Posted: Tue Mar 27, 2007 8:17 pm Post subject: [Asterisk-video] app_rtsp connecting to vlc server ?
Hi,
I'm playing with app_rtsp, app_transcoder, app_h324m etc.
I would like to be able to use my 3G phone to dial in to an TE410 board
PRI E1 EuroISDN
and watch the rtsp-stream provided from a vlc 0.8.6a instance serving
the content of an USB-Webcam.
Is app_rtsp compatible/tested with vlc?
Unfortunately the call just falls thru, see call log below.
Is it right that app_h324m is not needed in this scenario?
-- Accepting call from '01701234567' to '12345' on channel 1/9, span 1
-- Executing [12345@pri1:1] Answer("Zap/9-1", "") in new stack
-- Executing [12345@pri1:2] transcode("Zap/9-1",
"|s@video-cam1|h263@qcif/fps=10/kb=52/qmin=4/qmax=12/gs=50") in new stack
-- Executing [s@video-cam1:1] Answer("Local/s@video-cam1-a243,2",
"") in new stack
-- Executing [s@video-cam1:2] rtsp("Local/s@video-cam1-a243,2",
"rtsp://172.17.33.82:8080/xyz.sdp") in new stack
-- Executing [s@video-cam1:3] Hangup("Local/s@video-cam1-a243,2",
"") in new stack
== Spawn extension (video-cam1, s, 3) exited non-zero on
'Local/s@video-cam1-a243,2'
== Auto fallthrough, channel 'Zap/9-1' status is 'UNKNOWN'
-- Hungup 'Zap/9-1'
- - -
BTW: I didn't yet manage to get the app_mp4 demo with record & play to work,
asterisk segfaults upon trying to play the recorded file, although
playing another manually converted&hinted video file works..
This is on an debian/unstable system with nearly all dev-libs (but
/usr/src/pwlib_v1_10_3/) from the debian repository.
Posted: Tue Mar 27, 2007 9:33 pm Post subject: [Asterisk-video] app_rtsp connecting to vlc server ?
----- Original Message -----
From: "Bruno Voigt" <Bruno.Voigt@ic3s.de>
Sent: Tuesday, March 27, 2007 10:09 PM
Quote:
Hi,
I'm playing with app_rtsp, app_transcoder, app_h324m etc.
I would like to be able to use my 3G phone to dial in to an TE410 board
PRI E1 EuroISDN
and watch the rtsp-stream provided from a vlc 0.8.6a instance serving
the content of an USB-Webcam.
Is app_rtsp compatible/tested with vlc?
No, I have not tested it with vlc, so it would be great if you could do it.
If you got any error just try to capture the rtstp negotiation with ethereal
and
send it to me with the output error and I'll try to fix it.
Quote:
Unfortunately the call just falls thru, see call log below.
Is it right that app_h324m is not needed in this scenario?
The problem with the app_h324m and app_rtsp is that when asterisk
tries to get the best format for comunitating both channels it doesn't
use AMR so it behaves as if it were no audio and fails.
I think that the best way of solving it would be using the AMR codec
patch that Paul Bagyenda sent to the asterisk-dev list. At least
asterisk would now about the AMR codec and would let the call progress.
Quote:
BTW: I didn't yet manage to get the app_mp4 demo with record & play to
work,
Quote:
asterisk segfaults upon trying to play the recorded file, although
playing another manually converted&hinted video file works..
I think I wroke the mp4_save in one of the latest updates, I'll try to fix
it as soon as I can.
Anyway try running asterisk with -f -g options and get a backtrace output
from the core
dump and we could fix the dump on erroneous mp4 files.
Greetings
Sergio Garcia
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