[Asterisk-video] Need some help regarding Asterisk video setup
aman kumar
ursaman at gmail.com
Tue Apr 13 13:30:01 CDT 2010
Hello Klaus ,
I downloaded the source and tried to install but It seems there is some
problem .
It need libtools library , I download 2.2.6b but mpeg4ip tools are not
compatible with that version .
so I installed libtool 2.2.4 , but still getting same problem .
Could you please help , what version of libtool , we require to install
mpeg4ip tools and from where i can download it .
Also i am pasting my log file , it would be nice If someone help me to
understand the problem , why i am not able to play MP3 or Mp4 file.
Thanks ,
Aman
Extension.conf
[MYTEST]
exten => 2000,1,Dial(SIP/2000)
exten => 2001,1,Dial(SIP/2001)
exten => 2002,1,Answer()
exten => 2002,n,Playback(demo-thanks)
exten => 2002,n,Playback(vm-goodbye)
exten => 2003,1,Answer()
exten => 2003,n,MP3Player(/tmp/music/test1.mp3)
exten => 2003,n,Hangup()
exten => 2004,1,Answer()
exten => 2004,n,mp4play(/tmp/music/sample1.mp4)
exten => 2005,1,Answer()
exten => 2005,n,mp4play(/tmp/music/simpsons.mp4)
exten => 2006,1,Answer()
exten => 2006,n,Playback(simpsons)
exten => 2007,1,Answer()
exten => 2007,n,Playback(test3)
exten => 2008,1,Answer()
exten => 2008,n,Playback(simpsons.mp4)
exten => 2009,1,Answer()
exten => 2009,n,Playback(test3.mp3)
Log File :
Connected to Asterisk 1.6.2.6 currently running on d2c (pid = 3038)
Verbosity is at least 3
== Using SIP RTP CoS mark 5
-- Executing [2008 at MYTEST:1] Answer("SIP/2001-00000009", "") in new
stack
-- Executing [2008 at MYTEST:2] Playback("SIP/2001-00000009",
"simpsons.mp4") in new stack
[Apr 9 16:01:29] WARNING[22023]: file.c:650 ast_openstream_full: File
simpsons.mp4 does not exist in any format
[Apr 9 16:01:29] WARNING[22023]: file.c:953 ast_streamfile: Unable to open
simpsons.mp4 (format 0x4 (ulaw)): No such file or directory
[Apr 9 16:01:29] WARNING[22023]: app_playback.c:471 playback_exec:
ast_streamfile failed on SIP/2001-00000009 for simpsons.mp4
-- Auto fallthrough, channel 'SIP/2001-00000009' status is 'UNKNOWN'
== Using SIP RTP CoS mark 5
-- Executing [2002 at MYTEST:1] Answer("SIP/2001-0000000a", "") in new
stack
-- Executing [2002 at MYTEST:2] Playback("SIP/2001-0000000a",
"demo-thanks") in new stack
-- <SIP/2001-0000000a> Playing 'demo-thanks.slin' (language 'en')
== Spawn extension (MYTEST, 2002, 2) exited non-zero on
'SIP/2001-0000000a'
== Using SIP RTP CoS mark 5
-- Executing [2003 at MYTEST:1] Answer("SIP/2001-0000000b", "") in new
stack
-- Executing [2003 at MYTEST:2] MP3Player("SIP/2001-0000000b",
"/tmp/music/test1.mp3") in new stack
[Apr 9 16:01:47] NOTICE[22025]: app_mp3.c:113 timed_read: Poll timed
out/errored out with 0
-- Executing [2003 at MYTEST:3] Hangup("SIP/2001-0000000b", "") in new
stack
== Spawn extension (MYTEST, 2003, 3) exited non-zero on
'SIP/2001-0000000b'
== Using SIP RTP CoS mark 5
-- Executing [2004 at MYTEST:1] Answer("SIP/2001-0000000c", "") in new
stack
[Apr 9 16:01:55] WARNING[22027]: pbx.c:3680 pbx_extension_helper: No
application 'mp4play' for extension (MYTEST, 2004, 2)
== Spawn extension (MYTEST, 2004, 2) exited non-zero on
'SIP/2001-0000000c'
== Using SIP RTP CoS mark 5
-- Executing [2005 at MYTEST:1] Answer("SIP/2001-0000000d", "") in new
stack
[Apr 9 16:02:00] WARNING[22028]: pbx.c:3680 pbx_extension_helper: No
application 'mp4play' for extension (MYTEST, 2005, 2)
== Spawn extension (MYTEST, 2005, 2) exited non-zero on
'SIP/2001-0000000d'
[Apr 9 16:02:04] NOTICE[3325]: chan_sip.c:21369 handle_request_subscribe:
Received SIP subscribe for peer without mailbox: 2001
== Using SIP RTP CoS mark 5
-- Executing [2006 at MYTEST:1] Answer("SIP/2001-0000000e", "") in new
stack
-- Executing [2006 at MYTEST:2] Playback("SIP/2001-0000000e", "simpsons")
in new stack
[Apr 9 16:02:07] WARNING[22029]: file.c:650 ast_openstream_full: File
simpsons does not exist in any format
[Apr 9 16:02:07] WARNING[22029]: file.c:953 ast_streamfile: Unable to open
simpsons (format 0x4 (ulaw)): No such file or directory
[Apr 9 16:02:07] WARNING[22029]: app_playback.c:471 playback_exec:
ast_streamfile failed on SIP/2001-0000000e for simpsons
-- Auto fallthrough, channel 'SIP/2001-0000000e' status is 'UNKNOWN'
== Using SIP RTP CoS mark 5
-- Executing [2007 at MYTEST:1] Answer("SIP/2001-0000000f", "") in new
stack
-- Executing [2007 at MYTEST:2] Playback("SIP/2001-0000000f", "test3") in
new stack
-- <SIP/2001-0000000f> Playing 'test3.slin' (language 'en')
[Apr 9 16:02:14] WARNING[22030]: app_playback.c:471 playback_exec:
ast_streamfile failed on SIP/2001-0000000f for test3
-- Auto fallthrough, channel 'SIP/2001-0000000f' status is 'UNKNOWN'
== Using SIP RTP CoS mark 5
-- Executing [2008 at MYTEST:1] Answer("SIP/2001-00000010", "") in new
stack
-- Executing [2008 at MYTEST:2] Playback("SIP/2001-00000010",
"simpsons.mp4") in new stack
[Apr 9 16:02:27] WARNING[22033]: file.c:650 ast_openstream_full: File
simpsons.mp4 does not exist in any format
[Apr 9 16:02:27] WARNING[22033]: file.c:953 ast_streamfile: Unable to open
simpsons.mp4 (format 0x4 (ulaw)): No such file or directory
[Apr 9 16:02:27] WARNING[22033]: app_playback.c:471 playback_exec:
ast_streamfile failed on SIP/2001-00000010 for simpsons.mp4
-- Auto fallthrough, channel 'SIP/2001-00000010' status is 'UNKNOWN'
== Using SIP RTP CoS mark 5
-- Executing [2009 at MYTEST:1] Answer("SIP/2001-00000011", "") in new
stack
-- Executing [2009 at MYTEST:2] Playback("SIP/2001-00000011", "test3.mp3")
in new stack
[Apr 9 16:02:34] WARNING[22034]: file.c:650 ast_openstream_full: File
test3.mp3 does not exist in any format
[Apr 9 16:02:34] WARNING[22034]: file.c:953 ast_streamfile: Unable to open
test3.mp3 (format 0x4 (ulaw)): No such file or directory
[Apr 9 16:02:34] WARNING[22034]: app_playback.c:471 playback_exec:
ast_streamfile failed on SIP/2001-00000011 for test3.mp3
On Tue, Apr 6, 2010 at 3:42 AM, Klaus Darilion <klaus.mailinglists at pernau.at
> wrote:
> For app_mp4 you need the mpeg4ip tools.
>
> The tarball is very old, thus I usually checkout the code directly from
> CVS:
> http://sourceforge.net/projects/mpeg4ip/develop
>
> regards
> klaus
>
> Am 02.04.2010 20:21, schrieb aman kumar:
>
>> Hello Klaus ,
>> I am following your steps.But I am getting some problem after adding
>> "app_mp4.c", when I install asterisk . It is giving an error like
>> "mp4.h" include file not available . I tried to looked on the web but I
>> am not getting any source from where i could get this file.
>> Could you please suggest me something about it .
>> I would like to tell you about my setup configuration,
>> I installed CentOS5.4 + Asterisk 1.6.2.6 . + Asterisk Add on 1.6.2.0
>> + DAHDI 2.2.1.1 full package + LibPRI 1.4.10.2.
>> I also purchased Bria 3.0 (counterpath product , it supports video ) and
>> I am able to make SIP Voice calls through my asterisk server .
>> Thanks ,
>> Aman
>>
>>
>> On Mon, Mar 29, 2010 at 11:49 PM, Klaus Darilion
>> <klaus.mailinglists at pernau.at <mailto:klaus.mailinglists at pernau.at>>
>> wrote:
>>
>>
>>
>> Am 30.03.2010 02:12, schrieb aman kumar:
>>
>> Hello ,
>> I am a beginner user for asterisk and Linux environment . I
>> need some
>> help for setup video calls and video streaming with asterisk .I
>> will
>> appreciate , If you can me some links, white paper , examples or
>> any kind of help for that .I have downloaded AsteriskNow and
>> installed
>> in my server. I have following questions , please response
>> 1. Is AsteriskNow good for Video calls.
>>
>>
>> If you only want to make video calls between SIP phones, then it
>> should be OK.
>>
>> If you also do video streaming to SIP phones, or if you need video
>> telephony also to 3G UMTS video phones, then you need to patch
>> Asterisk and I think then it is better to use a plain Asterisk version.
>>
>>
>> 2. Do I need any addon / plug in / Hardware for the server for
>> video
>> calls .
>>
>>
>> For video calls: no, just enable video in sip.conf (search for video
>> in the sample config file)
>>
>> For 3G UMTS video calls: yes. You need ISDN (or SS7) interface cards
>>
>> For streaming (RTSP), recording and playback of video files: yes.
>> You need the applications from
>> https://asteriskvideo.svn.sourceforge.net/svnroot/asteriskvideo/
>>
>>
>> 3. How can we stream some static video contents through Asterisk.
>>
>>
>> For example you can store the video as mp4 file. Then use mp4play()
>> to playback the video to the caller. mp4play() is inlcuded in
>> app_mp4.c
>> (
>> https://asteriskvideo.svn.sourceforge.net/svnroot/asteriskvideo/app_mp4/app_mp4.c
>> )
>>
>>
>>
>> 4. Do we require any type of media server for contents or
>> streaming video.
>>
>>
>> You have 2 options:
>> 1. video is provided by Asterisk
>>
>> SIP phone <--SIP+RTP--> Asterisk
>> ^ (using mp4play())
>> |
>> video.mp4
>>
>>
>> 2. video is provided by a streaming server (e.g. darwin) and
>> Asterisk (app_rtsp.c) bridges between the SIP call and the video
>> stream.
>>
>> SIP phone <--SIP+RTP--> Asterisk <---RTSP-------> Darwin
>> (using rtsp()) ^
>> |
>> video.mp4
>>
>>
>>
>> 5. What are the basic steps for integration Video with asterisk
>> server.
>>
>>
>> Hint: only try the enxt step if the previous works
>>
>> 1. Install a "vanilla" (plain) Asterisk
>>
>> 2. Start with SIP-to-SIP video calls. By 2 webcams and use XLite to
>> make video calls. (enable video in sip.conf)
>>
>> 3. Try also with the Echo() application. Then the caller should see
>> itself.
>>
>> 4. Add app_mp4.c to Asterisk and reinstall Asterisk
>>
>> 5. Use mp4save() to record a video call from a SIP phone. Use
>> mpeg4ip tools to analyze the recorded video (audio, video and hint
>> tracks).
>>
>> 6. Use mp4play() to playback the recorded video to the SIP phone
>>
>> 7. use ffmpeg and mpeg4ip tools to create a video with hint tracks.
>> Then playback this video to the caller using mp4play.
>>
>> regards
>> klaus
>>
>>
>>
>> I have asked too many questions so that you can come to know ,
>> what is
>> my Goal. Please guide me .
>> Thanks ,
>> Aman
>>
>>
>>
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