[Asterisk-video] A question about video clip playback

Antoine Fressancourt af.devlist at gmail.com
Fri Jun 30 09:32:24 MST 2006


It is really strange.

I send the following SDP to Asterisk by pressing the "start video" button,
in a reinvite :

v=0
o=- 169854971 169855023 IN IP4 172.18.141.130
s=eyeBeam
c=IN IP4 172.18.141.130
t=0 0
m=audio 9530 RTP/AVP 100 6 0 8 3 18 97 5
a=alt:1 1 : 72161194 00000005 172.18.141.130 9530
a=rtpmap:100 speex/16000
a=rtpmap:97 speex/8000
a=sendrecv
m=video 18000 RTP/AVP 34
a=alt:1 1 : 834F7A89 00000060 172.18.141.130 18000
a=sendrecv

I get another announcement back, with no video inside :

v=0
o=root 4571 4571 IN IP4 172.18.141.25
s=session
c=IN IP4 172.18.141.25
t=0 0
m=audio 17898 RTP/AVP 8 0
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=silenceSupp:off - - - -
a=sendrecv

Of course, I get an error message on Eyebeam...

Thank you very much for your help,

Antoine

2006/6/30, Ramtin Amin <keytwho at hotmail.com>:
>
> And whit the same eyebeam
> can you put in [general]
> videosuport=yes
> disallow=all
> allow=ulaw
> allow=allow
> allow=h263
>
>
> and in extensions.conf in ur default,
>
> exten => 9876,1,Answer()
> exten => 9876,2,echo()
>
> and try to see once you pressed on "start" button for the video if you see
> urself up there
>
>
>
>
>
>
>
> ------------------------------
> Date: Fri, 30 Jun 2006 17:21:36 +0200
>
>
> From: af.devlist at gmail.com
> To: asterisk-video at lists.digium.com
> Subject: Re: [Asterisk-video] A question about video clip playback
>
>
> Hello again,
>
> I worked a bit on my problem, trying to get the video to work.
>
> First, I tried recording a video from my eyebeam client on Asterisk using
> the Record() application. I use the following dialplan:
>
> [general]
> static=yes
> writeprotect=no
>
> [video]
>
> exten => 1000,1,Answer()
>
> exten => 1000,n,Wait(1)
> exten => 1000,n,Record(testmessage:h263)
> exten => 1000,n,Hangup()
>
> my sip.conf is :
>
> [general]
> bindport=5060                   ; UDP Port to bind to (SIP standard port
> is 5060)
> bindaddr=0.0.0.0                ; IP address to bind to ( 0.0.0.0 binds to
> all)
> videosupport=yes
>
> [antoine]
> type=friend
> videosupport=yes
> secret=antoine
> callerid="antoine"
> host=dynamic
> context=video
> disallow=all
> allow=gsm
> allow=h263
> dtmfmode=rfc2833
> canreinvite=no
>
>
> I explicitly loaded the format_h263.so module in modules.conf
>
> I get eyebeam to send a correct SDP announcement, saying this (SDP
> contained in the 200 OK from Asterisk to eyebeam):
>
> v=0
> o=root 16577 16577 IN IP4 172.18.141.25
> s=session
> c=IN IP4 172.18.141.25
> b=CT:384
> t=0 0
> m=audio 10830 RTP/AVP 3
> a=rtpmap:3 GSM/8000
> a=silenceSupp:off - - - -
> a=sendrecv
> m=video 18812 RTP/AVP 34
> a=rtpmap:34 H263/90000
> a=sendrecv
>
> In Asterisk I get the following error message :
>
>     -- Executing [1000 at video:1] Answer("SIP/antoine-4225", "") in new
> stack
>     -- Executing [1000 at video:2] Wait("SIP/antoine-4225", "1") in new stack
>     -- Executing [1000 at video:3] Record("SIP/antoine-4225",
> "testmessage:h263") in new stack
>     -- Playing 'beep' (language 'en')
> Jun 30 15:56:58 WARNING[4422]: translate.c:265 ast_translator_build_path:
> No translator path from g723 to unknown
> Jun 30 15:56:58 WARNING[4422]: file.c:193 ast_writestream: Unable to
> translate to format h263, source format gsm
> Jun 30 15:56:58 WARNING[4422]: app_record.c:276 record_exec: Problem
> writing frame
>   == Spawn extension (video, 1000, 3) exited non-zero on
> 'SIP/antoine-4225'
>
> 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:
>
> v=0
> o=root 16577 16577 IN IP4 172.18.141.25
> s=session
> c=IN IP4 172.18.141.25
> b=CT:384
> t=0 0
> m=audio 13678 RTP/AVP 3
> a=rtpmap:3 GSM/8000
> a=silenceSupp:off - - - -
> a=sendrecv
> m=video 15878 RTP/AVP 34
> a=rtpmap:34 H263/90000
> a=sendrecv
>
> Sniffing the connection with Ethereal, I receive no RTP packet for the
> video.
>
> When trying to play the video alone, I get the following error :
>
>     -- Executing [1000 at video:2] Playback("SIP/antoine-de5e",
> "/etc/asterisk/video/test") in new stack
> Jun 30 16:05:39 WARNING[4460]: file.c:557 ast_openstream_full: File
> /etc/asterisk/video/test does not exist in any format
> 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
> 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
>     -- Executing [1000 at video:3] Hangup("SIP/antoine-de5e", "") in new
> stack
>   == Spawn extension (video, 1000, 3) exited non-zero on
> 'SIP/antoine-de5e'
>
>
> Do you have any explanation ?
>
> Thank you very much for your time and help,
>
> Antoine
>
>
>
> ------------------------------
> Le futur Hotmail : Essayez Windows Live Mail Beta<http://ideas.live.com/programpage.aspx?versionId=5d21c51a-b161-4314-9b0e-4911fb2b2e6d>
>
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
>
> asterisk-video mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-video
>
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-video/attachments/20060630/f6ae42f8/attachment.htm


More information about the asterisk-video mailing list