[Asterisk-video] Help:app_rtsp cannot work in asterisk1.8

Wed Feb 29 19:54:25 CST 2012

Hi,
  I upload the wireshark capture packet log in below link

http://www.sendspace.com/file/t9lb6m

 I already use the app_rtsp.c in sub dir 1.8 .
I capture the wireshark log in this email.

The IP 10.1.1.192 is the VLC rtsp server .
The IP 10.1.1.89 is the asterisk server .

The softphone also use 10.1.1.192 as I run it in the win 7 .

I can use VLC client and use rtsp to read the video from VLC rtsp serve without problem...

But when I use sofphone to estable the call , it is not work ...

The extensions.conf like below 

exten => 2002,1,Answer()
exten => 2002,n,rtsp(rtsp://10.1.1.192:5544/)
exten => 2002,n,WaitExten(5)
exten => 2002,n,Hangup()

Please advice what the problem is .

Thank

--- 2012年3月1日 星期四，kingman chui <chuikingman at yahoo.com.hk> 寫道﹕

寄件人: kingman chui <chuikingman at yahoo.com.hk>
主題: Re: [Asterisk-video] Help:app_rtsp cannot work in asterisk1.8
收件人: "Development discussion of video media support in Asterisk" <asterisk-video at lists.digium.com>
副本(CC): sergio.garcia at fontventa.com
日期: 2012年3月1日,星期四,上午9:37

Hi,
  The attached file for packet caputre log cannot post in the forum .
Please advice where I can put/upload the wireshark log file you want ???

I already use the app_rtsp.c under direcotry 1.8 ...

Thank
Regard/chui king man

--- 2012年2月22日 星期三，Sergio Garcia Murillo <sergio.garcia at fontventa.com> 寫道﹕

寄件人: Sergio Garcia Murillo <sergio.garcia at fontventa.com>
主題: Re: [Asterisk-video] Help:app_rtsp cannot work in asterisk1.8
收件人: asterisk-video at lists.digium.com
日期: 2012年2月22日,星期三,上午7:00

Hi,

Are you using the app_rtsp under the 1.8 subdirectory? Also, could you get an ethreal cpature of the RSTP connection?

Best regards
Sergio

El 20/02/2012 14:49, kingman chui escribió: 

Hi,
  I install app_rtsp in 1.8.10.0-rc2  .
I cannot connect to rtsp VLC streaming server .
the rtsp VLC server is working find with VLC client and can play video in H263 format.
no audio .I disable audio .

I paste the extension.conf in below 
=======
exten => 2002,1,Answer()
exten => 2002,n,rtsp(rtsp://192.168.1.104:5544/)
exten => 2002,n,WaitExten(5)
exten => 2002,n,Hangup()
=======
I already enable the vidoe and code for the sip.
[20001000]
videosupport=yes
allow=all

But in the debug , the rtsp stream have not initiate and have not play .then it end ..

I paste part of the debug log below 

================
[Feb 21 05:25:26] DEBUG[11854] app_rtsp.c: -rtsp play loop [0]
[Feb 21 05:25:26] DEBUG[3636] manager.c: Examining event:
Event: Newexten^M
Privilege: dialplan,all^M
Channel: SIP/20001000-00000004^M
Context: from-internal^M
Extension: 2002^M
Priority: 2^M
Application: rtsp^M
AppData: rtsp://192.168.1.104:5544/^M
Uniqueid: 1329773126.4^M
^M
[Feb 21 05:25:26] DEBUG[11854] app_rtsp.c: -Receiving describe
[Feb 21 05:25:26] DEBUG[11854] app_rtsp.c: -Describe response code [200]
[Feb 21 05:25:26] DEBUG[11854] app_rtsp.c: -Receiving describe
[Feb 21 05:25:26] DEBUG[11854] app_rtsp.c: -Describe response code [2147483647]
[Feb 21 05:25:26] DEBUG[11854] app_rtsp.c: -rtsp_play end loop [0]
[Feb 21 05:25:26] WARNING[11854] app_rtsp.c: <rtsp_play
[Feb 21 05:25:26] DEBUG[11854] pbx.c: Launching 'WaitExten'
[Feb 21 05:25:26] VERBOSE[11854] pbx.c:     -- Executing [2002 at from-internal:3] WaitExten("SIP/20001000-00000004", "5") in new stack
[Feb 21 05:25:26] DEBUG[3636] manager.c: Examining event:
Event: Newexten^M

======

I attach the whole debug in this email 

Please advice how to fix it .
I already  know the app_rtsp have diff version in asterisk 1.6 and asterisk 1.8 .

I install the app_rtsp in asterisk 1.6 with app_rtsp using in 1.6 before .'
it is working .
But now , I install asterisk 1.8 and use app_rtsp with asterisk 1.8 version .
It is not work ..I already use the same set up environemnt .The same as I test in asterisk 1.6 .

Please advice why the app_rtsp using in asterisk 1.8 version not work ..

Thank
Regard/chui king man

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