[Asterisk-video] A question about video clip playback
Antoine Fressancourt
af.devlist at gmail.com
Mon Jul 3 06:35:08 MST 2006
Hello,
I managed to get the SDP announcement correctly from Asterisk (it is still a
mystery, but now it works), And the problem is I don't get any video image
back after calling an Echo() extension. Which is troublesome.
Meanwhile, I tried again recording a video call on asterisk. I am using
H.263 codec, and I stiil have a message saying :
-- Executing [1001 at video:1] Answer("SIP/antoine-0361", "") in new stack
-- Executing [1001 at video:2] Wait("SIP/antoine-0361", "1") in new stack
-- Executing [1001 at video:3] Record("SIP/antoine-0361",
"testmessage:h263") in new stack
-- Playing 'beep' (language 'en')
Jul 3 14:15:48 WARNING[19363]: translate.c:265 ast_translator_build_path:
No translator path from g723 to unknown
Jul 3 14:15:48 WARNING[19363]: file.c:193 ast_writestream: Unable to
translate to format h263, source format gsm
Jul 3 14:15:48 WARNING[19363]: app_record.c:276 record_exec: Problem
writing frame
This makes me think that the H.263 codec is not properly supported by my
installation of Asterisk.
Do you have any idea about how to fix this ? Do you have a version of
Asterisk supporting video correctly ? In which branch of the CVS did you get
it ?
Thank you very much for your help,
Antoine
2006/6/30, Antoine Fressancourt <af.devlist at gmail.com>:
>
> 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>
> >
>
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