<div>Hi Sergio</div>
<div> </div>
<div>I am using app_rtsp, app_transcoder,linphone and vlc .</div>
<div> </div>
<div>My dialplan is </div>
<div>[default]</div>
<div>exten => 101,1,Answer<br>exten => 101,2,transcode(,s@camera,h263@qcif/fps=12/kb=52/qmin=4/qmax=12/gs=50)<br></div>
<div>[camera]</div>
<div>exten => s,1,Answer<br>exten => s,2,rtsp(rtsp://<a href="http://192.168.1.3:1234/stream.sdp">192.168.1.3:1234/stream.sdp</a>)</div>
<div>exten => s,3,Hangup</div>
<div> </div>
<div> </div>
<div>I have opened the rtsp stream using vlc.</div>
<div>I can play the stream using vlc client.</div>
<div> </div>
<div>I am dialing 101 from linphone and app_rtsp properly connects to vlc server.</div>
<div>I can see the describe, setup, play messages in ethereal.</div>
<div>Now the problem is, app_transcoder is unable to decode the incoming h264 or mpeg4 stream from vlc server.</div>
<div>I have gone through the traces; it says app_transcoder is able to find the decoder; but it fails while decoding the frames.</div>
<div>app_transcoder is throwing the message while exceuting avcodec_decode_video();</div>
<div>In case of h264; it is "no frame"</div>
<div>In case of mpeg4 ; it is "invalid picture size".</div>
<div> </div>
<div>I am feeding transcoded frames using vlc server.</div>
<div>i.e the picture captured from camera is transcoded in h264 or mpeg4.</div>
<div>I have even tried with mpg files. but same result.</div>
<div> </div>
<div>Is It related to the libavcodec library i am using?</div>
<div>But the library is able to open the decoders.</div>
<div> </div>
<div>Regards</div>
<div>Anand</div>
<div> </div>
<div> </div>
<div> </div>
<div><br><br> </div>
<div class="gmail_quote">2009/11/3 Sergio Garcia Murillo <span dir="ltr"><<a href="mailto:sergio.garcia@fontventa.com">sergio.garcia@fontventa.com</a>></span><br>
<blockquote style="BORDER-LEFT: #ccc 1px solid; MARGIN: 0px 0px 0px 0.8ex; PADDING-LEFT: 1ex" class="gmail_quote">
<div bgcolor="#ffffff" text="#000000">Hi anand<br><br>As I said before, app transcoder can only currently encode in h263p, so you are not going to be able to do it. <br>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.
<div>
<div></div>
<div class="h5"><br><br>Best regards<br>Sergio<br><br>anandadip mandal escribió:
<blockquote type="cite">
<div>Hi sergio</div>
<div> </div>
<div>I am not sure if i am using correct dialplan.</div>
<div>I want to transcode between two sip phone ; one is using mpeg4 and the other one h263p.</div>
<div>my dialplan:</div>
<div>
<div>[default]<br>exten => 101,1,Answer<br>exten => 101,2,transcode(,102@default,h263@qcif/fps=12/kb=52/qmin=4/qmax=12/gs=50)<br>exten => 102,1,Dial(SIP/101)</div>
<div> </div>
<div>102(mpeg4) is calling 101(h263p).</div>
<div> </div>
<div>Do i need to use any other module say app_rtsp?</div>
<div>Please suggest the correct dialplan.</div>
<div> </div>
<div> </div>
<div>Regards</div>
<div>Anand</div></div>
<div><br><br> </div>
<div><span class="gmail_quote">On 03/11/2009, <b class="gmail_sendername">anandadip mandal</b> <<a href="mailto:anandadip@gmail.com" target="_blank">anandadip@gmail.com</a>> wrote:</span>
<blockquote style="BORDER-LEFT: rgb(204,204,204) 1px solid; MARGIN: 0px 0px 0px 0.8ex; PADDING-LEFT: 1ex" class="gmail_quote">Hi Sergio<br>Thanks for the reply. app transcoder only supports h263p. I have a small doubt; please correct me if I am wrong.<br>
Consider the following use case:<br><br>Xlite is configured with h263-1996<br>Linphone is configured with h263p.<br>Xlite is placing call to linphone.<br>So ; the codec between xlite and asterisk is h263-1996; and between asterisk and linphone is h263p.<br>
App transcoder will convert incoming h263-1996 packets into h263p.So i can expect xlite will be able to send video to linphone.<br>Now my confusion is :<br>Will app transcoder also convert incoming h263 packets from linphone to h263-1996?<br>
Othewise it is not possible to send video from linphone to xlite.<br><br>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.<br>
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.<br>
<br>By the way ; what are the codecs are supported by libavcodec and asterisk?<br>I am interested in :<br>h261<br>h263<br>h263p<br>h264<br>mpeg-4<br><br>Thanks and regards<br>Anand <br><br><br>
<div class="gmail_quote">2009/11/3 Sergio Garcia Murillo <span dir="ltr"><<a href="mailto:sergio.garcia@fontventa.com" target="_blank">sergio.garcia@fontventa.com</a>></span>
<div><span><br>
<blockquote style="BORDER-LEFT: rgb(204,204,204) 1px solid; MARGIN: 0pt 0pt 0pt 0.8ex; PADDING-LEFT: 1ex" class="gmail_quote">
<div bgcolor="#ffffff" text="#000000">Hi anandapip,<br><br>app_transcoder only supports encoding in h263-1998/2000 (h263p), not in h263-1996.
<div>
<div><br><br>Best regards<br>Sergio<br><br>anandadip mandal escribió:
<blockquote type="cite">
<div>Hi Sergio</div>
<div>Thanks for the reply.</div>
<div>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.</div>
<div>But a strange problem I am facing now.</div>
<div>I have tried transcoding between h263 and h263+.I have used Xlite and linphone.</div>
<div>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.</div>
<div>I am not sure if app_transcode module is really encoding in h263 format thogh log says it is encoding.</div>
<div> </div>
<div>Thanks and regards</div>
<div>Anand</div>
<div><br><br> </div>
<div><span class="gmail_quote">On 02/11/2009, <b class="gmail_sendername">Sergio Garcia Murillo</b> <<a href="mailto:sergio.garcia@fontventa.com" target="_blank">sergio.garcia@fontventa.com</a>> wrote:</span>
<blockquote style="BORDER-LEFT: rgb(204,204,204) 1px solid; MARGIN: 0px 0px 0px 0.8ex; PADDING-LEFT: 1ex" class="gmail_quote">
<div bgcolor="#ffffff" text="#000000">Hi anandadip<br><br>Get the core dump and a back trace of asterisk when it seg faults<br><br>Best regards<br>Sergio<br><br>anandadip mandal escribió:
<blockquote type="cite">
<div><span>
<div>Hi</div>
<div>I want to make video call between two sip phone having different video codecs using app_transcoder.</div>
<div>I have used the following dialplan</div>
<div>[default]<br>exten => 101,1,Answer<br>exten => 101,2,transcode(,102@default,h263@qcif/fps=12/kb=52/qmin=4/qmax=12/gs=50)<br>exten => 102,1,Dial(SIP/101)</div>
<div> </div>
<div>the 102 ( having h263-1998 codec) extension is calling 101 (having h263 codec).</div>
<div>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.</div>
<div>Without app transcoder, video call works fine when both phone use h263-1998 codec.</div>
<div>I am using asterisk 1.4; the transcode module loads succesfully; even it executes and places a call to the configured extension)</div>
<div> </div>
<div>Please help me if i am using the correct dialplan or am i missing something.</div>
<div> </div>
<div>Any help will be much appreciated.</div>
<div> </div>
<div>Regards</div>
<div>Anand</div>
<div><br><br> </div>
<div><span class="gmail_quote">On 26/10/2009, <b class="gmail_sendername">anandadip mandal</b> <<a href="mailto:anandadip@gmail.com" target="_blank">anandadip@gmail.com</a>> wrote:</span>
<blockquote style="BORDER-LEFT: rgb(204,204,204) 1px solid; MARGIN: 0px 0px 0px 0.8ex; PADDING-LEFT: 1ex" class="gmail_quote">
<div>Hi </div>
<div>I have successfully compiled and able to load the app_transcoder.so;</div>
<div>I want to know the configuration of extension.conf to put the app_transcoder in use.</div>
<div>I have two sip soft phone(video capable) 3000, 3001 which are already registered to asterisk and I can make audio call between them;</div>
<div>Also please let me know if i have to add anything specific to extesion.conf and sip.conf for enabling video call.</div>
<div>Any help will be very much appreciated.</div>
<div>Thanks and regards</div>
<div>Anand</div>
<div><br> </div>
<div class="gmail_quote">2009/10/20 anandadip mandal <span dir="ltr"><<a href="mailto:anandadip@gmail.com" target="_blank">anandadip@gmail.com</a>></span><span><br>
<blockquote style="BORDER-LEFT: rgb(204,204,204) 1px solid; MARGIN: 0px 0px 0px 0.8ex; PADDING-LEFT: 1ex" class="gmail_quote">
<div>is there any document for compilation procedure of app transcoder?also could someone point me how to integrate it with asterisk?</div>
<div>Thanks</div>
<div>Anand<br clear="all"></div><br></blockquote></span></div><br><br clear="all"><br>-- <br><span>Anandadip Mandal<br></span></blockquote></div><br><br clear="all"><br>-- <br>Anandadip Mandal </span></div><pre><hr size="4" width="90%">
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</blockquote></div><br><br clear="all"><br>-- <br>Anandadip Mandal <pre><hr size="4" width="90%">
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</blockquote></span></div></div><br><br clear="all"><br>-- <br><span>Anandadip Mandal<br></span></blockquote></div><br><br clear="all"><br>-- <br>Anandadip Mandal <pre><hr size="4" width="90%">
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</blockquote></div><br><br clear="all"><br>-- <br>Anandadip Mandal<br>