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]
<mailto:[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]
<mailto:[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]
<mailto:[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]
<mailto:[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]
<mailto:[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
<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
_______________________________________________
--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
