[Asterisk-video] app transcoder

Mon Nov 2 09:50:33 CST 2009

Hi Anandadip,

Make sure your libavcodec is compiled with h263p support first then.

I think it was: ffmpeg -formats

Best regards
Sergio

anandadip mandal escribió:
> Hi
> I went through the log and it says "error opening encoder". I am using 
> the following libraries
>     >>ldd app_transcoder.so
>     linux-gate.so.1 =>  (0xb8091000)
>     libavcodec.so.52 => /usr/lib/i686/cmov/libavcodec.so.52 (0xb78a6000)
>     libswscale.so.0 => /usr/lib/i686/cmov/libswscale.so.0 (0xb7876000)
>     libc.so.6 => /lib/tls/i686/cmov/libc.so.6 (0xb7712000)
>     libavutil.so.49 => /usr/lib/i686/cmov/libavutil.so.49 (0xb7700000)
>     libz.so.1 => /lib/libz.so.1 (0xb76ea000)
>     libm.so.6 => /lib/tls/i686/cmov/libm.so.6 (0xb76c4000)
>     libgsm.so.1 => /usr/lib/libgsm.so.1 (0xb76b7000)
>     libschroedinger-1.0.so.0 => /usr/lib/libschroedinger-1.0.so.0 
> (0xb7646000)
>     libpthread.so.0 => /lib/tls/i686/cmov/libpthread.so.0 (0xb762d000)
>     libspeex.so.1 => /usr/lib/sse2/libspeex.so.1 (0xb7610000)
>     libtheora.so.0 => /usr/lib/libtheora.so.0 (0xb75bf000)
>     libvorbisenc.so.2 => /usr/lib/libvorbisenc.so.2 (0xb74c5000)
>     libvorbis.so.0 => /usr/lib/libvorbis.so.0 (0xb749b000)
>     /lib/ld-linux.so.2 (0xb8092000)
>     liboil-0.3.so.0 => /usr/lib/liboil-0.3.so.0 (0xb742b000)
>     libogg.so.0 => /usr/lib/libogg.so.0 (0xb7425000)
>     librt.so.1 => /lib/tls/i686/cmov/librt.so.1 (0xb741c000)
>
>  
>
> [Nov  2 02:39:59] WARNING[27345] app_transcoder.c: >Transcoding 
> [,102 at default,h2
> 63 at qcif/fps=10/kb=52/qmin=4/qmax=12/gs=50 
> <mailto:63 at qcif/fps=10/kb=52/qmin=4/qmax=12/gs=50>]
> [Nov  2 02:39:59] WARNING[27345] app_transcoder.c: >anand-Transcoding 
> [,102 at defa
> ult,h263 at qcif/fps=10/kb=52/qmin=4/qmax=12/gs=50]
> [Nov  2 02:39:59] WARNING[27345] app_transcoder.c: -Transcoder 
> [f=0,fps=10,kb=53
> 248,qmin=4,qmax=12,gs=50]
> [[Nov  2 02:39:59] ERROR[27345] app_transcoder.c: Error opening encoder
> [Nov  2 02:39:59] WARNING[27345] app_transcoder.c: -joining thread
> [Nov  2 02:39:59] WARNING[27345] app_transcoder.c: -joined thread
> [Nov  2 02:40:38] WARNING[27345] app_transcoder.c: -end loop[Nov  2 
> 02:40:38] WA
> RNING[27345] app_transcoder.c: -Hanging up
> [Nov  2 02:40:38] WARNING[27345] app_transcoder.c: <Transcoding
> [Nov  2 02:40:58] WARNING[27324] chan_sip.c: Maximum retries exceeded 
> on transmi
> ssion 1f511c6a092c41572f06f0c0353bdbc9 at 192.168.1.3 
> <mailto:1f511c6a092c41572f06f0c0353bdbc9 at 192.168.1.3> for seqno 103 
> (Non-critical R
> equest)
>  
> Please help me out.
> Thanks
> Anand
>
>
>  
> On 02/11/2009, *anandadip mandal* <anandadip at gmail.com 
> <mailto:anandadip at gmail.com>> wrote:
>
>     Hi
>     I want to make video call between two sip phone having different
>     video codecs using app_transcoder.
>     I have used the following dialplan
>     [default]
>     exten => 101,1,Answer
>     exten =>
>     101,2,transcode(,102 at default,h263 at qcif/fps=12/kb=52/qmin=4/qmax=12/gs=50)
>     exten => 102,1,Dial(SIP/101)
>      
>     the 102 ( having h263-1998 codec) extension is calling 101 (having
>     h263 codec).
>     I can see the call between the two phone established but no video;
>     also i dont see any ack coming from 101 and within seconds
>     asterisk gives a segfault.
>     Without app transcoder, video call works fine when both phone use
>     h263-1998 codec.
>     I am using asterisk 1.4; the transcode module loads succesfully;
>     even it executes and places a call to the configured extension)
>      
>     Please help me if i am using the correct dialplan or am i missing
>     something.
>      
>     Any help will be much appreciated.
>      
>     Regards
>     Anand
>
>
>      
>     On 26/10/2009, *anandadip mandal* <anandadip at gmail.com
>     <mailto:anandadip at gmail.com>> wrote:
>
>         Hi
>         I have successfully compiled and able to load the
>         app_transcoder.so;
>         I want to know the configuration of  extension.conf to put the
>         app_transcoder in use.
>         I have two sip soft phone(video capable) 3000, 3001 which are
>         already registered to asterisk and I can make audio call 
>         between them;
>         Also please let me know if i have to add anything specific to
>         extesion.conf and sip.conf  for enabling  video call.
>         Any help will be very much appreciated.
>         Thanks and regards
>         Anand
>
>          
>         2009/10/20 anandadip mandal <anandadip at gmail.com
>         <mailto:anandadip at gmail.com>>
>
>             is there any document for compilation procedure of app
>             transcoder?also could someone point me how to integrate it
>             with asterisk?
>             Thanks
>             Anand
>
>
>
>
>         -- 
>         Anandadip Mandal
>
>
>
>
>     -- 
>     Anandadip Mandal 
>
>
>
>
> -- 
> Anandadip Mandal
> ------------------------------------------------------------------------
>
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