[Asterisk-video] A question about video clip playback

[Antoine Fressancourt]

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
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