thanks > >> > > >> > On Mon, Nov 3, 2008 at 3:17 PM, Rawshan Iajdani <iajdani@provati.com> > >> > wrote: > >> >> > >> >> You should be able to do that in pjlib/include/pj/config_site.h and > >> >> pjmedia/include/pjmedia-codec/config.h file. But always remember > >> >> config_site.h settings will overwrite any settings done in config.h > >> >> file. > >> >> Good luck… > >> >> > >> >> > >> >> > >> >> > >> >> > >> >> > >> >> > >> >> > >> >> > >> >> From: pjsip-bounces@lists.pjsip.org > >> >> [mailto:pjsip-bounces@lists.pjsip.org] > >> >> On Behalf Of S. M. Nazmul Hasan (Opu) > >> >> Sent: Saturday, November 01, 2008 6:01 PM > >> >> To: pjsip list > >> >> Subject: Re: [pjsip] g729 codec > >> >> > >> >> > >> >> > >> >> Thanks Lajdani. My jumped start is running well. > >> >> > >> >> I am using Brekeke sip server to check the active session. and i am > >> >> getting the using codec (payload) is always PCMU/8000. even after > >> >> changing > >> >> the g729 for highest priority. > >> >> > >> >> pj_str_t codec_id = pj_str("g729"); > >> >> pjmedia_codec_mgr_set_codec_priority( > >> >> pjmedia_endpt_get_codec_mgr(pjsua_var.med_endpt), > >> >> &codec_id, PJMEDIA_CODEC_PRIO_HIGHEST); > >> >> > >> >> when i changed it to PJMEDIA_CODEC_PRIO_LOWEST the result was same. > >> >> even > >> >> after disabling all the codecs and after deleting all g711 files and > >> >> related > >> >> fields it showed the payload is PCMU/8000. > >> >> > >> >> How can i easily disable all the codecs except g729. > >> >> > >> >> Thanks > >> >> > >> >> > >> >> > >> >> On Thu, Oct 30, 2008 at 3:43 AM, Rawshan Iajdani < > iajdani@provati.com> > >> >> wrote: > >> >> > >> >> Well.. your attribute settings seems fine. For the 33 it will be 20. > >> >> And > >> >> in place of 320, it will be 160.. hope this will give u a jump > start.. > >> >> > >> >> > >> >> > >> >> > >> >> > >> >> > >> >> > >> >> Rawshan Iajdani > >> >> > >> >> > >> >> > >> >> > >> >> > >> >> From: pjsip-bounces@lists.pjsip.org > >> >> [mailto:pjsip-bounces@lists.pjsip.org] > >> >> On Behalf Of S. M. Nazmul Hasan (Opu) > >> >> Sent: Wednesday, October 29, 2008 6:12 PM > >> >> To: pjsip list > >> >> Subject: [pjsip] g729 codec > >> >> > >> >> > >> >> > >> >> Hi Benny, > >> >> > >> >> I am trying to add voice-age g729 codec with PJSIP for symbian. but i > >> >> am > >> >> getting some difficulties while following the gsm.c file. > >> >> > >> >> 1. while creating g729 codec private data > >> >> what should i declare in replace of > >> >> // struct gsm_state *encoder; > >> >> // struct gsm_state *decoder; > >> >> > >> >> in "l16.c" its only use the frame size .. > >> >> > >> >> 2. in Generate default attribute. > >> >> > >> >> i have changed the value to this. is it ok? > >> >> attr->info.clock_rate = 8000; > >> >> attr->info.channel_cnt = 1; > >> >> attr->info.avg_bps = 8000; > >> >> attr->info.max_bps = 8000; > >> >> attr->info.pcm_bits_per_sample = 16; > >> >> attr->info.frm_ptime = 10; > >> >> > >> >> 3. In gsm.c there is some hard coding in gsm_codec_parse( > >> >> -- what will be the value in replace of "33" here > >> >> > >> >> 4. frame_size and sample_per_frame are also not clear to me.. > >> >> > >> >> Sorry for lot of questions and may be those are lot easier to ask.. > but > >> >> really i am not good in codec.. > >> >> > >> >> waiting for reply. > >> >> > >> >> Thanks > >> >> > >> >> S. M. Nazmul Hasan Opu > >> >> Software Engineer > >> >> R & D Application > >> >> Dhaka, Bangladesh > >> >> Mob: +880 1712 901 764 > >> >> > >> >> _______________________________________________ > >> >> Visit our blog: http://blog.pjsip.org > >> >> > >> >> pjsip mailing list > >> >> pjsip@lists.pjsip.org > >> >> http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org > >> >> > >> >> > >> >> -- > >> >> S. M. Nazmul Hasan Opu > >> >> Software Engineer > >> >> R & D Application > >> >> Dhaka, Bangladesh > >> >> Mob: +880 1712 901 764 > >> >> > >> >> _______________________________________________ > >> >> Visit our blog: http://blog.pjsip.org > >> >> > >> >> pjsip mailing list > >> >> pjsip@lists.pjsip.org > >> >> http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org > >> >> > >> > > >> > > >> > > >> > -- > >> > S. M. Nazmul Hasan Opu > >> > Software Engineer > >> > R & D Application > >> > Dhaka, Bangladesh > >> > Mob: +880 1712 901 764 > >> > > >> > _______________________________________________ > >> > Visit our blog: http://blog.pjsip.org > >> > > >> > pjsip mailing list > >> > pjsip@lists.pjsip.org > >> > http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org > >> > > >> > > >> > >> _______________________________________________ > >> Visit our blog: http://blog.pjsip.org > >> > >> pjsip mailing list > >> pjsip@lists.pjsip.org > >> http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org > > > > > > > > -- > > S. M. Nazmul Hasan Opu > > Software Engineer > > R & D Application > > Dhaka, Bangladesh > > Mob: +880 1712 901 764 > > > > _______________________________________________ > > Visit our blog: http://blog.pjsip.org > > > > pjsip mailing list > > pjsip@lists.pjsip.org > > http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org > > > > > > _______________________________________________ > Visit our blog: http://blog.pjsip.org > > pjsip mailing list > pjsip@lists.pjsip.org > http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org > -- S. M. Nazmul Hasan Opu Software Engineer R & D Application Dhaka, Bangladesh Mob: +880 1712 901 764

         pj_str_t codec_id = pj_str("g729");
         pjmedia_codec_mgr_set_codec_priority(
             pjmedia_endpt_get_codec_mgr(pjsua_var.med_endpt),
             &codec_id, PJMEDIA_CODEC_PRIO_HIGHEST);

       Thanks

Hi, I think it would be better if you see the log file in the rejecting side (it can be the proxy/asterisk or x-lite) to see why it rejected the call. If you have pjsua with IPP enabled, you can just make a call to the pjsua and send along the pjsua log file. The log file in the Symbian side may help, but I am not sure it will help a lot, however to generate the log file on Symbian, just specify log filename in logging config for the pjsua_init(), e.g: log_cfg.log_filename = pj_str("c:\\data\\pjsua.log"); Regards, nanang On Thu, Nov 6, 2008 at 3:02 PM, S. M. Nazmul Hasan (Opu) <apus29@gmail.com> wrote: > Hi Nanag, > > Yes the other endpoint supports g729. my tested environment was: > > server: AsteriskNow (which supports g729) > client: pjsip (symbian_ua_gui) and Xlite Pro(which supports g729) > > 1. when i am using both endpoint as Xlite pro and fixing the codec as g729 > only then both using g729 and call established fine. > > 2. But when i am trying with symbian_ua_gui with Xlite Pro fixing the codec > to g729 then i am getting 488(Not acceptable here) response. > > Can you please tell me how can I get the log file for No. 2 test criteria. > > Thanks > > Opu > > On Wed, Nov 5, 2008 at 5:00 PM, Nanang Izzuddin <nanang@pjsip.org> wrote: >> >> Hi, >> >> Just in case, please make sure the other endpoint supports G729 too, >> e.g: make a call to PJSUA on desktop with IPP G729 enabled. And please >> check the pjsua log, see if G729 is in the INVITE SDP. If it is >> rejected, you may see the reason in the pjsua log. >> >> Regards, >> nanang >> >> >> On Wed, Nov 5, 2008 at 1:39 PM, S. M. Nazmul Hasan (Opu) >> <apus29@gmail.com> wrote: >> > Nanag, >> > >> > I think so Nanang, may be its not registering because i am getting >> > response >> > 488(Not acceptable here). But i did it register and unregister in >> > pjsua_media.c. >> > >> > Can you please see these files. >> > >> > Thanks >> > >> > Opu >> > >> > On Tue, Nov 4, 2008 at 10:18 PM, Nanang Izzuddin <nanang@pjsip.org> >> > wrote: >> >> >> >> Hi, >> >> >> >> Just did a quick skimming, things I can found so far: >> >> 1. For 8000kbps and 10ms frame time, the encoded frame size should be >> >> 10 bytes, not 20 bytes. >> >> 2. this line: >> >> pcm_in += 160; >> >> should be: >> >> pcm_in += 80; >> >> >> >> Numbers in a codec wrapper are very important, so please check and >> >> recheck carefully. >> >> >> >> Regards, >> >> nanang >> >> >> >> >> >> On Tue, Nov 4, 2008 at 8:42 PM, S. M. Nazmul Hasan (Opu) >> >> <apus29@gmail.com> wrote: >> >> > Dear Rawshan, >> >> > >> >> > Can you please check the file if it is ok to add voiceage g729 in >> >> > pjsip >> >> > for >> >> > symbian. i am trying this for several days but i can get it working >> >> > for >> >> > me. >> >> > >> >> > Help from anyone would be great appretiable. >> >> > >> >> > Thanks >> >> > >> >> > On Mon, Nov 3, 2008 at 3:17 PM, Rawshan Iajdani <iajdani@provati.com> >> >> > wrote: >> >> >> >> >> >> You should be able to do that in pjlib/include/pj/config_site.h and >> >> >> pjmedia/include/pjmedia-codec/config.h file. But always remember >> >> >> config_site.h settings will overwrite any settings done in config.h >> >> >> file. >> >> >> Good luck… >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> From: pjsip-bounces@lists.pjsip.org >> >> >> [mailto:pjsip-bounces@lists.pjsip.org] >> >> >> On Behalf Of S. M. Nazmul Hasan (Opu) >> >> >> Sent: Saturday, November 01, 2008 6:01 PM >> >> >> To: pjsip list >> >> >> Subject: Re: [pjsip] g729 codec >> >> >> >> >> >> >> >> >> >> >> >> Thanks Lajdani. My jumped start is running well. >> >> >> >> >> >> I am using Brekeke sip server to check the active session. and i am >> >> >> getting the using codec (payload) is always PCMU/8000. even after >> >> >> changing >> >> >> the g729 for highest priority. >> >> >> >> >> >> pj_str_t codec_id = pj_str("g729"); >> >> >> pjmedia_codec_mgr_set_codec_priority( >> >> >> pjmedia_endpt_get_codec_mgr(pjsua_var.med_endpt), >> >> >> &codec_id, PJMEDIA_CODEC_PRIO_HIGHEST); >> >> >> >> >> >> when i changed it to PJMEDIA_CODEC_PRIO_LOWEST the result was same. >> >> >> even >> >> >> after disabling all the codecs and after deleting all g711 files and >> >> >> related >> >> >> fields it showed the payload is PCMU/8000. >> >> >> >> >> >> How can i easily disable all the codecs except g729. >> >> >> >> >> >> Thanks >> >> >> >> >> >> >> >> >> >> >> >> On Thu, Oct 30, 2008 at 3:43 AM, Rawshan Iajdani >> >> >> <iajdani@provati.com> >> >> >> wrote: >> >> >> >> >> >> Well.. your attribute settings seems fine. For the 33 it will be 20. >> >> >> And >> >> >> in place of 320, it will be 160.. hope this will give u a jump >> >> >> start.. >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> Rawshan Iajdani >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> From: pjsip-bounces@lists.pjsip.org >> >> >> [mailto:pjsip-bounces@lists.pjsip.org] >> >> >> On Behalf Of S. M. Nazmul Hasan (Opu) >> >> >> Sent: Wednesday, October 29, 2008 6:12 PM >> >> >> To: pjsip list >> >> >> Subject: [pjsip] g729 codec >> >> >> >> >> >> >> >> >> >> >> >> Hi Benny, >> >> >> >> >> >> I am trying to add voice-age g729 codec with PJSIP for symbian. but >> >> >> i >> >> >> am >> >> >> getting some difficulties while following the gsm.c file. >> >> >> >> >> >> 1. while creating g729 codec private data >> >> >> what should i declare in replace of >> >> >> // struct gsm_state *encoder; >> >> >> // struct gsm_state *decoder; >> >> >> >> >> >> in "l16.c" its only use the frame size .. >> >> >> >> >> >> 2. in Generate default attribute. >> >> >> >> >> >> i have changed the value to this. is it ok? >> >> >> attr->info.clock_rate = 8000; >> >> >> attr->info.channel_cnt = 1; >> >> >> attr->info.avg_bps = 8000; >> >> >> attr->info.max_bps = 8000; >> >> >> attr->info.pcm_bits_per_sample = 16; >> >> >> attr->info.frm_ptime = 10; >> >> >> >> >> >> 3. In gsm.c there is some hard coding in gsm_codec_parse( >> >> >> -- what will be the value in replace of "33" here >> >> >> >> >> >> 4. frame_size and sample_per_frame are also not clear to me.. >> >> >> >> >> >> Sorry for lot of questions and may be those are lot easier to ask.. >> >> >> but >> >> >> really i am not good in codec.. >> >> >> >> >> >> waiting for reply. >> >> >> >> >> >> Thanks >> >> >> >> >> >> S. M. Nazmul Hasan Opu >> >> >> Software Engineer >> >> >> R & D Application >> >> >> Dhaka, Bangladesh >> >> >> Mob: +880 1712 901 764 >> >> >> >> >> >> _______________________________________________ >> >> >> Visit our blog: http://blog.pjsip.org >> >> >> >> >> >> pjsip mailing list >> >> >> pjsip@lists.pjsip.org >> >> >> http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org >> >> >> >> >> >> >> >> >> -- >> >> >> S. M. Nazmul Hasan Opu >> >> >> Software Engineer >> >> >> R & D Application >> >> >> Dhaka, Bangladesh >> >> >> Mob: +880 1712 901 764 >> >> >> >> >> >> _______________________________________________ >> >> >> Visit our blog: http://blog.pjsip.org >> >> >> >> >> >> pjsip mailing list >> >> >> pjsip@lists.pjsip.org >> >> >> http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org >> >> >> >> >> > >> >> > >> >> > >> >> > -- >> >> > S. M. Nazmul Hasan Opu >> >> > Software Engineer >> >> > R & D Application >> >> > Dhaka, Bangladesh >> >> > Mob: +880 1712 901 764 >> >> > >> >> > _______________________________________________ >> >> > Visit our blog: http://blog.pjsip.org >> >> > >> >> > pjsip mailing list >> >> > pjsip@lists.pjsip.org >> >> > http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org >> >> > >> >> > >> >> >> >> _______________________________________________ >> >> Visit our blog: http://blog.pjsip.org >> >> >> >> pjsip mailing list >> >> pjsip@lists.pjsip.org >> >> http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org >> > >> > >> > >> > -- >> > S. M. Nazmul Hasan Opu >> > Software Engineer >> > R & D Application >> > Dhaka, Bangladesh >> > Mob: +880 1712 901 764 >> > >> > _______________________________________________ >> > Visit our blog: http://blog.pjsip.org >> > >> > pjsip mailing list >> > pjsip@lists.pjsip.org >> > http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org >> > >> > >> >> _______________________________________________ >> Visit our blog: http://blog.pjsip.org >> >> pjsip mailing list >> pjsip@lists.pjsip.org >> http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org > > > > -- > S. M. Nazmul Hasan Opu > Software Engineer > R & D Application > Dhaka, Bangladesh > Mob: +880 1712 901 764 > > _______________________________________________ > Visit our blog: http://blog.pjsip.org > > pjsip mailing list > pjsip@lists.pjsip.org > http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org > >

We have different state of call, and one of them is confirmed. I get the confirmed state when the phone is ringing. But I don't find any state when the call is answered or connected. Ideas?????

I had the same problem. I solved it by using on_call_tsx_state callback to watch the replies come back. ----- Original Message ----- From: "Rawshan Iajdani" <iajdani@provati.com> To: "'pjsip list'" <pjsip@lists.pjsip.org> Sent: Thursday, November 06, 2008 8:49 AM Subject: [pjsip] Call State > We have different state of call, and one of them is confirmed. I get the > confirmed state when the phone is ringing. But I don't find any state when > the call is answered or connected. Ideas????? > > > _______________________________________________ > Visit our blog: http://blog.pjsip.org > > pjsip mailing list > pjsip@lists.pjsip.org > http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org >

         pj_str_t codec_id = pj_str("g729");
         pjmedia_codec_mgr_set_codec_priority(
             pjmedia_endpt_get_codec_mgr(pjsua_var.med_endpt),
             &codec_id, PJMEDIA_CODEC_PRIO_HIGHEST);

       Thanks

Hi Nanang, I have tested it to make a self call and getting 488(Not acceptable here) response. i saw the log file and the sdp , there speex, gsm , pcma and pcmu only presents . and no g729. so it looks like that g729 is not registered. and server rejected the call. I have checked the log file of server , there was a message: *No compatible codecs, not accepting this offer!* please give me a suggestion. Thanks opu On Thu, Nov 6, 2008 at 4:26 PM, Nanang Izzuddin <nanang@pjsip.org> wrote: > Hi, > > I think it would be better if you see the log file in the rejecting > side (it can be the proxy/asterisk or x-lite) to see why it rejected > the call. If you have pjsua with IPP enabled, you can just make a call > to the pjsua and send along the pjsua log file. > > The log file in the Symbian side may help, but I am not sure it will > help a lot, however to generate the log file on Symbian, just specify > log filename in logging config for the pjsua_init(), e.g: > log_cfg.log_filename = pj_str("c:\\data\\pjsua.log"); > > Regards, > nanang > > > On Thu, Nov 6, 2008 at 3:02 PM, S. M. Nazmul Hasan (Opu) > <apus29@gmail.com> wrote: > > Hi Nanag, > > > > Yes the other endpoint supports g729. my tested environment was: > > > > server: AsteriskNow (which supports g729) > > client: pjsip (symbian_ua_gui) and Xlite Pro(which supports g729) > > > > 1. when i am using both endpoint as Xlite pro and fixing the codec as > g729 > > only then both using g729 and call established fine. > > > > 2. But when i am trying with symbian_ua_gui with Xlite Pro fixing the > codec > > to g729 then i am getting 488(Not acceptable here) response. > > > > Can you please tell me how can I get the log file for No. 2 test > criteria. > > > > Thanks > > > > Opu > > > > On Wed, Nov 5, 2008 at 5:00 PM, Nanang Izzuddin <nanang@pjsip.org> > wrote: > >> > >> Hi, > >> > >> Just in case, please make sure the other endpoint supports G729 too, > >> e.g: make a call to PJSUA on desktop with IPP G729 enabled. And please > >> check the pjsua log, see if G729 is in the INVITE SDP. If it is > >> rejected, you may see the reason in the pjsua log. > >> > >> Regards, > >> nanang > >> > >> > >> On Wed, Nov 5, 2008 at 1:39 PM, S. M. Nazmul Hasan (Opu) > >> <apus29@gmail.com> wrote: > >> > Nanag, > >> > > >> > I think so Nanang, may be its not registering because i am getting > >> > response > >> > 488(Not acceptable here). But i did it register and unregister in > >> > pjsua_media.c. > >> > > >> > Can you please see these files. > >> > > >> > Thanks > >> > > >> > Opu > >> > > >> > On Tue, Nov 4, 2008 at 10:18 PM, Nanang Izzuddin <nanang@pjsip.org> > >> > wrote: > >> >> > >> >> Hi, > >> >> > >> >> Just did a quick skimming, things I can found so far: > >> >> 1. For 8000kbps and 10ms frame time, the encoded frame size should be > >> >> 10 bytes, not 20 bytes. > >> >> 2. this line: > >> >> pcm_in += 160; > >> >> should be: > >> >> pcm_in += 80; > >> >> > >> >> Numbers in a codec wrapper are very important, so please check and > >> >> recheck carefully. > >> >> > >> >> Regards, > >> >> nanang > >> >> > >> >> > >> >> On Tue, Nov 4, 2008 at 8:42 PM, S. M. Nazmul Hasan (Opu) > >> >> <apus29@gmail.com> wrote: > >> >> > Dear Rawshan, > >> >> > > >> >> > Can you please check the file if it is ok to add voiceage g729 in > >> >> > pjsip > >> >> > for > >> >> > symbian. i am trying this for several days but i can get it working > >> >> > for > >> >> > me. > >> >> > > >> >> > Help from anyone would be great appretiable. > >> >> > > >> >> > Thanks > >> >> > > >> >> > On Mon, Nov 3, 2008 at 3:17 PM, Rawshan Iajdani < > iajdani@provati.com> > >> >> > wrote: > >> >> >> > >> >> >> You should be able to do that in pjlib/include/pj/config_site.h > and > >> >> >> pjmedia/include/pjmedia-codec/config.h file. But always remember > >> >> >> config_site.h settings will overwrite any settings done in > config.h > >> >> >> file. > >> >> >> Good luck… > >> >> >> > >> >> >> > >> >> >> > >> >> >> > >> >> >> > >> >> >> > >> >> >> > >> >> >> > >> >> >> > >> >> >> From: pjsip-bounces@lists.pjsip.org > >> >> >> [mailto:pjsip-bounces@lists.pjsip.org] > >> >> >> On Behalf Of S. M. Nazmul Hasan (Opu) > >> >> >> Sent: Saturday, November 01, 2008 6:01 PM > >> >> >> To: pjsip list > >> >> >> Subject: Re: [pjsip] g729 codec > >> >> >> > >> >> >> > >> >> >> > >> >> >> Thanks Lajdani. My jumped start is running well. > >> >> >> > >> >> >> I am using Brekeke sip server to check the active session. and i > am > >> >> >> getting the using codec (payload) is always PCMU/8000. even after > >> >> >> changing > >> >> >> the g729 for highest priority. > >> >> >> > >> >> >> pj_str_t codec_id = pj_str("g729"); > >> >> >> pjmedia_codec_mgr_set_codec_priority( > >> >> >> pjmedia_endpt_get_codec_mgr(pjsua_var.med_endpt), > >> >> >> &codec_id, PJMEDIA_CODEC_PRIO_HIGHEST); > >> >> >> > >> >> >> when i changed it to PJMEDIA_CODEC_PRIO_LOWEST the result was > same. > >> >> >> even > >> >> >> after disabling all the codecs and after deleting all g711 files > and > >> >> >> related > >> >> >> fields it showed the payload is PCMU/8000. > >> >> >> > >> >> >> How can i easily disable all the codecs except g729. > >> >> >> > >> >> >> Thanks > >> >> >> > >> >> >> > >> >> >> > >> >> >> On Thu, Oct 30, 2008 at 3:43 AM, Rawshan Iajdani > >> >> >> <iajdani@provati.com> > >> >> >> wrote: > >> >> >> > >> >> >> Well.. your attribute settings seems fine. For the 33 it will be > 20. > >> >> >> And > >> >> >> in place of 320, it will be 160.. hope this will give u a jump > >> >> >> start.. > >> >> >> > >> >> >> > >> >> >> > >> >> >> > >> >> >> > >> >> >> > >> >> >> > >> >> >> Rawshan Iajdani > >> >> >> > >> >> >> > >> >> >> > >> >> >> > >> >> >> > >> >> >> From: pjsip-bounces@lists.pjsip.org > >> >> >> [mailto:pjsip-bounces@lists.pjsip.org] > >> >> >> On Behalf Of S. M. Nazmul Hasan (Opu) > >> >> >> Sent: Wednesday, October 29, 2008 6:12 PM > >> >> >> To: pjsip list > >> >> >> Subject: [pjsip] g729 codec > >> >> >> > >> >> >> > >> >> >> > >> >> >> Hi Benny, > >> >> >> > >> >> >> I am trying to add voice-age g729 codec with PJSIP for symbian. > but > >> >> >> i > >> >> >> am > >> >> >> getting some difficulties while following the gsm.c file. > >> >> >> > >> >> >> 1. while creating g729 codec private data > >> >> >> what should i declare in replace of > >> >> >> // struct gsm_state *encoder; > >> >> >> // struct gsm_state *decoder; > >> >> >> > >> >> >> in "l16.c" its only use the frame size .. > >> >> >> > >> >> >> 2. in Generate default attribute. > >> >> >> > >> >> >> i have changed the value to this. is it ok? > >> >> >> attr->info.clock_rate = 8000; > >> >> >> attr->info.channel_cnt = 1; > >> >> >> attr->info.avg_bps = 8000; > >> >> >> attr->info.max_bps = 8000; > >> >> >> attr->info.pcm_bits_per_sample = 16; > >> >> >> attr->info.frm_ptime = 10; > >> >> >> > >> >> >> 3. In gsm.c there is some hard coding in gsm_codec_parse( > >> >> >> -- what will be the value in replace of "33" here > >> >> >> > >> >> >> 4. frame_size and sample_per_frame are also not clear to me.. > >> >> >> > >> >> >> Sorry for lot of questions and may be those are lot easier to > ask.. > >> >> >> but > >> >> >> really i am not good in codec.. > >> >> >> > >> >> >> waiting for reply. > >> >> >> > >> >> >> Thanks > >> >> >> > >> >> >> S. M. Nazmul Hasan Opu > >> >> >> Software Engineer > >> >> >> R & D Application > >> >> >> Dhaka, Bangladesh > >> >> >> Mob: +880 1712 901 764 > >> >> >> > >> >> >> _______________________________________________ > >> >> >> Visit our blog: http://blog.pjsip.org > >> >> >> > >> >> >> pjsip mailing list > >> >> >> pjsip@lists.pjsip.org > >> >> >> http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org > >> >> >> > >> >> >> > >> >> >> -- > >> >> >> S. M. Nazmul Hasan Opu > >> >> >> Software Engineer > >> >> >> R & D Application > >> >> >> Dhaka, Bangladesh > >> >> >> Mob: +880 1712 901 764 > >> >> >> > >> >> >> _______________________________________________ > >> >> >> Visit our blog: http://blog.pjsip.org > >> >> >> > >> >> >> pjsip mailing list > >> >> >> pjsip@lists.pjsip.org > >> >> >> http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org > >> >> >> > >> >> > > >> >> > > >> >> > > >> >> > -- > >> >> > S. M. Nazmul Hasan Opu > >> >> > Software Engineer > >> >> > R & D Application > >> >> > Dhaka, Bangladesh > >> >> > Mob: +880 1712 901 764 > >> >> > > >> >> > _______________________________________________ > >> >> > Visit our blog: http://blog.pjsip.org > >> >> > > >> >> > pjsip mailing list > >> >> > pjsip@lists.pjsip.org > >> >> > http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org > >> >> > > >> >> > > >> >> > >> >> _______________________________________________ > >> >> Visit our blog: http://blog.pjsip.org > >> >> > >> >> pjsip mailing list > >> >> pjsip@lists.pjsip.org > >> >> http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org > >> > > >> > > >> > > >> > -- > >> > S. M. Nazmul Hasan Opu > >> > Software Engineer > >> > R & D Application > >> > Dhaka, Bangladesh > >> > Mob: +880 1712 901 764 > >> > > >> > _______________________________________________ > >> > Visit our blog: http://blog.pjsip.org > >> > > >> > pjsip mailing list > >> > pjsip@lists.pjsip.org > >> > http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org > >> > > >> > > >> > >> _______________________________________________ > >> Visit our blog: http://blog.pjsip.org > >> > >> pjsip mailing list > >> pjsip@lists.pjsip.org > >> http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org > > > > > > > > -- > > S. M. Nazmul Hasan Opu > > Software Engineer > > R & D Application > > Dhaka, Bangladesh > > Mob: +880 1712 901 764 > > > > _______________________________________________ > > Visit our blog: http://blog.pjsip.org > > > > pjsip mailing list > > pjsip@lists.pjsip.org > > http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org > > > > > > _______________________________________________ > Visit our blog: http://blog.pjsip.org > > pjsip mailing list > pjsip@lists.pjsip.org > http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org > -- S. M. Nazmul Hasan Opu Software Engineer R & D Application Dhaka, Bangladesh Mob: +880 1712 901 764

 {

    printf("Usage :%s bitstream_file  outputspeech_file\n",argv[0]);

    printf("\n");

    printf("Format for bitstream_file:\n");

    printf("  One (2-byte) synchronization word \n");

    printf("  One (2-byte) size word,\n");

    printf("  80 words (2-byte) containing 80 bits.\n");

    printf("\n");

    printf("Format for outputspeech_file:\n");

    printf("  Synthesis is written to a binary file of 16 bits

    exit( 1 );

 }

 {

    printf("%s - Error opening file  %s !!\n", argv[0], argv[1]);

    exit(0);

 }

 {

    printf("%s - Error opening file  %s !!\n", argv[0], argv[2]);

    exit(0);

 }

covertBit2Serial(iBuffer,serial);



printf("Frame =%d\r", frame++);



bits2prm_ld8k( &serial[2], &parm[1]);



/* the hardware detects frame erasures by checking if all bits

   are set to zero

 */

parm[0] = 0;           /* No frame erasure */

for (i=2; i < SERIAL_SIZE; i++)

  if (serial[i] == 0 ) parm[0] = 1; /* frame erased     */



/* check pitch parity and put 1 in parm[4] if parity error */



parm[4] = Check_Parity_Pitch(parm[3], parm[4]);



Decod_ld8a(parm, synth, Az_dec, T2);



Post_Filter(synth, Az_dec, T2);        /* Post-filter */



Post_Process(synth, L_FRAME);



fwrite(synth, sizeof(short), L_FRAME, f_syn);

  for(j=0;j<16;j++){

  Mask = 0x0001 << (15-j);

  

  if ((Mask | input[i]) == input[i])

        output[outIndex++] = BIT_1;

  

  else

        output[outIndex++] = BIT_0;

        

  }

  printf("%c",value & MASK ? '1' : '0' );



  value <<= 1;



 // if ( k == 8)

  //   printf( ' ');

I know you implemented ITU version of codecs. I included my test file where I tried to keep everything same as ITU, except that I convert the 10 bit frames to 80 16bit samples. After that I decode that to PCM. But the sound I get after decoding is just some metallic scrambled. Am I missing anything here? Look at the routine covertBit2Serial(); That was what written by me. void covertBit2Serial(Word16 *input, Word16 *output); void printbits(Word16 value); int main(int argc, char *argv[] ) { Word16 synth_buf[L_FRAME+M], *synth; /* Synthesis */ Word16 parm[PRM_SIZE+1]; /* Synthesis parameters */ Word16 serial[SERIAL_SIZE]; /* Serial stream */ Word16 Az_dec[MP1*2]; /* Decoded Az for post-filter */ Word16 T2[2]; /* Pitch lag for 2 subframes */ Word16 i, frame; FILE *f_syn, *f_serial; printf("\n"); printf("************ G.729a 8.0 KBIT/S SPEECH DECODER ************\n"); printf("\n"); printf("------------------- Fixed point C simulation ----------------\n"); printf("\n"); printf("------------ Version 1.1 (Release 2, November 2006) --------\n"); printf("\n"); /* Passed arguments */ if ( argc != 3) { printf("Usage :%s bitstream_file outputspeech_file\n",argv[0]); printf("\n"); printf("Format for bitstream_file:\n"); printf(" One (2-byte) synchronization word \n"); printf(" One (2-byte) size word,\n"); printf(" 80 words (2-byte) containing 80 bits.\n"); printf("\n"); printf("Format for outputspeech_file:\n"); printf(" Synthesis is written to a binary file of 16 bits data.\n"); exit( 1 ); } /* Open file for synthesis and packed serial stream */ if( (f_serial = fopen(argv[1],"rb") ) == NULL ) { printf("%s - Error opening file %s !!\n", argv[0], argv[1]); exit(0); } if( (f_syn = fopen(argv[2], "wb") ) == NULL ) { printf("%s - Error opening file %s !!\n", argv[0], argv[2]); exit(0); } printf("Input bitstream file : %s\n",argv[1]); printf("Synthesis speech file : %s\n",argv[2]); /*-----------------------------------------------------------------* * Initialization of decoder * *-----------------------------------------------------------------*/ for (i=0; i<M; i++) synth_buf[i] = 0; synth = synth_buf + M; bad_lsf = 0; /* Initialize bad LSF indicator */ Init_Decod_ld8a(); Init_Post_Filter(); Init_Post_Process(); /*-----------------------------------------------------------------* * Loop for each "L_FRAME" speech data * *-----------------------------------------------------------------*/ Word16 iBuffer[5]; frame = 0; while( fread(iBuffer, sizeof(Word16), 5, f_serial) == 5) { covertBit2Serial(iBuffer,serial); printf("Frame =%d\r", frame++); bits2prm_ld8k( &serial[2], &parm[1]); /* the hardware detects frame erasures by checking if all bits are set to zero */ parm[0] = 0; /* No frame erasure */ for (i=2; i < SERIAL_SIZE; i++) if (serial[i] == 0 ) parm[0] = 1; /* frame erased */ /* check pitch parity and put 1 in parm[4] if parity error */ parm[4] = Check_Parity_Pitch(parm[3], parm[4]); Decod_ld8a(parm, synth, Az_dec, T2); Post_Filter(synth, Az_dec, T2); /* Post-filter */ Post_Process(synth, L_FRAME); fwrite(synth, sizeof(short), L_FRAME, f_syn); } return(0); } void covertBit2Serial(Word16 *input, Word16 *output){ output[0]=SYNC_WORD; output[1]=SIZE_WORD; int outIndex =2; int i,j; Word16 Mask; for (i=0;i<5;i++){ for(j=0;j<16;j++){ Mask = 0x0001 << (15-j); if ((Mask | input[i]) == input[i]) output[outIndex++] = BIT_1; else output[outIndex++] = BIT_0; } } } void printbits(Word16 value){ const int SHIFT = 8 * sizeof( Word16 ) - 1; const Word16 MASK = 1 << SHIFT; //getchar(); //printf("buffer is \n"); int k = 0; for (k = 1; k <= SHIFT + 1; k++ ) { printf("%c",value & MASK ? '1' : '0' ); value <<= 1; // if ( k == 8) // printf( ' '); } //printf("\n"); }

         pj_str_t codec_id = pj_str("g729");
         pjmedia_codec_mgr_set_codec_priority(
             pjmedia_endpt_get_codec_mgr(pjsua_var.med_endpt),
             &codec_id, PJMEDIA_CODEC_PRIO_HIGHEST);

       Thanks

It seems the g729_enum_codecs() is well defined. So please make sure that pjmedia_codec_g729_init() is called, e.g: set PJMEDIA_HAS_G729_CODEC to 1 in your config_site.h. Regards, nanang On Sun, Nov 9, 2008 at 2:11 PM, S. M. Nazmul Hasan (Opu) <apus29@gmail.com> wrote: > Hi Nanang, > > I have tested it to make a self call and getting 488(Not acceptable here) > response. i saw > the log file and the sdp , there speex, gsm , pcma and pcmu only presents . > and no g729. > so it looks like that g729 is not registered. and server rejected the call. > > I have checked the log file of server , there was a message: > > No compatible codecs, not accepting this offer! > > please give me a suggestion. > > Thanks > > opu > > On Thu, Nov 6, 2008 at 4:26 PM, Nanang Izzuddin <nanang@pjsip.org> wrote: >> >> Hi, >> >> I think it would be better if you see the log file in the rejecting >> side (it can be the proxy/asterisk or x-lite) to see why it rejected >> the call. If you have pjsua with IPP enabled, you can just make a call >> to the pjsua and send along the pjsua log file. >> >> The log file in the Symbian side may help, but I am not sure it will >> help a lot, however to generate the log file on Symbian, just specify >> log filename in logging config for the pjsua_init(), e.g: >> log_cfg.log_filename = pj_str("c:\\data\\pjsua.log"); >> >> Regards, >> nanang >> >> >> On Thu, Nov 6, 2008 at 3:02 PM, S. M. Nazmul Hasan (Opu) >> <apus29@gmail.com> wrote: >> > Hi Nanag, >> > >> > Yes the other endpoint supports g729. my tested environment was: >> > >> > server: AsteriskNow (which supports g729) >> > client: pjsip (symbian_ua_gui) and Xlite Pro(which supports g729) >> > >> > 1. when i am using both endpoint as Xlite pro and fixing the codec as >> > g729 >> > only then both using g729 and call established fine. >> > >> > 2. But when i am trying with symbian_ua_gui with Xlite Pro fixing the >> > codec >> > to g729 then i am getting 488(Not acceptable here) response. >> > >> > Can you please tell me how can I get the log file for No. 2 test >> > criteria. >> > >> > Thanks >> > >> > Opu >> > >> > On Wed, Nov 5, 2008 at 5:00 PM, Nanang Izzuddin <nanang@pjsip.org> >> > wrote: >> >> >> >> Hi, >> >> >> >> Just in case, please make sure the other endpoint supports G729 too, >> >> e.g: make a call to PJSUA on desktop with IPP G729 enabled. And please >> >> check the pjsua log, see if G729 is in the INVITE SDP. If it is >> >> rejected, you may see the reason in the pjsua log. >> >> >> >> Regards, >> >> nanang >> >> >> >> >> >> On Wed, Nov 5, 2008 at 1:39 PM, S. M. Nazmul Hasan (Opu) >> >> <apus29@gmail.com> wrote: >> >> > Nanag, >> >> > >> >> > I think so Nanang, may be its not registering because i am getting >> >> > response >> >> > 488(Not acceptable here). But i did it register and unregister in >> >> > pjsua_media.c. >> >> > >> >> > Can you please see these files. >> >> > >> >> > Thanks >> >> > >> >> > Opu >> >> > >> >> > On Tue, Nov 4, 2008 at 10:18 PM, Nanang Izzuddin <nanang@pjsip.org> >> >> > wrote: >> >> >> >> >> >> Hi, >> >> >> >> >> >> Just did a quick skimming, things I can found so far: >> >> >> 1. For 8000kbps and 10ms frame time, the encoded frame size should >> >> >> be >> >> >> 10 bytes, not 20 bytes. >> >> >> 2. this line: >> >> >> pcm_in += 160; >> >> >> should be: >> >> >> pcm_in += 80; >> >> >> >> >> >> Numbers in a codec wrapper are very important, so please check and >> >> >> recheck carefully. >> >> >> >> >> >> Regards, >> >> >> nanang >> >> >> >> >> >> >> >> >> On Tue, Nov 4, 2008 at 8:42 PM, S. M. Nazmul Hasan (Opu) >> >> >> <apus29@gmail.com> wrote: >> >> >> > Dear Rawshan, >> >> >> > >> >> >> > Can you please check the file if it is ok to add voiceage g729 in >> >> >> > pjsip >> >> >> > for >> >> >> > symbian. i am trying this for several days but i can get it >> >> >> > working >> >> >> > for >> >> >> > me. >> >> >> > >> >> >> > Help from anyone would be great appretiable. >> >> >> > >> >> >> > Thanks >> >> >> > >> >> >> > On Mon, Nov 3, 2008 at 3:17 PM, Rawshan Iajdani >> >> >> > <iajdani@provati.com> >> >> >> > wrote: >> >> >> >> >> >> >> >> You should be able to do that in pjlib/include/pj/config_site.h >> >> >> >> and >> >> >> >> pjmedia/include/pjmedia-codec/config.h file. But always remember >> >> >> >> config_site.h settings will overwrite any settings done in >> >> >> >> config.h >> >> >> >> file. >> >> >> >> Good luck… >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> From: pjsip-bounces@lists.pjsip.org >> >> >> >> [mailto:pjsip-bounces@lists.pjsip.org] >> >> >> >> On Behalf Of S. M. Nazmul Hasan (Opu) >> >> >> >> Sent: Saturday, November 01, 2008 6:01 PM >> >> >> >> To: pjsip list >> >> >> >> Subject: Re: [pjsip] g729 codec >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> Thanks Lajdani. My jumped start is running well. >> >> >> >> >> >> >> >> I am using Brekeke sip server to check the active session. and i >> >> >> >> am >> >> >> >> getting the using codec (payload) is always PCMU/8000. even >> >> >> >> after >> >> >> >> changing >> >> >> >> the g729 for highest priority. >> >> >> >> >> >> >> >> pj_str_t codec_id = pj_str("g729"); >> >> >> >> pjmedia_codec_mgr_set_codec_priority( >> >> >> >> pjmedia_endpt_get_codec_mgr(pjsua_var.med_endpt), >> >> >> >> &codec_id, PJMEDIA_CODEC_PRIO_HIGHEST); >> >> >> >> >> >> >> >> when i changed it to PJMEDIA_CODEC_PRIO_LOWEST the result was >> >> >> >> same. >> >> >> >> even >> >> >> >> after disabling all the codecs and after deleting all g711 files >> >> >> >> and >> >> >> >> related >> >> >> >> fields it showed the payload is PCMU/8000. >> >> >> >> >> >> >> >> How can i easily disable all the codecs except g729. >> >> >> >> >> >> >> >> Thanks >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> On Thu, Oct 30, 2008 at 3:43 AM, Rawshan Iajdani >> >> >> >> <iajdani@provati.com> >> >> >> >> wrote: >> >> >> >> >> >> >> >> Well.. your attribute settings seems fine. For the 33 it will be >> >> >> >> 20. >> >> >> >> And >> >> >> >> in place of 320, it will be 160.. hope this will give u a jump >> >> >> >> start.. >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> Rawshan Iajdani >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> From: pjsip-bounces@lists.pjsip.org >> >> >> >> [mailto:pjsip-bounces@lists.pjsip.org] >> >> >> >> On Behalf Of S. M. Nazmul Hasan (Opu) >> >> >> >> Sent: Wednesday, October 29, 2008 6:12 PM >> >> >> >> To: pjsip list >> >> >> >> Subject: [pjsip] g729 codec >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> Hi Benny, >> >> >> >> >> >> >> >> I am trying to add voice-age g729 codec with PJSIP for symbian. >> >> >> >> but >> >> >> >> i >> >> >> >> am >> >> >> >> getting some difficulties while following the gsm.c file. >> >> >> >> >> >> >> >> 1. while creating g729 codec private data >> >> >> >> what should i declare in replace of >> >> >> >> // struct gsm_state *encoder; >> >> >> >> // struct gsm_state *decoder; >> >> >> >> >> >> >> >> in "l16.c" its only use the frame size .. >> >> >> >> >> >> >> >> 2. in Generate default attribute. >> >> >> >> >> >> >> >> i have changed the value to this. is it ok? >> >> >> >> attr->info.clock_rate = 8000; >> >> >> >> attr->info.channel_cnt = 1; >> >> >> >> attr->info.avg_bps = 8000; >> >> >> >> attr->info.max_bps = 8000; >> >> >> >> attr->info.pcm_bits_per_sample = 16; >> >> >> >> attr->info.frm_ptime = 10; >> >> >> >> >> >> >> >> 3. In gsm.c there is some hard coding in gsm_codec_parse( >> >> >> >> -- what will be the value in replace of "33" here >> >> >> >> >> >> >> >> 4. frame_size and sample_per_frame are also not clear to me.. >> >> >> >> >> >> >> >> Sorry for lot of questions and may be those are lot easier to >> >> >> >> ask.. >> >> >> >> but >> >> >> >> really i am not good in codec.. >> >> >> >> >> >> >> >> waiting for reply. >> >> >> >> >> >> >> >> Thanks >> >> >> >> >> >> >> >> S. M. Nazmul Hasan Opu >> >> >> >> Software Engineer >> >> >> >> R & D Application >> >> >> >> Dhaka, Bangladesh >> >> >> >> Mob: +880 1712 901 764 >> >> >> >> >> >> >> >> _______________________________________________ >> >> >> >> Visit our blog: http://blog.pjsip.org >> >> >> >> >> >> >> >> pjsip mailing list >> >> >> >> pjsip@lists.pjsip.org >> >> >> >> http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org >> >> >> >> >> >> >> >> >> >> >> >> -- >> >> >> >> S. M. Nazmul Hasan Opu >> >> >> >> Software Engineer >> >> >> >> R & D Application >> >> >> >> Dhaka, Bangladesh >> >> >> >> Mob: +880 1712 901 764 >> >> >> >> >> >> >> >> _______________________________________________ >> >> >> >> Visit our blog: http://blog.pjsip.org >> >> >> >> >> >> >> >> pjsip mailing list >> >> >> >> pjsip@lists.pjsip.org >> >> >> >> http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org >> >> >> >> >> >> >> > >> >> >> > >> >> >> > >> >> >> > -- >> >> >> > S. M. Nazmul Hasan Opu >> >> >> > Software Engineer >> >> >> > R & D Application >> >> >> > Dhaka, Bangladesh >> >> >> > Mob: +880 1712 901 764 >> >> >> > >> >> >> > _______________________________________________ >> >> >> > Visit our blog: http://blog.pjsip.org >> >> >> > >> >> >> > pjsip mailing list >> >> >> > pjsip@lists.pjsip.org >> >> >> > http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org >> >> >> > >> >> >> > >> >> >> >> >> >> _______________________________________________ >> >> >> Visit our blog: http://blog.pjsip.org >> >> >> >> >> >> pjsip mailing list >> >> >> pjsip@lists.pjsip.org >> >> >> http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org >> >> > >> >> > >> >> > >> >> > -- >> >> > S. M. Nazmul Hasan Opu >> >> > Software Engineer >> >> > R & D Application >> >> > Dhaka, Bangladesh >> >> > Mob: +880 1712 901 764 >> >> > >> >> > _______________________________________________ >> >> > Visit our blog: http://blog.pjsip.org >> >> > >> >> > pjsip mailing list >> >> > pjsip@lists.pjsip.org >> >> > http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org >> >> > >> >> > >> >> >> >> _______________________________________________ >> >> Visit our blog: http://blog.pjsip.org >> >> >> >> pjsip mailing list >> >> pjsip@lists.pjsip.org >> >> http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org >> > >> > >> > >> > -- >> > S. M. Nazmul Hasan Opu >> > Software Engineer >> > R & D Application >> > Dhaka, Bangladesh >> > Mob: +880 1712 901 764 >> > >> > _______________________________________________ >> > Visit our blog: http://blog.pjsip.org >> > >> > pjsip mailing list >> > pjsip@lists.pjsip.org >> > http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org >> > >> > >> >> _______________________________________________ >> Visit our blog: http://blog.pjsip.org >> >> pjsip mailing list >> pjsip@lists.pjsip.org >> http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org > > > > -- > S. M. Nazmul Hasan Opu > Software Engineer > R & D Application > Dhaka, Bangladesh > Mob: +880 1712 901 764 > > _______________________________________________ > Visit our blog: http://blog.pjsip.org > > pjsip mailing list > pjsip@lists.pjsip.org > http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org > >

/* Passed arguments */



if ( argc != 3)

  {

     printf("Usage :%s bitstream_file  outputspeech_file\n",argv[0]);

     printf("\n");

     printf("Format for bitstream_file:\n");

     printf("  One (2-byte) synchronization word \n");

     printf("  One (2-byte) size word,\n");

     printf("  80 words (2-byte) containing 80 bits.\n");

     printf("\n");

     printf("Format for outputspeech_file:\n");

     printf("  Synthesis is written to a binary file of 16 bits

     exit( 1 );

  }



/* Open file for synthesis and packed serial stream */



if( (f_serial = fopen(argv[1],"rb") ) == NULL )

  {

     printf("%s - Error opening file  %s !!\n", argv[0], argv[1]);

     exit(0);

  }



if( (f_syn = fopen(argv[2], "wb") ) == NULL )

  {

     printf("%s - Error opening file  %s !!\n", argv[0], argv[2]);

     exit(0);

  }



printf("Input bitstream file  :   %s\n",argv[1]);

printf("Synthesis speech file :   %s\n",argv[2]);

 covertBit2Serial(iBuffer,serial);



 printf("Frame =%d\r", frame++);



 bits2prm_ld8k( &serial[2], &parm[1]);



 /* the hardware detects frame erasures by checking if all bits

    are set to zero

  */

 parm[0] = 0;           /* No frame erasure */

 for (i=2; i < SERIAL_SIZE; i++)

   if (serial[i] == 0 ) parm[0] = 1; /* frame erased     */



 /* check pitch parity and put 1 in parm[4] if parity error */



 parm[4] = Check_Parity_Pitch(parm[3], parm[4]);



 Decod_ld8a(parm, synth, Az_dec, T2);



 Post_Filter(synth, Az_dec, T2);        /* Post-filter */



 Post_Process(synth, L_FRAME);



 fwrite(synth, sizeof(short), L_FRAME, f_syn);

   for(j=0;j<16;j++){

   Mask = 0x0001 << (15-j);



   if ((Mask | input[i]) == input[i])

         output[outIndex++] = BIT_1;



   else

         output[outIndex++] = BIT_0;



   }



}

const Word16 MASK = 1 << SHIFT;

for (k = 1; k <= SHIFT + 1; k++ )

{

   printf("%c",value & MASK ? '1' : '0' );



   value <<= 1;



  // if ( k == 8)

   //   printf( ' ');

}

Hi, Sorry, in case this is for me (IIRC I have never seen another Nanang in this mailing list :D), then I am afraid you've mistaken, I have never integrated any ITU codecs into PJSIP. Regards, nanang On Mon, Nov 10, 2008 at 2:43 AM, Rawshan Iajdani <iajdani@provati.com> wrote: > I know you implemented ITU version of codecs. I included my test file where > I tried to keep everything same as ITU, except that I convert the 10 bit > frames to 80 16bit samples. After that I decode that to PCM. But the sound > I get after decoding is just some metallic scrambled. Am I missing anything > here? Look at the routine covertBit2Serial(); That was what written by me. > > > > > > > > > > > > > > void covertBit2Serial(Word16 *input, Word16 *output); > > void printbits(Word16 value); > > > > int main(int argc, char *argv[] ) > > { > > Word16 synth_buf[L_FRAME+M], *synth; /* Synthesis */ > > Word16 parm[PRM_SIZE+1]; /* Synthesis parameters */ > > Word16 serial[SERIAL_SIZE]; /* Serial stream */ > > Word16 Az_dec[MP1*2]; /* Decoded Az for post-filter */ > > Word16 T2[2]; /* Pitch lag for 2 subframes */ > > > > > > Word16 i, frame; > > FILE *f_syn, *f_serial; > > > > printf("\n"); > > printf("************ G.729a 8.0 KBIT/S SPEECH DECODER ************\n"); > > printf("\n"); > > printf("------------------- Fixed point C simulation ----------------\n"); > > printf("\n"); > > printf("------------ Version 1.1 (Release 2, November 2006) --------\n"); > > printf("\n"); > > > > /* Passed arguments */ > > > > if ( argc != 3) > > { > > printf("Usage :%s bitstream_file outputspeech_file\n",argv[0]); > > printf("\n"); > > printf("Format for bitstream_file:\n"); > > printf(" One (2-byte) synchronization word \n"); > > printf(" One (2-byte) size word,\n"); > > printf(" 80 words (2-byte) containing 80 bits.\n"); > > printf("\n"); > > printf("Format for outputspeech_file:\n"); > > printf(" Synthesis is written to a binary file of 16 bits > data.\n"); > > exit( 1 ); > > } > > > > /* Open file for synthesis and packed serial stream */ > > > > if( (f_serial = fopen(argv[1],"rb") ) == NULL ) > > { > > printf("%s - Error opening file %s !!\n", argv[0], argv[1]); > > exit(0); > > } > > > > if( (f_syn = fopen(argv[2], "wb") ) == NULL ) > > { > > printf("%s - Error opening file %s !!\n", argv[0], argv[2]); > > exit(0); > > } > > > > printf("Input bitstream file : %s\n",argv[1]); > > printf("Synthesis speech file : %s\n",argv[2]); > > > > /*-----------------------------------------------------------------* > > * Initialization of decoder * > > *-----------------------------------------------------------------*/ > > > > for (i=0; i<M; i++) synth_buf[i] = 0; > > synth = synth_buf + M; > > > > bad_lsf = 0; /* Initialize bad LSF indicator */ > > Init_Decod_ld8a(); > > Init_Post_Filter(); > > Init_Post_Process(); > > > > > > /*-----------------------------------------------------------------* > > * Loop for each "L_FRAME" speech data * > > *-----------------------------------------------------------------*/ > > Word16 iBuffer[5]; > > > > frame = 0; > > while( fread(iBuffer, sizeof(Word16), 5, f_serial) == 5) > > { > > covertBit2Serial(iBuffer,serial); > > > > printf("Frame =%d\r", frame++); > > > > bits2prm_ld8k( &serial[2], &parm[1]); > > > > /* the hardware detects frame erasures by checking if all bits > > are set to zero > > */ > > parm[0] = 0; /* No frame erasure */ > > for (i=2; i < SERIAL_SIZE; i++) > > if (serial[i] == 0 ) parm[0] = 1; /* frame erased */ > > > > /* check pitch parity and put 1 in parm[4] if parity error */ > > > > parm[4] = Check_Parity_Pitch(parm[3], parm[4]); > > > > Decod_ld8a(parm, synth, Az_dec, T2); > > > > Post_Filter(synth, Az_dec, T2); /* Post-filter */ > > > > Post_Process(synth, L_FRAME); > > > > fwrite(synth, sizeof(short), L_FRAME, f_syn); > > > > } > > return(0); > > } > > > > > > > > > > void covertBit2Serial(Word16 *input, Word16 *output){ > > output[0]=SYNC_WORD; > > output[1]=SIZE_WORD; > > int outIndex =2; > > int i,j; > > Word16 Mask; > > for (i=0;i<5;i++){ > > for(j=0;j<16;j++){ > > Mask = 0x0001 << (15-j); > > > > if ((Mask | input[i]) == input[i]) > > output[outIndex++] = BIT_1; > > > > else > > output[outIndex++] = BIT_0; > > > > } > > > > } > > } > > > > > > > > void printbits(Word16 value){ > > const int SHIFT = 8 * sizeof( Word16 ) - 1; > > const Word16 MASK = 1 << SHIFT; > > //getchar(); > > > > //printf("buffer is \n"); > > > > int k = 0; > > for (k = 1; k <= SHIFT + 1; k++ ) > > { > > printf("%c",value & MASK ? '1' : '0' ); > > > > value <<= 1; > > > > // if ( k == 8) > > // printf( ' '); > > } > > //printf("\n"); > > } > > > > > > _______________________________________________ > Visit our blog: http://blog.pjsip.org > > pjsip mailing list > pjsip@lists.pjsip.org > http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org > >

          PJ_LOG(4,(THIS_FILE, "............writing

         pj_str_t codec_id = pj_str("g729");
         pjmedia_codec_mgr_set_codec_priority(

             &codec_id, PJMEDIA_CODEC_PRIO_HIGHEST);

       Thanks

Nanag, Can you please tell me how i can debug the various source file of pjmedia or pjsip library. For example : i have tried to debug g729.c as well as gsm.c by using the line PJ_LOG(4,(THIS_FILE, "............writing something........")); this should be shown in the log file ....right? I have put the in pjmedia_codec_g729_init( pjmedia_endpt *endpt ) and pjmedia_codec_gsm_init( pjmedia_endpt *endpt ) function... but no result is shown in the log file. actually this is not the right debugging procedure.. so how can i understand whether any function is being called or not... can you please clarify me.... Regards OPu On Mon, Nov 10, 2008 at 2:09 PM, Nanang Izzuddin <nanang@pjsip.org> wrote: > It seems the g729_enum_codecs() is well defined. So please make sure > that pjmedia_codec_g729_init() is called, e.g: set > PJMEDIA_HAS_G729_CODEC to 1 in your config_site.h. > > Regards, > nanang > > > On Sun, Nov 9, 2008 at 2:11 PM, S. M. Nazmul Hasan (Opu) > <apus29@gmail.com> wrote: > > Hi Nanang, > > > > I have tested it to make a self call and getting 488(Not acceptable here) > > response. i saw > > the log file and the sdp , there speex, gsm , pcma and pcmu only presents > . > > and no g729. > > so it looks like that g729 is not registered. and server rejected the > call. > > > > I have checked the log file of server , there was a message: > > > > No compatible codecs, not accepting this offer! > > > > please give me a suggestion. > > > > Thanks > > > > opu > > > > On Thu, Nov 6, 2008 at 4:26 PM, Nanang Izzuddin <nanang@pjsip.org> > wrote: > >> > >> Hi, > >> > >> I think it would be better if you see the log file in the rejecting > >> side (it can be the proxy/asterisk or x-lite) to see why it rejected > >> the call. If you have pjsua with IPP enabled, you can just make a call > >> to the pjsua and send along the pjsua log file. > >> > >> The log file in the Symbian side may help, but I am not sure it will > >> help a lot, however to generate the log file on Symbian, just specify > >> log filename in logging config for the pjsua_init(), e.g: > >> log_cfg.log_filename = pj_str("c:\\data\\pjsua.log"); > >> > >> Regards, > >> nanang > >> > >> > >> On Thu, Nov 6, 2008 at 3:02 PM, S. M. Nazmul Hasan (Opu) > >> <apus29@gmail.com> wrote: > >> > Hi Nanag, > >> > > >> > Yes the other endpoint supports g729. my tested environment was: > >> > > >> > server: AsteriskNow (which supports g729) > >> > client: pjsip (symbian_ua_gui) and Xlite Pro(which supports g729) > >> > > >> > 1. when i am using both endpoint as Xlite pro and fixing the codec as > >> > g729 > >> > only then both using g729 and call established fine. > >> > > >> > 2. But when i am trying with symbian_ua_gui with Xlite Pro fixing the > >> > codec > >> > to g729 then i am getting 488(Not acceptable here) response. > >> > > >> > Can you please tell me how can I get the log file for No. 2 test > >> > criteria. > >> > > >> > Thanks > >> > > >> > Opu > >> > > >> > On Wed, Nov 5, 2008 at 5:00 PM, Nanang Izzuddin <nanang@pjsip.org> > >> > wrote: > >> >> > >> >> Hi, > >> >> > >> >> Just in case, please make sure the other endpoint supports G729 too, > >> >> e.g: make a call to PJSUA on desktop with IPP G729 enabled. And > please > >> >> check the pjsua log, see if G729 is in the INVITE SDP. If it is > >> >> rejected, you may see the reason in the pjsua log. > >> >> > >> >> Regards, > >> >> nanang > >> >> > >> >> > >> >> On Wed, Nov 5, 2008 at 1:39 PM, S. M. Nazmul Hasan (Opu) > >> >> <apus29@gmail.com> wrote: > >> >> > Nanag, > >> >> > > >> >> > I think so Nanang, may be its not registering because i am getting > >> >> > response > >> >> > 488(Not acceptable here). But i did it register and unregister in > >> >> > pjsua_media.c. > >> >> > > >> >> > Can you please see these files. > >> >> > > >> >> > Thanks > >> >> > > >> >> > Opu > >> >> > > >> >> > On Tue, Nov 4, 2008 at 10:18 PM, Nanang Izzuddin <nanang@pjsip.org > > > >> >> > wrote: > >> >> >> > >> >> >> Hi, > >> >> >> > >> >> >> Just did a quick skimming, things I can found so far: > >> >> >> 1. For 8000kbps and 10ms frame time, the encoded frame size should > >> >> >> be > >> >> >> 10 bytes, not 20 bytes. > >> >> >> 2. this line: > >> >> >> pcm_in += 160; > >> >> >> should be: > >> >> >> pcm_in += 80; > >> >> >> > >> >> >> Numbers in a codec wrapper are very important, so please check and > >> >> >> recheck carefully. > >> >> >> > >> >> >> Regards, > >> >> >> nanang > >> >> >> > >> >> >> > >> >> >> On Tue, Nov 4, 2008 at 8:42 PM, S. M. Nazmul Hasan (Opu) > >> >> >> <apus29@gmail.com> wrote: > >> >> >> > Dear Rawshan, > >> >> >> > > >> >> >> > Can you please check the file if it is ok to add voiceage g729 > in > >> >> >> > pjsip > >> >> >> > for > >> >> >> > symbian. i am trying this for several days but i can get it > >> >> >> > working > >> >> >> > for > >> >> >> > me. > >> >> >> > > >> >> >> > Help from anyone would be great appretiable. > >> >> >> > > >> >> >> > Thanks > >> >> >> > > >> >> >> > On Mon, Nov 3, 2008 at 3:17 PM, Rawshan Iajdani > >> >> >> > <iajdani@provati.com> > >> >> >> > wrote: > >> >> >> >> > >> >> >> >> You should be able to do that in pjlib/include/pj/config_site.h > >> >> >> >> and > >> >> >> >> pjmedia/include/pjmedia-codec/config.h file. But always > remember > >> >> >> >> config_site.h settings will overwrite any settings done in > >> >> >> >> config.h > >> >> >> >> file. > >> >> >> >> Good luck… > >> >> >> >> > >> >> >> >> > >> >> >> >> > >> >> >> >> > >> >> >> >> > >> >> >> >> > >> >> >> >> > >> >> >> >> > >> >> >> >> > >> >> >> >> From: pjsip-bounces@lists.pjsip.org > >> >> >> >> [mailto:pjsip-bounces@lists.pjsip.org] > >> >> >> >> On Behalf Of S. M. Nazmul Hasan (Opu) > >> >> >> >> Sent: Saturday, November 01, 2008 6:01 PM > >> >> >> >> To: pjsip list > >> >> >> >> Subject: Re: [pjsip] g729 codec > >> >> >> >> > >> >> >> >> > >> >> >> >> > >> >> >> >> Thanks Lajdani. My jumped start is running well. > >> >> >> >> > >> >> >> >> I am using Brekeke sip server to check the active session. and > i > >> >> >> >> am > >> >> >> >> getting the using codec (payload) is always PCMU/8000. even > >> >> >> >> after > >> >> >> >> changing > >> >> >> >> the g729 for highest priority. > >> >> >> >> > >> >> >> >> pj_str_t codec_id = pj_str("g729"); > >> >> >> >> pjmedia_codec_mgr_set_codec_priority( > >> >> >> >> > pjmedia_endpt_get_codec_mgr(pjsua_var.med_endpt), > >> >> >> >> &codec_id, PJMEDIA_CODEC_PRIO_HIGHEST); > >> >> >> >> > >> >> >> >> when i changed it to PJMEDIA_CODEC_PRIO_LOWEST the result was > >> >> >> >> same. > >> >> >> >> even > >> >> >> >> after disabling all the codecs and after deleting all g711 > files > >> >> >> >> and > >> >> >> >> related > >> >> >> >> fields it showed the payload is PCMU/8000. > >> >> >> >> > >> >> >> >> How can i easily disable all the codecs except g729. > >> >> >> >> > >> >> >> >> Thanks > >> >> >> >> > >> >> >> >> > >> >> >> >> > >> >> >> >> On Thu, Oct 30, 2008 at 3:43 AM, Rawshan Iajdani > >> >> >> >> <iajdani@provati.com> > >> >> >> >> wrote: > >> >> >> >> > >> >> >> >> Well.. your attribute settings seems fine. For the 33 it will > be > >> >> >> >> 20. > >> >> >> >> And > >> >> >> >> in place of 320, it will be 160.. hope this will give u a jump > >> >> >> >> start.. > >> >> >> >> > >> >> >> >> > >> >> >> >> > >> >> >> >> > >> >> >> >> > >> >> >> >> > >> >> >> >> > >> >> >> >> Rawshan Iajdani > >> >> >> >> > >> >> >> >> > >> >> >> >> > >> >> >> >> > >> >> >> >> > >> >> >> >> From: pjsip-bounces@lists.pjsip.org > >> >> >> >> [mailto:pjsip-bounces@lists.pjsip.org] > >> >> >> >> On Behalf Of S. M. Nazmul Hasan (Opu) > >> >> >> >> Sent: Wednesday, October 29, 2008 6:12 PM > >> >> >> >> To: pjsip list > >> >> >> >> Subject: [pjsip] g729 codec > >> >> >> >> > >> >> >> >> > >> >> >> >> > >> >> >> >> Hi Benny, > >> >> >> >> > >> >> >> >> I am trying to add voice-age g729 codec with PJSIP for symbian. > >> >> >> >> but > >> >> >> >> i > >> >> >> >> am > >> >> >> >> getting some difficulties while following the gsm.c file. > >> >> >> >> > >> >> >> >> 1. while creating g729 codec private data > >> >> >> >> what should i declare in replace of > >> >> >> >> // struct gsm_state *encoder; > >> >> >> >> // struct gsm_state *decoder; > >> >> >> >> > >> >> >> >> in "l16.c" its only use the frame size .. > >> >> >> >> > >> >> >> >> 2. in Generate default attribute. > >> >> >> >> > >> >> >> >> i have changed the value to this. is it ok? > >> >> >> >> attr->info.clock_rate = 8000; > >> >> >> >> attr->info.channel_cnt = 1; > >> >> >> >> attr->info.avg_bps = 8000; > >> >> >> >> attr->info.max_bps = 8000; > >> >> >> >> attr->info.pcm_bits_per_sample = 16; > >> >> >> >> attr->info.frm_ptime = 10; > >> >> >> >> > >> >> >> >> 3. In gsm.c there is some hard coding in gsm_codec_parse( > >> >> >> >> -- what will be the value in replace of "33" here > >> >> >> >> > >> >> >> >> 4. frame_size and sample_per_frame are also not clear to me.. > >> >> >> >> > >> >> >> >> Sorry for lot of questions and may be those are lot easier to > >> >> >> >> ask.. > >> >> >> >> but > >> >> >> >> really i am not good in codec.. > >> >> >> >> > >> >> >> >> waiting for reply. > >> >> >> >> > >> >> >> >> Thanks > >> >> >> >> > >> >> >> >> S. M. Nazmul Hasan Opu > >> >> >> >> Software Engineer > >> >> >> >> R & D Application > >> >> >> >> Dhaka, Bangladesh > >> >> >> >> Mob: +880 1712 901 764 > >> >> >> >> > >> >> >> >> _______________________________________________ > >> >> >> >> Visit our blog: http://blog.pjsip.org > >> >> >> >> > >> >> >> >> pjsip mailing list > >> >> >> >> pjsip@lists.pjsip.org > >> >> >> >> http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org > >> >> >> >> > >> >> >> >> > >> >> >> >> -- > >> >> >> >> S. M. Nazmul Hasan Opu > >> >> >> >> Software Engineer > >> >> >> >> R & D Application > >> >> >> >> Dhaka, Bangladesh > >> >> >> >> Mob: +880 1712 901 764 > >> >> >> >> > >> >> >> >> _______________________________________________ > >> >> >> >> Visit our blog: http://blog.pjsip.org > >> >> >> >> > >> >> >> >> pjsip mailing list > >> >> >> >> pjsip@lists.pjsip.org > >> >> >> >> http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org > >> >> >> >> > >> >> >> > > >> >> >> > > >> >> >> > > >> >> >> > -- > >> >> >> > S. M. Nazmul Hasan Opu > >> >> >> > Software Engineer > >> >> >> > R & D Application > >> >> >> > Dhaka, Bangladesh > >> >> >> > Mob: +880 1712 901 764 > >> >> >> > > >> >> >> > _______________________________________________ > >> >> >> > Visit our blog: http://blog.pjsip.org > >> >> >> > > >> >> >> > pjsip mailing list > >> >> >> > pjsip@lists.pjsip.org > >> >> >> > http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org > >> >> >> > > >> >> >> > > >> >> >> > >> >> >> _______________________________________________ > >> >> >> Visit our blog: http://blog.pjsip.org > >> >> >> > >> >> >> pjsip mailing list > >> >> >> pjsip@lists.pjsip.org > >> >> >> http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org > >> >> > > >> >> > > >> >> > > >> >> > -- > >> >> > S. M. Nazmul Hasan Opu > >> >> > Software Engineer > >> >> > R & D Application > >> >> > Dhaka, Bangladesh > >> >> > Mob: +880 1712 901 764 > >> >> > > >> >> > _______________________________________________ > >> >> > Visit our blog: http://blog.pjsip.org > >> >> > > >> >> > pjsip mailing list > >> >> > pjsip@lists.pjsip.org > >> >> > http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org > >> >> > > >> >> > > >> >> > >> >> _______________________________________________ > >> >> Visit our blog: http://blog.pjsip.org > >> >> > >> >> pjsip mailing list > >> >> pjsip@lists.pjsip.org > >> >> http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org > >> > > >> > > >> > > >> > -- > >> > S. M. Nazmul Hasan Opu > >> > Software Engineer > >> > R & D Application > >> > Dhaka, Bangladesh > >> > Mob: +880 1712 901 764 > >> > > >> > _______________________________________________ > >> > Visit our blog: http://blog.pjsip.org > >> > > >> > pjsip mailing list > >> > pjsip@lists.pjsip.org > >> > http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org > >> > > >> > > >> > >> _______________________________________________ > >> Visit our blog: http://blog.pjsip.org > >> > >> pjsip mailing list > >> pjsip@lists.pjsip.org > >> http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org > > > > > > > > -- > > S. M. Nazmul Hasan Opu > > Software Engineer > > R & D Application > > Dhaka, Bangladesh > > Mob: +880 1712 901 764 > > > > _______________________________________________ > > Visit our blog: http://blog.pjsip.org > > > > pjsip mailing list > > pjsip@lists.pjsip.org > > http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org > > > > > > _______________________________________________ > Visit our blog: http://blog.pjsip.org > > pjsip mailing list > pjsip@lists.pjsip.org > http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org > -- S. M. Nazmul Hasan Opu Software Engineer R & D Application Dhaka, Bangladesh Mob: +880 1712 901 764

           PJ_LOG(4,(THIS_FILE, "............writing

         pj_str_t codec_id = pj_str("g729");
         pjmedia_codec_mgr_set_codec_priority(

       Thanks

Hi, If you are using Carbide, it shouldn't be a problem. However, using logging should be fine too, you just have to make sure the log level used, i.e: by default symbian_ua_gui uses log level 3, so any log messages with log level > 3 won't be printed. Regards, nanang On Mon, Nov 10, 2008 at 4:06 PM, S. M. Nazmul Hasan (Opu) <apus29@gmail.com> wrote: > Nanag, > > Can you please tell me how i can debug the various source file of pjmedia or > pjsip library. For example : i have tried to debug g729.c as well as gsm.c > by using the line > > PJ_LOG(4,(THIS_FILE, "............writing > something........")); > > this should be shown in the log file ....right? > > I have put the in > pjmedia_codec_g729_init( pjmedia_endpt *endpt ) and > > pjmedia_codec_gsm_init( pjmedia_endpt *endpt ) function... > > but no result is shown in the log file. actually this is not the right > debugging procedure.. > so how can i understand whether any function is being called or not... can > you please clarify > me.... > > Regards > > OPu > > On Mon, Nov 10, 2008 at 2:09 PM, Nanang Izzuddin <nanang@pjsip.org> wrote: >> >> It seems the g729_enum_codecs() is well defined. So please make sure >> that pjmedia_codec_g729_init() is called, e.g: set >> PJMEDIA_HAS_G729_CODEC to 1 in your config_site.h. >> >> Regards, >> nanang >> >> >> On Sun, Nov 9, 2008 at 2:11 PM, S. M. Nazmul Hasan (Opu) >> <apus29@gmail.com> wrote: >> > Hi Nanang, >> > >> > I have tested it to make a self call and getting 488(Not acceptable >> > here) >> > response. i saw >> > the log file and the sdp , there speex, gsm , pcma and pcmu only >> > presents . >> > and no g729. >> > so it looks like that g729 is not registered. and server rejected the >> > call. >> > >> > I have checked the log file of server , there was a message: >> > >> > No compatible codecs, not accepting this offer! >> > >> > please give me a suggestion. >> > >> > Thanks >> > >> > opu >> > >> > On Thu, Nov 6, 2008 at 4:26 PM, Nanang Izzuddin <nanang@pjsip.org> >> > wrote: >> >> >> >> Hi, >> >> >> >> I think it would be better if you see the log file in the rejecting >> >> side (it can be the proxy/asterisk or x-lite) to see why it rejected >> >> the call. If you have pjsua with IPP enabled, you can just make a call >> >> to the pjsua and send along the pjsua log file. >> >> >> >> The log file in the Symbian side may help, but I am not sure it will >> >> help a lot, however to generate the log file on Symbian, just specify >> >> log filename in logging config for the pjsua_init(), e.g: >> >> log_cfg.log_filename = pj_str("c:\\data\\pjsua.log"); >> >> >> >> Regards, >> >> nanang >> >> >> >> >> >> On Thu, Nov 6, 2008 at 3:02 PM, S. M. Nazmul Hasan (Opu) >> >> <apus29@gmail.com> wrote: >> >> > Hi Nanag, >> >> > >> >> > Yes the other endpoint supports g729. my tested environment was: >> >> > >> >> > server: AsteriskNow (which supports g729) >> >> > client: pjsip (symbian_ua_gui) and Xlite Pro(which supports g729) >> >> > >> >> > 1. when i am using both endpoint as Xlite pro and fixing the codec as >> >> > g729 >> >> > only then both using g729 and call established fine. >> >> > >> >> > 2. But when i am trying with symbian_ua_gui with Xlite Pro fixing the >> >> > codec >> >> > to g729 then i am getting 488(Not acceptable here) response. >> >> > >> >> > Can you please tell me how can I get the log file for No. 2 test >> >> > criteria. >> >> > >> >> > Thanks >> >> > >> >> > Opu >> >> > >> >> > On Wed, Nov 5, 2008 at 5:00 PM, Nanang Izzuddin <nanang@pjsip.org> >> >> > wrote: >> >> >> >> >> >> Hi, >> >> >> >> >> >> Just in case, please make sure the other endpoint supports G729 too, >> >> >> e.g: make a call to PJSUA on desktop with IPP G729 enabled. And >> >> >> please >> >> >> check the pjsua log, see if G729 is in the INVITE SDP. If it is >> >> >> rejected, you may see the reason in the pjsua log. >> >> >> >> >> >> Regards, >> >> >> nanang >> >> >> >> >> >> >> >> >> On Wed, Nov 5, 2008 at 1:39 PM, S. M. Nazmul Hasan (Opu) >> >> >> <apus29@gmail.com> wrote: >> >> >> > Nanag, >> >> >> > >> >> >> > I think so Nanang, may be its not registering because i am getting >> >> >> > response >> >> >> > 488(Not acceptable here). But i did it register and unregister in >> >> >> > pjsua_media.c. >> >> >> > >> >> >> > Can you please see these files. >> >> >> > >> >> >> > Thanks >> >> >> > >> >> >> > Opu >> >> >> > >> >> >> > On Tue, Nov 4, 2008 at 10:18 PM, Nanang Izzuddin >> >> >> > <nanang@pjsip.org> >> >> >> > wrote: >> >> >> >> >> >> >> >> Hi, >> >> >> >> >> >> >> >> Just did a quick skimming, things I can found so far: >> >> >> >> 1. For 8000kbps and 10ms frame time, the encoded frame size >> >> >> >> should >> >> >> >> be >> >> >> >> 10 bytes, not 20 bytes. >> >> >> >> 2. this line: >> >> >> >> pcm_in += 160; >> >> >> >> should be: >> >> >> >> pcm_in += 80; >> >> >> >> >> >> >> >> Numbers in a codec wrapper are very important, so please check >> >> >> >> and >> >> >> >> recheck carefully. >> >> >> >> >> >> >> >> Regards, >> >> >> >> nanang >> >> >> >> >> >> >> >> >> >> >> >> On Tue, Nov 4, 2008 at 8:42 PM, S. M. Nazmul Hasan (Opu) >> >> >> >> <apus29@gmail.com> wrote: >> >> >> >> > Dear Rawshan, >> >> >> >> > >> >> >> >> > Can you please check the file if it is ok to add voiceage g729 >> >> >> >> > in >> >> >> >> > pjsip >> >> >> >> > for >> >> >> >> > symbian. i am trying this for several days but i can get it >> >> >> >> > working >> >> >> >> > for >> >> >> >> > me. >> >> >> >> > >> >> >> >> > Help from anyone would be great appretiable. >> >> >> >> > >> >> >> >> > Thanks >> >> >> >> > >> >> >> >> > On Mon, Nov 3, 2008 at 3:17 PM, Rawshan Iajdani >> >> >> >> > <iajdani@provati.com> >> >> >> >> > wrote: >> >> >> >> >> >> >> >> >> >> You should be able to do that in >> >> >> >> >> pjlib/include/pj/config_site.h >> >> >> >> >> and >> >> >> >> >> pjmedia/include/pjmedia-codec/config.h file. But always >> >> >> >> >> remember >> >> >> >> >> config_site.h settings will overwrite any settings done in >> >> >> >> >> config.h >> >> >> >> >> file. >> >> >> >> >> Good luck… >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> From: pjsip-bounces@lists.pjsip.org >> >> >> >> >> [mailto:pjsip-bounces@lists.pjsip.org] >> >> >> >> >> On Behalf Of S. M. Nazmul Hasan (Opu) >> >> >> >> >> Sent: Saturday, November 01, 2008 6:01 PM >> >> >> >> >> To: pjsip list >> >> >> >> >> Subject: Re: [pjsip] g729 codec >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> Thanks Lajdani. My jumped start is running well. >> >> >> >> >> >> >> >> >> >> I am using Brekeke sip server to check the active session. and >> >> >> >> >> i >> >> >> >> >> am >> >> >> >> >> getting the using codec (payload) is always PCMU/8000. even >> >> >> >> >> after >> >> >> >> >> changing >> >> >> >> >> the g729 for highest priority. >> >> >> >> >> >> >> >> >> >> pj_str_t codec_id = pj_str("g729"); >> >> >> >> >> pjmedia_codec_mgr_set_codec_priority( >> >> >> >> >> >> >> >> >> >> pjmedia_endpt_get_codec_mgr(pjsua_var.med_endpt), >> >> >> >> >> &codec_id, PJMEDIA_CODEC_PRIO_HIGHEST); >> >> >> >> >> >> >> >> >> >> when i changed it to PJMEDIA_CODEC_PRIO_LOWEST the result was >> >> >> >> >> same. >> >> >> >> >> even >> >> >> >> >> after disabling all the codecs and after deleting all g711 >> >> >> >> >> files >> >> >> >> >> and >> >> >> >> >> related >> >> >> >> >> fields it showed the payload is PCMU/8000. >> >> >> >> >> >> >> >> >> >> How can i easily disable all the codecs except g729. >> >> >> >> >> >> >> >> >> >> Thanks >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> On Thu, Oct 30, 2008 at 3:43 AM, Rawshan Iajdani >> >> >> >> >> <iajdani@provati.com> >> >> >> >> >> wrote: >> >> >> >> >> >> >> >> >> >> Well.. your attribute settings seems fine. For the 33 it will >> >> >> >> >> be >> >> >> >> >> 20. >> >> >> >> >> And >> >> >> >> >> in place of 320, it will be 160.. hope this will give u a jump >> >> >> >> >> start.. >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> Rawshan Iajdani >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> From: pjsip-bounces@lists.pjsip.org >> >> >> >> >> [mailto:pjsip-bounces@lists.pjsip.org] >> >> >> >> >> On Behalf Of S. M. Nazmul Hasan (Opu) >> >> >> >> >> Sent: Wednesday, October 29, 2008 6:12 PM >> >> >> >> >> To: pjsip list >> >> >> >> >> Subject: [pjsip] g729 codec >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> Hi Benny, >> >> >> >> >> >> >> >> >> >> I am trying to add voice-age g729 codec with PJSIP for >> >> >> >> >> symbian. >> >> >> >> >> but >> >> >> >> >> i >> >> >> >> >> am >> >> >> >> >> getting some difficulties while following the gsm.c file. >> >> >> >> >> >> >> >> >> >> 1. while creating g729 codec private data >> >> >> >> >> what should i declare in replace of >> >> >> >> >> // struct gsm_state *encoder; >> >> >> >> >> // struct gsm_state *decoder; >> >> >> >> >> >> >> >> >> >> in "l16.c" its only use the frame size .. >> >> >> >> >> >> >> >> >> >> 2. in Generate default attribute. >> >> >> >> >> >> >> >> >> >> i have changed the value to this. is it ok? >> >> >> >> >> attr->info.clock_rate = 8000; >> >> >> >> >> attr->info.channel_cnt = 1; >> >> >> >> >> attr->info.avg_bps = 8000; >> >> >> >> >> attr->info.max_bps = 8000; >> >> >> >> >> attr->info.pcm_bits_per_sample = 16; >> >> >> >> >> attr->info.frm_ptime = 10; >> >> >> >> >> >> >> >> >> >> 3. In gsm.c there is some hard coding in gsm_codec_parse( >> >> >> >> >> -- what will be the value in replace of "33" here >> >> >> >> >> >> >> >> >> >> 4. frame_size and sample_per_frame are also not clear to me.. >> >> >> >> >> >> >> >> >> >> Sorry for lot of questions and may be those are lot easier to >> >> >> >> >> ask.. >> >> >> >> >> but >> >> >> >> >> really i am not good in codec.. >> >> >> >> >> >> >> >> >> >> waiting for reply. >> >> >> >> >> >> >> >> >> >> Thanks >> >> >> >> >> >> >> >> >> >> S. M. Nazmul Hasan Opu >> >> >> >> >> Software Engineer >> >> >> >> >> R & D Application >> >> >> >> >> Dhaka, Bangladesh >> >> >> >> >> Mob: +880 1712 901 764 >> >> >> >> >> >> >> >> >> >> _______________________________________________ >> >> >> >> >> Visit our blog: http://blog.pjsip.org >> >> >> >> >> >> >> >> >> >> pjsip mailing list >> >> >> >> >> pjsip@lists.pjsip.org >> >> >> >> >> http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> -- >> >> >> >> >> S. M. Nazmul Hasan Opu >> >> >> >> >> Software Engineer >> >> >> >> >> R & D Application >> >> >> >> >> Dhaka, Bangladesh >> >> >> >> >> Mob: +880 1712 901 764 >> >> >> >> >> >> >> >> >> >> _______________________________________________ >> >> >> >> >> Visit our blog: http://blog.pjsip.org >> >> >> >> >> >> >> >> >> >> pjsip mailing list >> >> >> >> >> pjsip@lists.pjsip.org >> >> >> >> >> http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org >> >> >> >> >> >> >> >> >> > >> >> >> >> > >> >> >> >> > >> >> >> >> > -- >> >> >> >> > S. M. Nazmul Hasan Opu >> >> >> >> > Software Engineer >> >> >> >> > R & D Application >> >> >> >> > Dhaka, Bangladesh >> >> >> >> > Mob: +880 1712 901 764 >> >> >> >> > >> >> >> >> > _______________________________________________ >> >> >> >> > Visit our blog: http://blog.pjsip.org >> >> >> >> > >> >> >> >> > pjsip mailing list >> >> >> >> > pjsip@lists.pjsip.org >> >> >> >> > http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org >> >> >> >> > >> >> >> >> > >> >> >> >> >> >> >> >> _______________________________________________ >> >> >> >> Visit our blog: http://blog.pjsip.org >> >> >> >> >> >> >> >> pjsip mailing list >> >> >> >> pjsip@lists.pjsip.org >> >> >> >> http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org >> >> >> > >> >> >> > >> >> >> > >> >> >> > -- >> >> >> > S. M. Nazmul Hasan Opu >> >> >> > Software Engineer >> >> >> > R & D Application >> >> >> > Dhaka, Bangladesh >> >> >> > Mob: +880 1712 901 764 >> >> >> > >> >> >> > _______________________________________________ >> >> >> > Visit our blog: http://blog.pjsip.org >> >> >> > >> >> >> > pjsip mailing list >> >> >> > pjsip@lists.pjsip.org >> >> >> > http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org >> >> >> > >> >> >> > >> >> >> >> >> >> _______________________________________________ >> >> >> Visit our blog: http://blog.pjsip.org >> >> >> >> >> >> pjsip mailing list >> >> >> pjsip@lists.pjsip.org >> >> >> http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org >> >> > >> >> > >> >> > >> >> > -- >> >> > S. M. Nazmul Hasan Opu >> >> > Software Engineer >> >> > R & D Application >> >> > Dhaka, Bangladesh >> >> > Mob: +880 1712 901 764 >> >> > >> >> > _______________________________________________ >> >> > Visit our blog: http://blog.pjsip.org >> >> > >> >> > pjsip mailing list >> >> > pjsip@lists.pjsip.org >> >> > http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org >> >> > >> >> > >> >> >> >> _______________________________________________ >> >> Visit our blog: http://blog.pjsip.org >> >> >> >> pjsip mailing list >> >> pjsip@lists.pjsip.org >> >> http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org >> > >> > >> > >> > -- >> > S. M. Nazmul Hasan Opu >> > Software Engineer >> > R & D Application >> > Dhaka, Bangladesh >> > Mob: +880 1712 901 764 >> > >> > _______________________________________________ >> > Visit our blog: http://blog.pjsip.org >> > >> > pjsip mailing list >> > pjsip@lists.pjsip.org >> > http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org >> > >> > >> >> _______________________________________________ >> Visit our blog: http://blog.pjsip.org >> >> pjsip mailing list >> pjsip@lists.pjsip.org >> http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org > > > > -- > S. M. Nazmul Hasan Opu > Software Engineer > R & D Application > Dhaka, Bangladesh > Mob: +880 1712 901 764 > > _______________________________________________ > Visit our blog: http://blog.pjsip.org > > pjsip mailing list > pjsip@lists.pjsip.org > http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org > >

It is fine for pjsip-ua. I didn't try pjsua api. regards, Gang On Thu, Nov 6, 2008 at 9:49 PM, Rawshan Iajdani <iajdani@provati.com> wrote: > We have different state of call, and one of them is confirmed. I get the > confirmed state when the phone is ringing. But I don't find any state when > the call is answered or connected. Ideas????? > > > _______________________________________________ > Visit our blog: http://blog.pjsip.org > > pjsip mailing list > pjsip@lists.pjsip.org > http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org >

in symbian_ua on_call_state function you get the status PJSIP_INV_STATE_CONFIRMED for connected and PJSIP_INV_STATE_EARLY for ringing. Try it... On Mon, Nov 10, 2008 at 11:00 AM, Gang Liu <gangban.lau@gmail.com> wrote: > It is fine for pjsip-ua. I didn't try pjsua api. > > regards, > Gang > > > On Thu, Nov 6, 2008 at 9:49 PM, Rawshan Iajdani <iajdani@provati.com>wrote: > >> We have different state of call, and one of them is confirmed. I get the >> confirmed state when the phone is ringing. But I don't find any state when >> the call is answered or connected. Ideas????? >> >> >> _______________________________________________ >> Visit our blog: http://blog.pjsip.org >> >> pjsip mailing list >> pjsip@lists.pjsip.org >> http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org >> > > > _______________________________________________ > Visit our blog: http://blog.pjsip.org > > pjsip mailing list > pjsip@lists.pjsip.org > http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org > > -- Salahuddin Ahmed

          PJ_LOG(4,(THIS_FILE, "............writing

         pj_str_t codec_id = pj_str("g729");
         pjmedia_codec_mgr_set_codec_priority(

             &codec_id, PJMEDIA_CODEC_PRIO_HIGHEST);

       Thanks

Hey man.. Just put a break point on your code and try to debug. Why you bother to go to see log file only to see whether a function is called or not. From: pjsip-bounces@lists.pjsip.org [mailto:pjsip-bounces@lists.pjsip.org] On Behalf Of S. M. Nazmul Hasan (Opu) Sent: Monday, November 10, 2008 3:07 PM To: pjsip list Subject: Re: [pjsip] g729 codec Nanag, Can you please tell me how i can debug the various source file of pjmedia or pjsip library. For example : i have tried to debug g729.c as well as gsm.c by using the line PJ_LOG(4,(THIS_FILE, "............writing something........")); this should be shown in the log file ....right? I have put the in pjmedia_codec_g729_init( pjmedia_endpt *endpt ) and pjmedia_codec_gsm_init( pjmedia_endpt *endpt ) function... but no result is shown in the log file. actually this is not the right debugging procedure.. so how can i understand whether any function is being called or not... can you please clarify me.... Regards OPu On Mon, Nov 10, 2008 at 2:09 PM, Nanang Izzuddin <nanang@pjsip.org> wrote: It seems the g729_enum_codecs() is well defined. So please make sure that pjmedia_codec_g729_init() is called, e.g: set PJMEDIA_HAS_G729_CODEC to 1 in your config_site.h. Regards, nanang On Sun, Nov 9, 2008 at 2:11 PM, S. M. Nazmul Hasan (Opu) <apus29@gmail.com> wrote: > Hi Nanang, > > I have tested it to make a self call and getting 488(Not acceptable here) > response. i saw > the log file and the sdp , there speex, gsm , pcma and pcmu only presents . > and no g729. > so it looks like that g729 is not registered. and server rejected the call. > > I have checked the log file of server , there was a message: > > No compatible codecs, not accepting this offer! > > please give me a suggestion. > > Thanks > > opu > > On Thu, Nov 6, 2008 at 4:26 PM, Nanang Izzuddin <nanang@pjsip.org> wrote: >> >> Hi, >> >> I think it would be better if you see the log file in the rejecting >> side (it can be the proxy/asterisk or x-lite) to see why it rejected >> the call. If you have pjsua with IPP enabled, you can just make a call >> to the pjsua and send along the pjsua log file. >> >> The log file in the Symbian side may help, but I am not sure it will >> help a lot, however to generate the log file on Symbian, just specify >> log filename in logging config for the pjsua_init(), e.g: >> log_cfg.log_filename = pj_str("c:\\data\\pjsua.log"); >> >> Regards, >> nanang >> >> >> On Thu, Nov 6, 2008 at 3:02 PM, S. M. Nazmul Hasan (Opu) >> <apus29@gmail.com> wrote: >> > Hi Nanag, >> > >> > Yes the other endpoint supports g729. my tested environment was: >> > >> > server: AsteriskNow (which supports g729) >> > client: pjsip (symbian_ua_gui) and Xlite Pro(which supports g729) >> > >> > 1. when i am using both endpoint as Xlite pro and fixing the codec as >> > g729 >> > only then both using g729 and call established fine. >> > >> > 2. But when i am trying with symbian_ua_gui with Xlite Pro fixing the >> > codec >> > to g729 then i am getting 488(Not acceptable here) response. >> > >> > Can you please tell me how can I get the log file for No. 2 test >> > criteria. >> > >> > Thanks >> > >> > Opu >> > >> > On Wed, Nov 5, 2008 at 5:00 PM, Nanang Izzuddin <nanang@pjsip.org> >> > wrote: >> >> >> >> Hi, >> >> >> >> Just in case, please make sure the other endpoint supports G729 too, >> >> e.g: make a call to PJSUA on desktop with IPP G729 enabled. And please >> >> check the pjsua log, see if G729 is in the INVITE SDP. If it is >> >> rejected, you may see the reason in the pjsua log. >> >> >> >> Regards, >> >> nanang >> >> >> >> >> >> On Wed, Nov 5, 2008 at 1:39 PM, S. M. Nazmul Hasan (Opu) >> >> <apus29@gmail.com> wrote: >> >> > Nanag, >> >> > >> >> > I think so Nanang, may be its not registering because i am getting >> >> > response >> >> > 488(Not acceptable here). But i did it register and unregister in >> >> > pjsua_media.c. >> >> > >> >> > Can you please see these files. >> >> > >> >> > Thanks >> >> > >> >> > Opu >> >> > >> >> > On Tue, Nov 4, 2008 at 10:18 PM, Nanang Izzuddin <nanang@pjsip.org> >> >> > wrote: >> >> >> >> >> >> Hi, >> >> >> >> >> >> Just did a quick skimming, things I can found so far: >> >> >> 1. For 8000kbps and 10ms frame time, the encoded frame size should >> >> >> be >> >> >> 10 bytes, not 20 bytes. >> >> >> 2. this line: >> >> >> pcm_in += 160; >> >> >> should be: >> >> >> pcm_in += 80; >> >> >> >> >> >> Numbers in a codec wrapper are very important, so please check and >> >> >> recheck carefully. >> >> >> >> >> >> Regards, >> >> >> nanang >> >> >> >> >> >> >> >> >> On Tue, Nov 4, 2008 at 8:42 PM, S. M. Nazmul Hasan (Opu) >> >> >> <apus29@gmail.com> wrote: >> >> >> > Dear Rawshan, >> >> >> > >> >> >> > Can you please check the file if it is ok to add voiceage g729 in >> >> >> > pjsip >> >> >> > for >> >> >> > symbian. i am trying this for several days but i can get it >> >> >> > working >> >> >> > for >> >> >> > me. >> >> >> > >> >> >> > Help from anyone would be great appretiable. >> >> >> > >> >> >> > Thanks >> >> >> > >> >> >> > On Mon, Nov 3, 2008 at 3:17 PM, Rawshan Iajdani >> >> >> > <iajdani@provati.com> >> >> >> > wrote: >> >> >> >> >> >> >> >> You should be able to do that in pjlib/include/pj/config_site.h >> >> >> >> and >> >> >> >> pjmedia/include/pjmedia-codec/config.h file. But always remember >> >> >> >> config_site.h settings will overwrite any settings done in >> >> >> >> config.h >> >> >> >> file. >> >> >> >> Good luck. >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> From: pjsip-bounces@lists.pjsip.org >> >> >> >> [mailto:pjsip-bounces@lists.pjsip.org] >> >> >> >> On Behalf Of S. M. Nazmul Hasan (Opu) >> >> >> >> Sent: Saturday, November 01, 2008 6:01 PM >> >> >> >> To: pjsip list >> >> >> >> Subject: Re: [pjsip] g729 codec >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> Thanks Lajdani. My jumped start is running well. >> >> >> >> >> >> >> >> I am using Brekeke sip server to check the active session. and i >> >> >> >> am >> >> >> >> getting the using codec (payload) is always PCMU/8000. even >> >> >> >> after >> >> >> >> changing >> >> >> >> the g729 for highest priority. >> >> >> >> >> >> >> >> pj_str_t codec_id = pj_str("g729"); >> >> >> >> pjmedia_codec_mgr_set_codec_priority( >> >> >> >> pjmedia_endpt_get_codec_mgr(pjsua_var.med_endpt), >> >> >> >> &codec_id, PJMEDIA_CODEC_PRIO_HIGHEST); >> >> >> >> >> >> >> >> when i changed it to PJMEDIA_CODEC_PRIO_LOWEST the result was >> >> >> >> same. >> >> >> >> even >> >> >> >> after disabling all the codecs and after deleting all g711 files >> >> >> >> and >> >> >> >> related >> >> >> >> fields it showed the payload is PCMU/8000. >> >> >> >> >> >> >> >> How can i easily disable all the codecs except g729. >> >> >> >> >> >> >> >> Thanks >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> On Thu, Oct 30, 2008 at 3:43 AM, Rawshan Iajdani >> >> >> >> <iajdani@provati.com> >> >> >> >> wrote: >> >> >> >> >> >> >> >> Well.. your attribute settings seems fine. For the 33 it will be >> >> >> >> 20. >> >> >> >> And >> >> >> >> in place of 320, it will be 160.. hope this will give u a jump >> >> >> >> start.. >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> Rawshan Iajdani >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> From: pjsip-bounces@lists.pjsip.org >> >> >> >> [mailto:pjsip-bounces@lists.pjsip.org] >> >> >> >> On Behalf Of S. M. Nazmul Hasan (Opu) >> >> >> >> Sent: Wednesday, October 29, 2008 6:12 PM >> >> >> >> To: pjsip list >> >> >> >> Subject: [pjsip] g729 codec >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> Hi Benny, >> >> >> >> >> >> >> >> I am trying to add voice-age g729 codec with PJSIP for symbian. >> >> >> >> but >> >> >> >> i >> >> >> >> am >> >> >> >> getting some difficulties while following the gsm.c file. >> >> >> >> >> >> >> >> 1. while creating g729 codec private data >> >> >> >> what should i declare in replace of >> >> >> >> // struct gsm_state *encoder; >> >> >> >> // struct gsm_state *decoder; >> >> >> >> >> >> >> >> in "l16.c" its only use the frame size .. >> >> >> >> >> >> >> >> 2. in Generate default attribute. >> >> >> >> >> >> >> >> i have changed the value to this. is it ok? >> >> >> >> attr->info.clock_rate = 8000; >> >> >> >> attr->info.channel_cnt = 1; >> >> >> >> attr->info.avg_bps = 8000; >> >> >> >> attr->info.max_bps = 8000; >> >> >> >> attr->info.pcm_bits_per_sample = 16; >> >> >> >> attr->info.frm_ptime = 10; >> >> >> >> >> >> >> >> 3. In gsm.c there is some hard coding in gsm_codec_parse( >> >> >> >> -- what will be the value in replace of "33" here >> >> >> >> >> >> >> >> 4. frame_size and sample_per_frame are also not clear to me.. >> >> >> >> >> >> >> >> Sorry for lot of questions and may be those are lot easier to >> >> >> >> ask.. >> >> >> >> but >> >> >> >> really i am not good in codec.. >> >> >> >> >> >> >> >> waiting for reply. >> >> >> >> >> >> >> >> Thanks >> >> >> >> >> >> >> >> S. M. Nazmul Hasan Opu >> >> >> >> Software Engineer >> >> >> >> R & D Application >> >> >> >> Dhaka, Bangladesh >> >> >> >> Mob: +880 1712 901 764 >> >> >> >> >> >> >> >> _______________________________________________ >> >> >> >> Visit our blog: http://blog.pjsip.org >> >> >> >> >> >> >> >> pjsip mailing list >> >> >> >> pjsip@lists.pjsip.org >> >> >> >> http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org >> >> >> >> >> >> >> >> >> >> >> >> -- >> >> >> >> S. M. Nazmul Hasan Opu >> >> >> >> Software Engineer >> >> >> >> R & D Application >> >> >> >> Dhaka, Bangladesh >> >> >> >> Mob: +880 1712 901 764 >> >> >> >> >> >> >> >> _______________________________________________ >> >> >> >> Visit our blog: http://blog.pjsip.org >> >> >> >> >> >> >> >> pjsip mailing list >> >> >> >> pjsip@lists.pjsip.org >> >> >> >> http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org >> >> >> >> >> >> >> > >> >> >> > >> >> >> > >> >> >> > -- >> >> >> > S. M. Nazmul Hasan Opu >> >> >> > Software Engineer >> >> >> > R & D Application >> >> >> > Dhaka, Bangladesh >> >> >> > Mob: +880 1712 901 764 >> >> >> > >> >> >> > _______________________________________________ >> >> >> > Visit our blog: http://blog.pjsip.org >> >> >> > >> >> >> > pjsip mailing list >> >> >> > pjsip@lists.pjsip.org >> >> >> > http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org >> >> >> > >> >> >> > >> >> >> >> >> >> _______________________________________________ >> >> >> Visit our blog: http://blog.pjsip.org >> >> >> >> >> >> pjsip mailing list >> >> >> pjsip@lists.pjsip.org >> >> >> http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org >> >> > >> >> > >> >> > >> >> > -- >> >> > S. M. Nazmul Hasan Opu >> >> > Software Engineer >> >> > R & D Application >> >> > Dhaka, Bangladesh >> >> > Mob: +880 1712 901 764 >> >> > >> >> > _______________________________________________ >> >> > Visit our blog: http://blog.pjsip.org >> >> > >> >> > pjsip mailing list >> >> > pjsip@lists.pjsip.org >> >> > http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org >> >> > >> >> > >> >> >> >> _______________________________________________ >> >> Visit our blog: http://blog.pjsip.org >> >> >> >> pjsip mailing list >> >> pjsip@lists.pjsip.org >> >> http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org >> > >> > >> > >> > -- >> > S. M. Nazmul Hasan Opu >> > Software Engineer >> > R & D Application >> > Dhaka, Bangladesh >> > Mob: +880 1712 901 764 >> > >> > _______________________________________________ >> > Visit our blog: http://blog.pjsip.org >> > >> > pjsip mailing list >> > pjsip@lists.pjsip.org >> > http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org >> > >> > >> >> _______________________________________________ >> Visit our blog: http://blog.pjsip.org >> >> pjsip mailing list >> pjsip@lists.pjsip.org >> http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org > > > > -- > S. M. Nazmul Hasan Opu > Software Engineer > R & D Application > Dhaka, Bangladesh > Mob: +880 1712 901 764 > > _______________________________________________ > Visit our blog: http://blog.pjsip.org > > pjsip mailing list > pjsip@lists.pjsip.org > http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org > > _______________________________________________ Visit our blog: http://blog.pjsip.org pjsip mailing list pjsip@lists.pjsip.org http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org -- S. M. Nazmul Hasan Opu Software Engineer R & D Application Dhaka, Bangladesh Mob: +880 1712 901 764