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
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