Posted: Tue Jun 20, 2006 12:48 am Post subject: [Asterisk-video] A question about video clip playback
Hello,
I have a question that may seem dumb, but I have tried to figure my problem
out by myself without success.
I try to connect to Asterisk with Eyebeam and have a video clip played back.
i followed instructions published on voipinfo, but I can't have my video
sent to the phone.
Looking at the SIP messages sent to the softphone, it seems that the video
is not advertised in the SDP messages. So it looks like the problem come
from my Asterisk configuration.
Posted: Tue Jun 20, 2006 12:53 am Post subject: [Asterisk-video] A question about video clip playback
Hello antoine
You have to give a lil bit more info about your tests
1st, I would recommend you to try what you are doing with a voicemail.
So first save the message with your client, then make sure you receive it when you play back voicemail.
At the same time, get a capture of the packets with ethereal so we could have a look at what's going on... and make sure there is something going through the RTP and also that the file message00x.h263 is not empty or only 0x00
Finally, try to use a more recent asterisk...
--
Amin Ramtin
Date: Tue, 20 Jun 2006 11:48:38 +0200From: af.devlist@gmail.comTo: asterisk-video@lists.digium.comSubject: [Asterisk-video] A question about video clip playbackHello,I have a question that may seem dumb, but I have tried to figure my problem out by myself without success.I try to connect to Asterisk with Eyebeam and have a video clip played back. i followed instructions published on voipinfo, but I can't have my video sent to the phone. Looking at the SIP messages sent to the softphone, it seems that the video is not advertised in the SDP messages. So it looks like the problem come from my Asterisk configuration.I here enclose my configuration files : -- sip.conf --[general];context=default bindport=5060 bindaddr=0.0.0.0videosupport=yes[antoine]type=friendsecret=antoinecallerid="Antoine"host=dynamiccontext=videodisallow=allallow=gsmallow=h263dtmfmode=rfc2833canreinvite=no-- extensions.conf --[general]static=yeswriteprotect=no[video]exten => 1000,1,Answer()exten => 1000,n,Playback(/etc/asterisk/video/test)exten => 1000,n,Hangup()the /etc/asterisk/video/ directory contains a test.gsm file and a test.h263 file.I run Asterisk 1.2.7.1 built from the sources published on the website.Thank you in advance for your help,Antoine Fressancourt
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Posted: Fri Jun 30, 2006 12:05 am Post subject: [Asterisk-video] A question about video clip playback
Hello?,
Sorry for not answering before.
I upgraded to the SVN version of Asterisk, following your advises. I am just
wondering before performing the tests you recommend if there is a specific
change to make in the voicemail.conf in order to accept video voicemails. I
think there may be a change to make in the codec property line, but I don't
know what to put there.
Sorry for these questions that may seem stupid to most of you guys :-)
Antoine
2006/6/20, Ramtin Amin <keytwho@hotmail.com>:
Quote:
Hello antoine
You have to give a lil bit more info about your tests
1st, I would recommend you to try what you are doing with a voicemail.
So first save the message with your client, then make sure you receive it
when you play back voicemail.
At the same time, get a capture of the packets with ethereal so we could
have a look at what's going on... and make sure there is something going
through the RTP and also that the file message00x.h263 is not empty or
only 0x00
I have a question that may seem dumb, but I have tried to figure my
problem out by myself without success.
I try to connect to Asterisk with Eyebeam and have a video clip played
back. i followed instructions published on voipinfo, but I can't have my
video sent to the phone.
Looking at the SIP messages sent to the softphone, it seems that the video
is not advertised in the SDP messages. So it looks like the problem come
from my Asterisk configuration.
the /etc/asterisk/video/ directory contains a test.gsm file and a
test.h263 file.
I run Asterisk 1.2.7.1 built from the sources published on the website.
Thank you in advance for your help,
Antoine Fressancourt
------------------------------
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[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
-- Executing [1000@video:1] Answer("SIP/antoine-4225", "") in new stack
-- Executing [1000@video:2] Wait("SIP/antoine-4225", "1") in new stack
-- Executing [1000@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:
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@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@video:3] Hangup("SIP/antoine-de5e", "") in new stack
== Spawn extension (video, 1000, 3) exited non-zero on 'SIP/antoine-de5e'
Posted: Fri Jun 30, 2006 6:25 am Post subject: [Asterisk-video] A question about video clip playback
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 +0200From: af.devlist@gmail.comTo: asterisk-video@lists.digium.comSubject: Re: [Asterisk-video] A question about video clip playbackHello 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=yeswriteprotect=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=friendvideosupport=yessecret=antoinecallerid="antoine"host=dynamiccontext=videodisallow=allallow=gsmallow=h263 dtmfmode=rfc2833canreinvite=noI 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=0o=root 16577 16577 IN IP4 172.18.141.25s=sessionc=IN IP4 172.18.141.25b=CT:384t=0 0m=audio 10830 RTP/AVP 3a=rtpmap:3 GSM/8000 a=silenceSupp:off - - - -a=sendrecvm=video 18812 RTP/AVP 34a=rtpmap:34 H263/90000a=sendrecvIn Asterisk I get the following error message : -- Executing [1000@video:1] Answer("SIP/antoine-4225", "") in new stack -- Executing [1000@video:2] Wait("SIP/antoine-4225", "1") in new stack -- Executing [1000@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 unknownJun 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=0o=root 16577 16577 IN IP4 172.18.141.25s=sessionc=IN IP4 172.18.141.25b=CT:384t=0 0m=audio 13678 RTP/AVP 3a=rtpmap:3 GSM/8000a=silenceSupp:off - - - - a=sendrecvm=video 15878 RTP/AVP 34a=rtpmap:34 H263/90000a=sendrecvSniffing 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@video:2] Playback("SIP/antoine-de5e", "/etc/asterisk/video/test") in new stackJun 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 directoryJun 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@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
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[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
-- Executing [1000@video:1] Answer("SIP/antoine-4225", "") in new
stack
-- Executing [1000@video:2] Wait("SIP/antoine-4225", "1") in new stack
-- Executing [1000@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:
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@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@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
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Posted: Mon Jul 03, 2006 4:35 am Post subject: [Asterisk-video] A question about video clip playback
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@video:1] Answer("SIP/antoine-0361", "") in new stack
-- Executing [1001@video:2] Wait("SIP/antoine-0361", "1") in new stack
-- Executing [1001@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@gmail.com>:
Quote:
It is really strange.
I send the following SDP to Asterisk by pressing the "start video" button,
in a reinvite :
2006/6/30, Ramtin Amin < keytwho@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@gmail.com
> To: asterisk-video@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@video:1] Answer("SIP/antoine-4225", "") in new
> stack
> -- Executing [1000@video:2] Wait("SIP/antoine-4225", "1") in new
> stack
> -- Executing [1000@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@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@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
>
>
>
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