[Asterisk-video] app transcoder

Sergio Garcia Murillo sergio.garcia at fontventa.com
Tue Nov 3 10:12:48 CST 2009


Hi anand

As I said before, app transcoder can only currently encode in h263p, so 
you are not going to be able to do it.
The application main pourpose was as a complment to the h324m library 
(to adjust the video bitrate from the videophone) and to use the video 
from a network camera in asterisk with app_rtsp.

Best regards
Sergio

anandadip mandal escribió:
> Hi sergio
>  
> I am not sure if i am using correct dialplan.
> I want to transcode between two sip phone ; one is using mpeg4 and the 
> other one h263p.
> my 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)
>  
> 102(mpeg4)  is calling 101(h263p).
>  
> Do i need to use any other module say app_rtsp?
> Please suggest the correct dialplan.
>  
>  
> Regards
> Anand
>
>
>  
> On 03/11/2009, *anandadip mandal* <anandadip at gmail.com 
> <mailto:anandadip at gmail.com>> wrote:
>
>     Hi Sergio
>     Thanks for the reply. app transcoder only supports h263p. I have a
>     small doubt; please correct me if I am wrong.
>     Consider the following use case:
>
>     Xlite is configured with h263-1996
>     Linphone is configured with h263p.
>     Xlite is placing call to linphone.
>     So ; the codec between xlite and asterisk is h263-1996; and
>     between asterisk and linphone is h263p.
>     App transcoder will convert incoming h263-1996 packets into
>     h263p.So i can expect xlite will be able to send video to linphone.
>     Now my confusion is :
>     Will app transcoder also convert incoming h263 packets from
>     linphone to h263-1996?
>     Othewise it is not possible to send video from linphone to xlite.
>
>     Since app transcoder supports h263p; if i keep codecs in both the
>     phones h263p; video will appear in both the phone. But then. i do
>     not really need app transcoder; asterisk is capable of doing it
>     without app transcoder.
>     It seems app_transcoder only supports oneway video; Because if we
>     use transcoding between h263p and other codecs ( say
>     mpeg/h263/h261); app_transcoder will be able to encode other
>     codecs to h263p but it will not be able to do the opposite; and we
>     will only see one way video.
>
>     By the way ; what are the codecs are supported by libavcodec and
>     asterisk?
>     I am interested in :
>     h261
>     h263
>     h263p
>     h264
>     mpeg-4
>
>     Thanks and regards
>     Anand
>
>
>     2009/11/3 Sergio Garcia Murillo <sergio.garcia at fontventa.com
>     <mailto:sergio.garcia at fontventa.com>>
>
>         Hi anandapip,
>
>         app_transcoder only supports encoding in h263-1998/2000
>         (h263p), not in h263-1996.
>
>
>         Best regards
>         Sergio
>
>         anandadip mandal escribió:
>>         Hi Sergio
>>         Thanks for the reply.
>>         There was a problem in my ffmpeg (livavcodec) which was not
>>         buit with videocodec support.I have replaced it and now not
>>         getting the error.
>>         But a strange problem I am facing now.
>>         I have tried transcoding between h263 and h263+.I have used
>>         Xlite and linphone.
>>         I am calling from linphone which is using h263-1998 codec;
>>         App transcoder encodes the incoming h263-1998 to h263 and
>>         places call to xlite. It is also evident from the sip
>>         signalling traces that codec between asterisk and linphone is
>>         h263-1998 and between asterisk and xlite is h263.But if i
>>         configure xlite only for h263 ; no video is apperaing. But if
>>         i keep codec in xlite h263-1998 (i.e h263+) video appears.
>>         I am not sure if app_transcode module is really encoding in
>>         h263 format thogh log says it is encoding.
>>          
>>         Thanks and regards
>>         Anand
>>
>>
>>          
>>         On 02/11/2009, *Sergio Garcia Murillo*
>>         <sergio.garcia at fontventa.com
>>         <mailto:sergio.garcia at fontventa.com>> wrote:
>>
>>             Hi anandadip
>>
>>             Get the core dump and a back trace of asterisk when it
>>             seg faults
>>
>>             Best regards
>>             Sergio
>>
>>             anandadip mandal escribió:
>>>             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
>>>             ------------------------------------------------------------------------
>>>
>>>
>>>
>>>             _______________________________________________
>>>             --Bandwidth and Colocation Provided by http://www.api-digital.com <http://www.api-digital.com/>--
>>>
>>>             asterisk-video mailing list
>>>             To UNSUBSCRIBE or update options visit:
>>>                http://lists.digium.com/mailman/listinfo/asterisk-video
>>
>>              
>>
>>             _______________________________________________
>>             --Bandwidth and Colocation Provided by
>>             http://www.api-digital.com-- <http://www.api-digital.com--/>
>>
>>             asterisk-video mailing list
>>             To UNSUBSCRIBE or update options visit:
>>               http://lists.digium.com/mailman/listinfo/asterisk-video
>>
>>
>>
>>
>>         -- 
>>         Anandadip Mandal
>>         ------------------------------------------------------------------------
>>
>>         _______________________________________________
>>         --Bandwidth and Colocation Provided by http://www.api-digital.com <http://www.api-digital.com/>--
>>
>>         asterisk-video mailing list
>>         To UNSUBSCRIBE or update options visit:
>>            http://lists.digium.com/mailman/listinfo/asterisk-video
>
>          
>
>         _______________________________________________
>         --Bandwidth and Colocation Provided by
>         http://www.api-digital.com-- <http://www.api-digital.com--/>
>
>         asterisk-video mailing list
>         To UNSUBSCRIBE or update options visit:
>           http://lists.digium.com/mailman/listinfo/asterisk-video
>
>
>
>
>     -- 
>     Anandadip Mandal
>
>
>
>
> -- 
> Anandadip Mandal
> ------------------------------------------------------------------------
>
> _______________________________________________
> --Bandwidth and Colocation Provided by http://www.api-digital.com--
>
> asterisk-video mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-video

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-video/attachments/20091103/5e285635/attachment.htm 


More information about the asterisk-video mailing list