Error in app_rtsp , asterisk 1.6

Hi,
I use asterisk 1.6.
I install the app_Rtsp.
In my config,
I make extensions.conf and use rtsp to access a vlc server (rtsp media server and play h263 video.

It is worked when I use a soft phone to make call to this rtsp media server through asterisk .
The video can play.

But the call can only last 21 min , around 1300 sec ,then it will drop call .
In asterisk log ,
there is below error

[Jan 24 04:40:11] DEBUG[3638] chan_sip.c:  Header  0 [  3]: jaK

[Jan 24 04:40:13] DEBUG[8399] rtp.c: Got RTCP report of 132 bytes

[Jan 24 04:40:15] DEBUG[8399] app_rtsp.c: -Sent rtcp video report [89]

[Jan 24 04:40:15] DEBUG[8399] rtp.c: Got RTCP report of 108 bytes

[Jan 24 04:40:16] DEBUG[8399] rtp.c: Got RTCP report of 132 bytes

[Jan 24 04:40:16] DEBUG[8399] app_rtsp.c: -Sent rtcp audio report [89]

[Jan 24 04:40:16] DEBUG[8399] app_rtsp.c: -Sent rtcp video report [89]

[Jan 24 04:40:16] DEBUG[8399] app_rtsp.c: >OPTIONS [/]

[Jan 24 04:40:16] DEBUG[8399] app_rtsp.c: <OPTIONS [/]

[Jan 24 04:40:16] DEBUG[8399] app_rtsp.c: -Sending OPTIONS and reseting RTCP timer

[Jan 24 04:40:16] ERROR[8399] app_rtsp.c: Error receiving response [0,89].Destination address required

[Jan 24 04:40:16] DEBUG[8399] app_rtsp.c: -rtsp_play end loop [0]

[Jan 24 04:40:16] DEBUG[8399] app_rtsp.c: -TEARDOWN

[Jan 24 04:40:16] WARNING[8399] app_rtsp.c: <rtsp_play

[Jan 24 04:40:16] DEBUG[8399] pbx.c: Launching 'WaitExten'

[Jan 24 04:40:16] VERBOSE[8399] pbx.c:     -- Executing [[email protected]:3] WaitExten("SIP/20001000-0000001b", "1") in new stack

[Jan 24 04:40:17] VERBOSE[8399] pbx.c:     -- Timeout on SIP/20001000-0000001b, continuing...

[Jan 24 04:40:17] DEBUG[8399] pbx.c: Launching 'Hangup'

What happen in thr rtcp message , why it say there is need destination address suddenly as the rtcp message is ok in the first 21 min …

Please advice what setting I can change to last the call without drop …

Thank
Regard/chui king man

Are you running FreePBX? You did not mention which version? Is the install from the distro or built by hand?

my freepb is 2.7

Please advice what the problem is in app_rtsp.c ???

This is not a FreePBX issue, it’s an Asterisk issue.

You may try the Asterisk forums.