Hi Sergio I am using app_rtsp, app_transcoder,linphone and vlc .
My dialplan is [default] exten => 101,1,Answer exten => 101,2,transcode(,s...@camera,h...@qcif /fps=12/kb=52/qmin=4/qmax=12/gs=50) [camera] exten => s,1,Answer exten => s,2,rtsp(rtsp://192.168.1.3:1234/stream.sdp) exten => s,3,Hangup I have opened the rtsp stream using vlc. I can play the stream using vlc client. I am dialing 101 from linphone and app_rtsp properly connects to vlc server. I can see the describe, setup, play messages in ethereal. Now the problem is, app_transcoder is unable to decode the incoming h264 or mpeg4 stream from vlc server. I have gone through the traces; it says app_transcoder is able to find the decoder; but it fails while decoding the frames. app_transcoder is throwing the message while exceuting avcodec_decode_video(); In case of h264; it is "no frame" In case of mpeg4 ; it is "invalid picture size". I am feeding transcoded frames using vlc server. i.e the picture captured from camera is transcoded in h264 or mpeg4. I have even tried with mpg files. but same result. Is It related to the libavcodec library i am using? But the library is able to open the decoders. Regards Anand 2009/11/3 Sergio Garcia Murillo <[email protected]> > 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(,1...@default,h...@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 <[email protected]> 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 <[email protected]> >> >> 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 <[email protected]> >>> 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(,1...@default,h...@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 <[email protected]> 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 <[email protected]> >>>>> >>>>>> 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-- >>>> >>>> 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-- >>>> >>>> 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-- >>> >>> 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-- >>> >>> 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 > > > > _______________________________________________ > --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 > -- Anandadip Mandal
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