It is really strange.<br><br>I send the following SDP to Asterisk by pressing the "start video" button, in a reinvite :<br><br>v=0<br>o=- 169854971 169855023 IN IP4 <a href="http://172.18.141.130">172.18.141.130</a>
<br>s=eyeBeam<br>c=IN IP4 <a href="http://172.18.141.130">172.18.141.130</a><br>t=0 0<br>m=audio 9530 RTP/AVP 100 6 0 8 3 18 97 5<br>a=alt:1 1 : 72161194 00000005 <a href="http://172.18.141.130">172.18.141.130</a> 9530<br>
a=rtpmap:100 speex/16000<br>a=rtpmap:97 speex/8000<br>a=sendrecv<br>m=video 18000 RTP/AVP 34<br>a=alt:1 1 : 834F7A89 00000060 <a href="http://172.18.141.130">172.18.141.130</a> 18000<br>a=sendrecv<br><br>I get another announcement back, with no video inside :
<br><br>v=0<br>o=root 4571 4571 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>t=0 0<br>m=audio 17898 RTP/AVP 8 0<br>a=rtpmap:8 PCMA/8000
<br>a=rtpmap:0 PCMU/8000<br>a=silenceSupp:off - - - -<br>a=sendrecv<br><br>Of course, I get an error message on Eyebeam...<br><br>Thank you very much for your help,<br><br>Antoine<br><br><div><span class="gmail_quote">2006/6/30, Ramtin Amin <
<a href="mailto:keytwho@hotmail.com">keytwho@hotmail.com</a>>:</span><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;"><div>
<div>And whit the same eyebeam<br>
can you put in [general]<br>
videosuport=yes<br>
disallow=all<br>
allow=ulaw<br>
allow=allow<br>
allow=h263<br>
<br>
<br>
and in <font>extensions.conf in ur default,</font><br>
<br>
exten => 9876,1,Answer()<br>
exten => 9876,2,echo()<br>
<br>
and try to see once you pressed on "start" button for the video if you see urself up there<br>
<br><br><br><br><br> <br>
<blockquote style="border-left: 2px solid rgb(0, 128, 128); padding-left: 5px; margin-left: 5px; margin-right: 0px;">
<hr>
Date: Fri, 30 Jun 2006 17:21:36 +0200</blockquote></div><div><span class="q"><br>From: <a href="mailto:af.devlist@gmail.com" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">af.devlist@gmail.com</a>
<br>To: <a href="mailto:asterisk-video@lists.digium.com" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">asterisk-video@lists.digium.com</a><br></span></div><div>Subject: Re: [Asterisk-video] A question about video clip playback
</div><div><span class="e" id="q_10c258b7d9d517e9_3"><br><br>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/" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">
0.0.0.0</a> ; IP address to bind to (<a href="http://0.0.0.0/" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)"> 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/" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">
172.18.141.25</a><br>s=session<br>c=IN IP4 <a href="http://172.18.141.25/" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">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/" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">172.18.141.25</a><br>s=session<br>c=IN IP4 <a href="http://172.18.141.25/" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">
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></span></div><div></div>
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