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">sergio.garcia@fontventa.com</a>></span><br>
<blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
<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></div><div class="h5"><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 class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0px 0px 0px 0.8ex; padding-left: 1ex;">
<div text="#000000" bgcolor="#ffffff">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 class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0px 0px 0px 0.8ex; padding-left: 1ex;">
<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 class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0px 0px 0px 0.8ex; padding-left: 1ex;">
<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%">
_______________________________________________
--Bandwidth and Colocation Provided by <a href="http://www.api-digital.com/" target="_blank">http://www.api-digital.com</a>--
asterisk-video mailing list
To UNSUBSCRIBE or update options visit:
<a href="http://lists.digium.com/mailman/listinfo/asterisk-video" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-video</a></pre>
</blockquote>
<br>
</div>
<br>
_______________________________________________<br>
--Bandwidth and Colocation Provided by <a href="http://www.api-digital.com--/" target="_blank">http://www.api-digital.com--</a><br>
<br>
asterisk-video mailing list<br>
To UNSUBSCRIBE or update options visit:<br>
<a href="http://lists.digium.com/mailman/listinfo/asterisk-video" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-video</a><br>
</blockquote>
</div>
<br>
<br clear="all">
<br>
-- <br>
Anandadip Mandal
<pre><hr size="4" width="90%">
_______________________________________________
--Bandwidth and Colocation Provided by <a href="http://www.api-digital.com" target="_blank">http://www.api-digital.com</a>--
asterisk-video mailing list
To UNSUBSCRIBE or update options visit:
<a href="http://lists.digium.com/mailman/listinfo/asterisk-video" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-video</a></pre>
</blockquote>
<br>
</div></div></div>
<br>_______________________________________________<br>
--Bandwidth and Colocation Provided by <a href="http://www.api-digital.com--" target="_blank">http://www.api-digital.com--</a><br>
<br>
asterisk-video mailing list<br>
To UNSUBSCRIBE or update options visit:<br>
<a href="http://lists.digium.com/mailman/listinfo/asterisk-video" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-video</a><br></blockquote></div><br><br clear="all"><br>-- <br>Anandadip Mandal<br>