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 [2002@internal: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