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[Asterisk-video] app_transcoder

 
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sergio.garcia at fontvent
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PostPosted: Fri Mar 09, 2007 11:36 am    Post subject: [Asterisk-video] app_transcoder

Hi all..
 
I have just added a first (and early version) of a video transcoding application.
Currently only MPEG4 to H263 is working which will be needed if you plan to
test the app_rtsp application with a network camera, but i hope to have H263 to H263
for use with h324m gateway and been able to bridge a 3g videocall with a video
softphone.
I also still have to implement a better bitrate control and handle default parametes so
it doesn't segfault when one is missing... :)
 
You can have more info here:
http://sip.fontventa.com/content/view/30/57/
 
Greetings
Sergio Garcia
http://sip.fontventa.com/
 
 
 
[url=http://sip.fontventa.com/content/view/30/57/][/url]
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mbrancaleoni at espia.it
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PostPosted: Mon Mar 12, 2007 11:33 am    Post subject: [Asterisk-video] app_transcoder

Hi,

On Fri, 2007-03-09 at 12:49 +0100, Sergio Garcia Murillo wrote:
Quote:
Hi all..

I have just added a first (and early version) of a video transcoding
application.
Currently only MPEG4 to H263 is working which will be needed if you
plan to
test the app_rtsp application with a network camera, but i hope to
have H263 to H263
for use with h324m gateway and been able to bridge a 3g videocall with
a video
softphone.
I also still have to implement a better bitrate control and handle
default parametes so
it doesn't segfault when one is missing... :)

uhm...
Asterisk can build the app but cannot start it:
CLI> load app_transcoder.so
[Mar 12 12:29:28] WARNING[16670]: loader.c:362 load_dynamic_module:
Error loading module
'app_transcoder.so': /usr/lib/asterisk/modules/app_transcoder.so:
undefined symbol: av_realloc

uhm...
ld path is ok, since app_mp4 and libh324 that're on same libavcodec
path works ok....

any hint?

Greetings,
Matteo

P.S. I've here an axis netcam that handles rtsp. With ffplay and mplayer
I can see it, so I think will be no probs...



--
Matteo Brancaleoni
R&D Director
Tel :+39.02.70633354
Voip :sip:matteo@sip.voismart.it

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mbrancaleoni at espia.it
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PostPosted: Mon Mar 12, 2007 11:35 am    Post subject: [Asterisk-video] app_transcoder

Hi,

On Mon, 2007-03-12 at 12:32 +0100, matteo brancaleoni wrote:
Quote:
uhm...
Asterisk can build the app but cannot start it:
CLI> load app_transcoder.so
[Mar 12 12:29:28] WARNING[16670]: loader.c:362 load_dynamic_module:
Error loading module
'app_transcoder.so': /usr/lib/asterisk/modules/app_transcoder.so:
undefined symbol: av_realloc

just some more info about my libavcodec:

FFmpeg version SVN-r8164, Copyright (c) 2000-2007 Fabrice Bellard, et
al.
configuration: --enable-amr_nb --enable-amr_wb
libavutil version: 49.3.0
libavcodec version: 51.35.0
libavformat version: 51.10.0
built on Mar 1 2007 09:46:49, gcc: 3.4.6 20060404 (Red Hat 3.4.6-3)

Regards,

Matteo.

--
Matteo Brancaleoni
R&D Director
Tel :+39.02.70633354
Voip :sip:matteo@sip.voismart.it

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mbrancaleoni at espia.it
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PostPosted: Mon Mar 12, 2007 12:17 pm    Post subject: [Asterisk-video] app_transcoder

uhm,

On Mon, 2007-03-12 at 12:35 +0100, matteo brancaleoni wrote:

Quote:
FFmpeg version SVN-r8164, Copyright (c) 2000-2007 Fabrice Bellard, et
al.
configuration: --enable-amr_nb --enable-amr_wb
libavutil version: 49.3.0
libavcodec version: 51.35.0
libavformat version: 51.10.0
built on Mar 1 2007 09:46:49, gcc: 3.4.6 20060404 (Red Hat 3.4.6-3)


even with the latest svn of ffmpeg the issue is present...

matteo.


--
Matteo Brancaleoni
R&D Director
Tel :+39.02.70633354
Voip :sip:matteo@sip.voismart.it

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sergio.garcia at fontvent
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PostPosted: Mon Mar 12, 2007 1:07 pm    Post subject: [Asterisk-video] app_transcoder

My version of ffmpeg is

FFmpeg version SVN-rUNKNOWN, Copyright (c) 2000-2006 Fabrice Bellard, et al.


configuration: --prefix=/usr --libdir=${prefix}/lib --shlibdir=${prefix}/li
b --incdir=${prefix}/include/ffmpeg --enable-shared --enable-mp3lame --enabl
e-gpl --enable-faad --mandir=${prefix}/share/man --enable-vorbis --enable-pt
hreads --enable-faac --enable-xvid --enable-dts --enable-amr_nb --enable-amr
_wb --enable-pp --enable-libogg --enable-libgsm --enable-x264 --enable-a52 -
-extra-cflags=-Wall -g -fPIC -DPIC --cc=ccache cc
libavutil version: 49.1.0
libavcodec version: 51.28.0
libavformat version: 51.7.0
built on Jan 19 2007 17:05:58, gcc: 4.1.2 20061115 (prerelease) (Debian
4.1.1-21)

My app_transcoder.so doesn't have the av_realloc symboll, could you check
your're using the correct avcodec.h file?


----- Original Message -----
From: "matteo brancaleoni" <mbrancaleoni@espia.it>
To: "Development discussion of video media support in Asterisk"
<asterisk-video@lists.digium.com>
Sent: Monday, March 12, 2007 1:16 PM
Subject: Re: [Asterisk-video] app_transcoder


Quote:
uhm,

On Mon, 2007-03-12 at 12:35 +0100, matteo brancaleoni wrote:

> FFmpeg version SVN-r8164, Copyright (c) 2000-2007 Fabrice Bellard, et
> al.
> configuration: --enable-amr_nb --enable-amr_wb
> libavutil version: 49.3.0
> libavcodec version: 51.35.0
> libavformat version: 51.10.0
> built on Mar 1 2007 09:46:49, gcc: 3.4.6 20060404 (Red Hat 3.4.6-3)


even with the latest svn of ffmpeg the issue is present...

matteo.


--
Matteo Brancaleoni
R&D Director
Tel :+39.02.70633354
Voip :sip:matteo@sip.voismart.it

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sergio.garcia at fontvent
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PostPosted: Mon Mar 12, 2007 1:07 pm    Post subject: [Asterisk-video] app_transcoder

----- Original Message -----
From: "matteo brancaleoni" <mbrancaleoni@espia.it>
Sent: Monday, March 12, 2007 12:32 PM
Quote:
uhm...
ld path is ok, since app_mp4 and libh324 that're on same libavcodec
path works ok....

Neither app_mp4 nor libh324m uses the libavcodec library, so please chek
again
you're linking aginst the correct library file.

Greetings
Sergio Garcia

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mbrancaleoni at espia.it
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PostPosted: Mon Mar 12, 2007 1:38 pm    Post subject: [Asterisk-video] app_transcoder

Hi,

On Mon, 2007-03-12 at 14:18 +0100, Sergio Garcia Murillo wrote:
Quote:
My version of ffmpeg is
<snip>

Quote:
My app_transcoder.so doesn't have the av_realloc symboll, could you check
your're using the correct avcodec.h file?
the problem was int not having the shared libs.

I enabled them in ffmpeg and now starts ok.

BUT another problem arises....
seems that the transcode app does not like something.

Here's asterisk log:

-- Executing [0229411616@isdn:1] h324m_gw("mISDN/1-u0",
"201@isdn-video") in new stack
[Mar 12 14:29:32] DEBUG[25754]: app_queue.c:546 changethread: Device
'mISDN/1' changed to state '6' (Ringing) but we don't care because
they're not a member of any queue.
[Mar 12 14:29:32] DEBUG[25753]: app_h324m.c:386 app_h324m_gw:
h324m_loopback
[Mar 12 14:29:32] WARNING[25753]: channel.c:627 ast_best_codec: Don't
know any of 0x2000 formats
P[ 1] MGMT: SSTATUS: L2_ESTABLISH
[Mar 12 14:29:32] DEBUG[25755]: pbx.c:1767 pbx_extension_helper:
Launching 'Answer'
-- Executing [201@isdn-video:1]
Answer("Local/201@isdn-video-aeaf,2", "") in new stack
[Mar 12 14:29:32] DEBUG[25755]: devicestate.c:303
__ast_device_state_changed_literal: Notification of state change to be
queued on device/channel Local/201@isdn-video-aeaf,2
[Mar 12 14:29:32] DEBUG[25725]: devicestate.c:161 ast_device_state: No
provider found, checking channel drivers for Local - 201@isdn-video
[Mar 12 14:29:32] DEBUG[25725]: chan_local.c:145 local_devicestate:
Checking if extension 201@isdn-video exists (devicestate)
[Mar 12 14:29:32] DEBUG[25725]: channel.c:943 channel_find_locked:
Avoiding initial deadlock for channel '0x891cce0'
[Mar 12 14:29:32] DEBUG[25755]: pbx.c:1767 pbx_extension_helper:
Launching 'transcode'
[Mar 12 14:29:32] DEBUG[25725]: devicestate.c:287 do_state_change:
Changing state for Local/201@isdn-video - state 2 (In use)
[Mar 12 14:29:32] DEBUG[25756]: app_queue.c:546 changethread: Device
'Local/201@isdn-video' changed to state '2' (In use) but we don't care
because they're not a member of any queue.
-- Executing [201@isdn-video:2]
transcode("Local/201@isdn-video-aeaf,2", "|s@camera|
h263@qcif/fps=10/kb=52/qmin=4/qmax=12/gs=50") in new stack
-transcoding [,s@camera,h263@qcif/fps=10/kb=52/qmin=4/qmax=12/gs=50]
[Mar 12 14:29:32] WARNING[25755]: channel.c:2895 ast_request: No
translator path exists for channel type Local (native -1) to 5767168
[Mar 12 14:29:32] DEBUG[25755]: pbx.c:1767 pbx_extension_helper:
Launching 'Hangup'
-- Executing [201@isdn-video:3]
Hangup("Local/201@isdn-video-aeaf,2", "") in new stack
[Mar 12 14:29:32] DEBUG[25755]: pbx.c:2363 __ast_pbx_run: Spawn
extension (isdn-video,201,3) exited non-zero on
'Local/201@isdn-video-aeaf,2'
== Spawn extension (isdn-video, 201, 3) exited non-zero on
'Local/201@isdn-video-aeaf,2'


and this's my dialplan:
[isdn-video]
exten => 201,1,Answer
exten =>
201,2,transcode(,s@camera,h263@qcif/fps=10/kb=52/qmin=4/qmax=12/gs=50)
exten => 201,3,HangUp

[camera]
exten => s,1,Answer
exten => s,2,rtsp(rtsp://192.168.1.62/mpeg4/media.amp)
exten => s,3,HangUp

[from-isdn]
exten => _XXX.,1,h324m_gw(201@isdn-video)

any hint?

best regards,
matteo.

--
Matteo Brancaleoni
R&D Director
Tel :+39.02.70633354
Voip :sip:matteo@sip.voismart.it

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sergio.garcia at fontvent
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PostPosted: Mon Mar 12, 2007 2:19 pm    Post subject: [Asterisk-video] app_transcoder

Could you try first with a softphone without using the h324m ?

----- Original Message -----
From: "matteo brancaleoni" <mbrancaleoni@espia.it>
To: "Development discussion of video media support in Asterisk"
<asterisk-video@lists.digium.com>
Sent: Monday, March 12, 2007 2:35 PM
Subject: Re: [Asterisk-video] app_transcoder


Quote:
Hi,

On Mon, 2007-03-12 at 14:18 +0100, Sergio Garcia Murillo wrote:
> My version of ffmpeg is
<snip>
> My app_transcoder.so doesn't have the av_realloc symboll, could you
check

Quote:
> your're using the correct avcodec.h file?
the problem was int not having the shared libs.
I enabled them in ffmpeg and now starts ok.

BUT another problem arises....
seems that the transcode app does not like something.

Here's asterisk log:

-- Executing [0229411616@isdn:1] h324m_gw("mISDN/1-u0",
"201@isdn-video") in new stack
[Mar 12 14:29:32] DEBUG[25754]: app_queue.c:546 changethread: Device
'mISDN/1' changed to state '6' (Ringing) but we don't care because
they're not a member of any queue.
[Mar 12 14:29:32] DEBUG[25753]: app_h324m.c:386 app_h324m_gw:
h324m_loopback
[Mar 12 14:29:32] WARNING[25753]: channel.c:627 ast_best_codec: Don't
know any of 0x2000 formats
P[ 1] MGMT: SSTATUS: L2_ESTABLISH
[Mar 12 14:29:32] DEBUG[25755]: pbx.c:1767 pbx_extension_helper:
Launching 'Answer'
-- Executing [201@isdn-video:1]
Answer("Local/201@isdn-video-aeaf,2", "") in new stack
[Mar 12 14:29:32] DEBUG[25755]: devicestate.c:303
__ast_device_state_changed_literal: Notification of state change to be
queued on device/channel Local/201@isdn-video-aeaf,2
[Mar 12 14:29:32] DEBUG[25725]: devicestate.c:161 ast_device_state: No
provider found, checking channel drivers for Local - 201@isdn-video
[Mar 12 14:29:32] DEBUG[25725]: chan_local.c:145 local_devicestate:
Checking if extension 201@isdn-video exists (devicestate)
[Mar 12 14:29:32] DEBUG[25725]: channel.c:943 channel_find_locked:
Avoiding initial deadlock for channel '0x891cce0'
[Mar 12 14:29:32] DEBUG[25755]: pbx.c:1767 pbx_extension_helper:
Launching 'transcode'
[Mar 12 14:29:32] DEBUG[25725]: devicestate.c:287 do_state_change:
Changing state for Local/201@isdn-video - state 2 (In use)
[Mar 12 14:29:32] DEBUG[25756]: app_queue.c:546 changethread: Device
'Local/201@isdn-video' changed to state '2' (In use) but we don't care
because they're not a member of any queue.
-- Executing [201@isdn-video:2]
transcode("Local/201@isdn-video-aeaf,2", "|s@camera|
h263@qcif/fps=10/kb=52/qmin=4/qmax=12/gs=50") in new stack
-transcoding [,s@camera,h263@qcif/fps=10/kb=52/qmin=4/qmax=12/gs=50]
[Mar 12 14:29:32] WARNING[25755]: channel.c:2895 ast_request: No
translator path exists for channel type Local (native -1) to 5767168
[Mar 12 14:29:32] DEBUG[25755]: pbx.c:1767 pbx_extension_helper:
Launching 'Hangup'
-- Executing [201@isdn-video:3]
Hangup("Local/201@isdn-video-aeaf,2", "") in new stack
[Mar 12 14:29:32] DEBUG[25755]: pbx.c:2363 __ast_pbx_run: Spawn
extension (isdn-video,201,3) exited non-zero on
'Local/201@isdn-video-aeaf,2'
== Spawn extension (isdn-video, 201, 3) exited non-zero on
'Local/201@isdn-video-aeaf,2'


and this's my dialplan:
[isdn-video]
exten => 201,1,Answer
exten =>
201,2,transcode(,s@camera,h263@qcif/fps=10/kb=52/qmin=4/qmax=12/gs=50)
exten => 201,3,HangUp

[camera]
exten => s,1,Answer
exten => s,2,rtsp(rtsp://192.168.1.62/mpeg4/media.amp)
exten => s,3,HangUp

[from-isdn]
exten => _XXX.,1,h324m_gw(201@isdn-video)

any hint?

best regards,
matteo.

--
Matteo Brancaleoni
R&D Director
Tel :+39.02.70633354
Voip :sip:matteo@sip.voismart.it

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mbrancaleoni at espia.it
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PostPosted: Mon Mar 12, 2007 6:21 pm    Post subject: [Asterisk-video] app_transcoder

Hi,

On Mon, 2007-03-12 at 15:25 +0100, Sergio Garcia Murillo wrote:
Quote:
Could you try first with a softphone without using the h324m ?

I tried with a video softphone.

The app exits with a segfault after calling app_rtsp
(and app rtsp does nothing)

Here's the gdb log:
0 0x00fd74be in avcodec_close () from /usr/local/lib/libavcodec.so.51
(gdb) bt
#0 0x00fd74be in avcodec_close () from /usr/local/lib/libavcodec.so.51
#1 0x09c6a290 in ?? ()
#2 0x00000000 in ?? ()
(gdb) bt full
#0 0x00fd74be in avcodec_close () from /usr/local/lib/libavcodec.so.51
No symbol table info available.
#1 0x09c6a290 in ?? ()
No symbol table info available.
#2 0x00000000 in ?? ()
No symbol table info available.
(gdb)

not very useful, uh

then asterisk output:
[Mar 12 16:22:57] DEBUG[25895]: pbx.c:1767 pbx_extension_helper:
Launching 'Answer'
-- Executing [s@camera:1] Answer("Local/s@camera-3227,2", "") in new
stack
[Mar 12 16:22:57] DEBUG[25895]: devicestate.c:303
__ast_device_state_changed_literal: Notification of state change to be
queued on device/channel Local/s@camera-3227,2
[Mar 12 16:22:57] DEBUG[25895]: pbx.c:1767 pbx_extension_helper:
Launching 'rtsp'
-- Executing [s@camera:2] rtsp("Local/s@camera-3227,2",
"rtsp://192.168.1.62/mpeg4/media.amp") in new stack
[Mar 12 16:22:57] DEBUG[25862]: devicestate.c:161 ast_device_state: No
provider found, checking channel drivers for Local - s@camera
[Mar 12 16:22:57] DEBUG[25862]: chan_local.c:145 local_devicestate:
Checking if extension s@camera exists (devicestate)
[Mar 12 16:22:57] DEBUG[25862]: devicestate.c:287 do_state_change:
Changing state for Local/s@camera - state 2 (In use)
-rtsp_play loop
Quote:
DESCRIBE [/mpeg4/media.amp]
<DESCRIBE [/mpeg4/media.amp]

[Mar 12 16:22:57] DEBUG[25896]: app_queue.c:546 changethread: Device
'Local/s@camera' changed to state '2' (In use) but we don't care because
they're not a member of any queue.
[Mar 12 16:22:57] DEBUG[25891]: devicestate.c:303
__ast_device_state_changed_literal: Notification of state change to be
queued on device/channel Local/s@camera-3227,1
[Mar 12 16:22:57] DEBUG[25862]: devicestate.c:161 ast_device_state: No
provider found, checking channel drivers for Local - s@camera
[Mar 12 16:22:57] DEBUG[25862]: chan_local.c:145 local_devicestate:
Checking if extension s@camera exists (devicestate)
[Mar 12 16:22:57] DEBUG[25862]: devicestate.c:287 do_state_change:
Changing state for Local/s@camera - state 2 (In use)
[Mar 12 16:22:57] DEBUG[25897]: app_queue.c:546 changethread: Device
'Local/s@camera' changed to state '2' (In use) but we don't care because
they're not a member of any queue.
[Mar 12 16:22:57] DEBUG[25868]: chan_sip.c:4315 find_call: = Found Their
Call ID: KmqZucElAT@192.168.1.228 Their Tag ar8OwsdC4Z Our tag:
as735a27f3
[Mar 12 16:22:57] DEBUG[25868]: chan_sip.c:14453 handle_request: ****
Received ACK (6) - Command in SIP ACK
[Mar 12 16:22:57] DEBUG[25868]: chan_sip.c:2071 __sip_ack: Stopping
retransmission on 'KmqZucElAT@192.168.1.228' of Response 1: Match Not
Found
-Receiving describe
<rtsp_play[Mar 12 16:22:57] DEBUG[25895]: pbx.c:1767
pbx_extension_helper: Launching 'Hangup'
-- Executing [s@camera:3] Hangup("Local/s@camera-3227,2", "") in new
stack

any hint?

uhm
maybe I have to try with a vivotek cam ?

I've one here...

matteo.

--
Matteo Brancaleoni
R&D Director
Tel :+39.02.70633354
Voip :sip:matteo@sip.voismart.it

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sergio.garcia at fontvent
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PostPosted: Mon Mar 12, 2007 7:00 pm    Post subject: [Asterisk-video] app_transcoder

From: "matteo brancaleoni" <mbrancaleoni@espia.it>
Sent: Monday, March 12, 2007 6:46 PM


Quote:
I tried with a video softphone.

The app exits with a segfault after calling app_rtsp
(and app rtsp does nothing)

I've just uploaded two new version of app_transcoding and app_rtsp that fix
the problems you describe.
Please try with them again

Greetings
Sergio Garcia
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PostPosted: Tue Mar 13, 2007 8:39 am    Post subject: [Asterisk-video] app_transcoder

Hi,

On Mon, 2007-03-12 at 19:45 +0100, Sergio Garcia Murillo wrote:

Quote:
I've just uploaded two new version of app_transcoding and app_rtsp that fix
the problems you describe.
Please try with them again

ok, no more segs, now the calls goes up, but no video is seen,
and I have a lot of mpeg4 header damaged messages.
Here's my log:
-- Executing [s@camera:2] rtsp("Local/s@camera-e573,2",
"rtsp://192.168.1.62/mpeg4/media.amp") in new stack
[Mar 13 09:34:00] DEBUG[28295]: devicestate.c:303
__ast_device_state_changed_literal: Notification of state change to be
queued on device/channel Local/s@camera-e573,1
[Mar 13 09:34:00] DEBUG[28254]: devicestate.c:161 ast_device_state: No
provider found, checking channel drivers for Local - s@camera
[Mar 13 09:34:00] DEBUG[28254]: chan_local.c:145 local_devicestate:
Checking if extension s@camera exists (devicestate)
[Mar 13 09:34:00] DEBUG[28254]: devicestate.c:287 do_state_change:
Changing state for Local/s@camera - state 2 (In use)
[Mar 13 09:34:00] DEBUG[28301]: app_queue.c:546 changethread: Device
'Local/s@camera' changed to state '2' (In use) but we don't care because
they're not a member of any queue.
-rtsp_play loop
Quote:
DESCRIBE [/mpeg4/media.amp]
<DESCRIBE [/mpeg4/media.amp]

-Receiving describe
-line [v=0]
-line [o=- 1173778572783193 1173778572783202 IN IP4 192.168.1.62]
-line [s=Media Presentation]
-line [e=NONE]
-line [c=IN IP4 0.0.0.0]
-line [b=AS:2048]
-line [t=0 0]
-line [a=control:*]
-line [a=range:npt=now-]
-line [a=mpeg4-iod:
"data:application/mpeg4-iod;base64,AoEAAE8BAf71AQOAkwABQHRkYXRhOmFwcGxpY2F0aW9uL21wZWc0LW9kLWF1O2Jhc2U2NCxBVGdCR3dVZkF4Y0F5U1FBWlFRTklCRUVrK0FBSDBBQUFCOUFBQVlCQkFFWkFwOERGUUJsQlFRTlFCVUFDN2dBQUFBQUFBQUFBQVlCQXc9PQQNAQUABAAAAAAAAAAAAAYJAQAAAAAAAAAAA0IAAkA+ZGF0YTphcHBsaWNhdGlvbi9tcGVnNC1iaWZzLWF1O2Jhc2U2NCx3QkFTZ1RBcUJYSmhCSWhRUlFVL0FBPT0EEgINAAACAAAAAAAAAAAFAwAAQAYJAQAAAAAAAAAA"]
-line [m=video 0 RTP/AVP 96]
-creating media [1,m=video 0 RTP/AVP 96]
-line [b=AS:2048]
-line [a=control:trackID=1]
-line [a=rtpmap:96 MP4V-ES/90000]
-line [a=fmtp:96 profile-level-id=245;
config=000001B0F5000001B509000001000000012008D48D8803250F045014440F;]
-line [a=mpeg4-esid:201]
-line [m=audio 0 RTP/AVP 0]
-creating media [1,m=audio 0 RTP/AVP 0]
-line [a=control:trackID=2]
-line [l-id=245;
config=000001B0F5000001B509000001000000012008D48D8803250F045014440F;]
-line [a=mpeg4-esid:201]
-line [m=audio 0 RTP/AVP 0]
-creating media [1,m=audio 0 RTP/AVP 0]
-line [a=control:trackID=2]
-audio [0,-1,trackID=2]
-video [4194304,96,trackID=1]
-SETUP VIDEO [trackID=1]
-PLAY [/mpeg4/media.amp]
[Mar 13 09:34:00] DEBUG[28260]: chan_sip.c:4315 find_call: = Found Their
Call ID: cTKZf7rfdc@192.168.1.228 Their Tag VZoJzWtrSw Our tag:
as4a6068ab
[Mar 13 09:34:00] DEBUG[28260]: chan_sip.c:14453 handle_request: ****
Received ACK (6) - Command in SIP ACK
[Mar 13 09:34:00] DEBUG[28260]: chan_sip.c:2071 __sip_ack: Stopping
retransmission on 'cTKZf7rfdc@192.168.1.228' of Response 1: Match Not
Found
[mpeg4 @ 0x4142148]header damaged
[mpeg4 @ 0x4142148]header damaged
[mpeg4 @ 0x4142148]header damaged
[Mar 13 09:34:01] DEBUG[28295]: rtp.c:862 ast_rtcp_read: Got RTCP report
of 60 bytes
[mpeg4 @ 0x4142148]header damaged
[Mar 13 09:34:01] NOTICE[28295]: rtp.c:1245 ast_rtp_read: Unknown RTP
codec 120 received from '192.168.1.228'
[Mar 13 09:34:01] DEBUG[28295]: rtp.c:862 ast_rtcp_read: Got RTCP report
of 60 bytes
[mpeg4 @ 0x4142148]header damaged
[mpeg4 @ 0x4142148]header damaged
...last message repeated until hungup....

mind that with ffplay I can see the rtsp url.
The netcam is an axis one.

any hint on where I can look @ ?

Greetings,

Matteo
--
Matteo Brancaleoni
R&D Director
Tel :+39.02.70633354
Voip :sip:matteo@sip.voismart.it

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sergio.garcia at fontvent
Guest





PostPosted: Tue Mar 13, 2007 10:54 pm    Post subject: [Asterisk-video] app_transcoder

From: "matteo brancaleoni" <mbrancaleoni@espia.it>
Sent: Tuesday, March 13, 2007 9:39 AM


Quote:
On Mon, 2007-03-12 at 19:45 +0100, Sergio Garcia Murillo wrote:

> I've just uploaded two new version of app_transcoding and app_rtsp that
fix

Quote:
> the problems you describe.
> Please try with them again

ok, no more segs, now the calls goes up, but no video is seen,
and I have a lot of mpeg4 header damaged messages.

I think it should be fixed now, there was a stupid misplaced semicolom
and a bad closing bracket :)

Greetings
Sergio García
http://sip.fontventa.com

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