Hello again,<br><br>I worked a bit on my problem, trying to get the video to work.<br><br>First, I tried recording a video from my eyebeam client on Asterisk using the Record() application. I use the following dialplan:<br>
<br>[general]<br>static=yes<br>writeprotect=no<br><br>[video]<br><br>exten => 1000,1,Answer()<br><br>exten => 1000,n,Wait(1)<br>exten => 1000,n,Record(testmessage:h263)<br>exten => 1000,n,Hangup()<br><br>my sip.conf
is :<br><br>[general]<br>bindport=5060 ; UDP Port to bind to (SIP standard port is 5060)<br>bindaddr=<a href="http://0.0.0.0">0.0.0.0</a> ; IP address to bind to (<a href="http://0.0.0.0">
0.0.0.0</a> binds to all)<br>videosupport=yes<br><br>[antoine]<br>type=friend<br>videosupport=yes<br>secret=antoine<br>callerid="antoine"<br>host=dynamic<br>context=video<br>disallow=all<br>allow=gsm<br>allow=h263
<br>dtmfmode=rfc2833<br>canreinvite=no<br><br><br>I explicitly loaded the format_h263.so module in modules.conf <br><br>I get eyebeam to send a correct SDP announcement, saying this (SDP contained in the 200 OK from Asterisk to eyebeam):
<br><br>v=0<br>o=root 16577 16577 IN IP4 <a href="http://172.18.141.25">172.18.141.25</a><br>s=session<br>c=IN IP4 <a href="http://172.18.141.25">172.18.141.25</a><br>b=CT:384<br>t=0 0<br>m=audio 10830 RTP/AVP 3<br>a=rtpmap:3 GSM/8000
<br>a=silenceSupp:off - - - -<br>a=sendrecv<br>m=video 18812 RTP/AVP 34<br>a=rtpmap:34 H263/90000<br>a=sendrecv<br><br>In Asterisk I get the following error message :<br><br> -- Executing [1000@video:1] Answer("SIP/antoine-4225", "") in new stack
<br> -- Executing [1000@video:2] Wait("SIP/antoine-4225", "1") in new stack<br> -- Executing [1000@video:3] Record("SIP/antoine-4225", "testmessage:h263") in new stack<br> -- Playing 'beep' (language 'en')
<br>Jun 30 15:56:58 WARNING[4422]: translate.c:265 ast_translator_build_path: No translator path from g723 to unknown<br>Jun 30 15:56:58 WARNING[4422]: file.c:193 ast_writestream: Unable to translate to format h263, source format gsm
<br>Jun 30 15:56:58 WARNING[4422]: app_record.c:276 record_exec: Problem writing frame<br> == Spawn extension (video, 1000, 3) exited non-zero on 'SIP/antoine-4225'<br><br>I performed other tests trying to play test.h263
encoded with ffmpeg and test.gsm containing the subsequent audio track. They were built from the same video file, and are the same length. I just have the sound track, while the SDP message stipulates:<br><br>v=0<br>o=root 16577 16577 IN IP4
<a href="http://172.18.141.25">172.18.141.25</a><br>s=session<br>c=IN IP4 <a href="http://172.18.141.25">172.18.141.25</a><br>b=CT:384<br>t=0 0<br>m=audio 13678 RTP/AVP 3<br>a=rtpmap:3 GSM/8000<br>a=silenceSupp:off - - - -
<br>a=sendrecv<br>m=video 15878 RTP/AVP 34<br>a=rtpmap:34 H263/90000<br>a=sendrecv<br><br>Sniffing the connection with Ethereal, I receive no RTP packet for the video.<br><br>When trying to play the video alone, I get the following error :
<br><br> -- Executing [1000@video:2] Playback("SIP/antoine-de5e", "/etc/asterisk/video/test") in new stack<br>Jun 30 16:05:39 WARNING[4460]: file.c:557 ast_openstream_full: File /etc/asterisk/video/test does not exist in any format
<br>Jun 30 16:05:39 WARNING[4460]: file.c:810 ast_streamfile: Unable to open /etc/asterisk/video/test (format 0x80002 (gsm|h263)): No such file or directory<br>Jun 30 16:05:39 WARNING[4460]: app_playback.c:439 playback_exec: ast_streamfile failed on SIP/antoine-de5e for /etc/asterisk/video/test
<br> -- Executing [1000@video:3] Hangup("SIP/antoine-de5e", "") in new stack<br> == Spawn extension (video, 1000, 3) exited non-zero on 'SIP/antoine-de5e'<br><br><br>Do you have any explanation ?<br>
<br>Thank you very much for your time and help,<br><br>Antoine<br><br><br>