NI
Nanang Izzuddin
Mon, Nov 10, 2008 9:43 AM
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)
-
when i am using both endpoint as Xlite pro and fixing the codec as
g729
only then both using g729 and call established fine.
-
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:
- For 8000kbps and 10ms frame time, the encoded frame size
should
be
10 bytes, not 20 bytes.
- 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.
- 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 ..
- 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;
-
In gsm.c there is some hard coding in gsm_codec_parse(
-- what will be the value in replace of "33" here
-
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
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
>
>
GL
Gang Liu
Mon, Nov 10, 2008 10:00 AM
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:
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
>
SA
Salahuddin Ahmed
Mon, Nov 10, 2008 10:26 AM
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.comwrote:
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
RI
Rawshan Iajdani
Wed, Nov 12, 2008 10:25 PM
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
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)
-
when i am using both endpoint as Xlite pro and fixing the codec as
g729
only then both using g729 and call established fine.
-
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:
- For 8000kbps and 10ms frame time, the encoded frame size should
be
10 bytes, not 20 bytes.
- 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.
- 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 ..
- 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;
-
In gsm.c there is some hard coding in gsm_codec_parse(
-- what will be the value in replace of "33" here
-
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
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
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> S. M. Nazmul Hasan Opu
> Software Engineer
> R & D Application
> Dhaka, Bangladesh
> Mob: +880 1712 901 764
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--
S. M. Nazmul Hasan Opu
Software Engineer
R & D Application
Dhaka, Bangladesh
Mob: +880 1712 901 764