VLC live video streaming via rtsp??

[redhawk]

I've been experimenting with VideoLAN VLC Player (2.0.1) and from what I understand it can act like a media server supporting several different protocols.
Unfortunately I have had no success trying to stream my TV card, web cam or media files using rtsp:// transport and I was wondering if anyone could enlighten me as to where I am going wrong.

To stream with VLC I do the following:

-open VLC
-"Media", "Open Capture Device" and setup a/v accordingly
-"Stream"
-"Next"
-tick "Display local"
-add "RTSP"
-tick "activate transcoding" with h.264 + aac (mp4)
-"Stream"

When I click on stream it plays the content as "display local" just fine however when I try to connect with another VLC window using rstp:// it fails every time.
I've tried changing the port numbers and url path, ip address local and external ISP IP but still no success.
My operating system Windows XP there are no network filewalls or blocks I can see VLC has created a listening socket but doesn't appear to want to communicate with the VLC client.

Any suggestions??

Richard S.

[edwardw]

What are the messages logs (Tools→Messages, set verbosity to 2) on both machines when running? Open Messages window and set verbosity to two on both client and server, and then try your stream. If it fails again, post the logs.

[redhawk]

I've fixed it for some reason I was missing encoder parameters this was only aparent when I got to the last window "Generated stream output string" it was set to :sout=#rtp{sdp=rtsp://:8554/test.sdp} :sout-keep
After reading another forum post it should have looked something like :sout=#transcode{vcodec=h264,vb=130,fps=5,scale=0.4,acodec=mp4a,ab=40,channels=1,samplerate=24000}:rtp{sdp=rtsp://:8554/test.sdp}

This works with VLC now unfortnately Android is having problems decoding the stream via hardware my video player keeps reverting to software rendering which is frankly useless.

Any idea what encoding parameters I need to make this compatible with Android??

Richard S.

[redhawk]

I've found a work around for Android and that's to use the h263 encoder unfortunately it seems playback only lasts for about 60 seconds.
I also have the same problem when playing audio only streams too via rtsp my it lasts for approximately 60 seconds then stops.
My Android handsets (2.1 and 2.3.7 CM7) works with other rtsp sites like TVCatchup so I either VLC's rtsp support is buggy or I am missing some extra settings.

Any ideas on how to fix this??

logcat data:
I/StagefrightPlayer( 131): setDataSource('rtsp://192.168.1.100:8554/test.sdp')
I/MyHandler( 131): connection request completed with result 0 (Unknown error: 0)
I/ARTSPConnection( 131): status: RTSP/1.0 200 OK
I/MyHandler( 131): DESCRIBE completed with result 0 (Unknown error: 0)
I/ASessionDescription( 131): v=0
I/ASessionDescription( 131): o=- 15245472888904482816 15245472888904482816 IN IP4 localhost
I/ASessionDescription( 131): s=Unnamed
I/ASessionDescription( 131): i=N/A
I/ASessionDescription( 131): c=IN IP4 0.0.0.0
I/ASessionDescription( 131): t=0 0
I/ASessionDescription( 131): a=tool:vlc 2.0.2
I/ASessionDescription( 131): a=recvonly
I/ASessionDescription( 131): a=type:broadcast
I/ASessionDescription( 131): a=charset:UTF-8
I/ASessionDescription( 131): a=control:rtsp://192.168.1.100:8554/test.sdp
I/ASessionDescription( 131): m=audio 0 RTP/AVP 96
I/ASessionDescription( 131): b=AS:40
I/ASessionDescription( 131): b=RR:0
I/ASessionDescription( 131): a=rtpmap:96 mpeg4-generic/24000
I/ASessionDescription( 131): a=fmtp:96 streamtype=5; profile-level-id=15; mode=AAC-hbr; config=130856e500; SizeLength=13; IndexLength=3; IndexDeltaLength=3; Profile=1;
I/ASessionDescription( 131): a=control:rtsp://192.168.1.100:8554/test.sdp/trackID=0
I/ASessionDescription( 131): m=video 0 RTP/AVP 96
I/ASessionDescription( 131): b=AS:130
I/ASessionDescription( 131): b=RR:0
I/ASessionDescription( 131): a=rtpmap:96 H263-1998/90000
I/ASessionDescription( 131): a=control:rtsp://192.168.1.100:8554/test.sdp/trackID=1
I/ARTSPConnection( 131): status: RTSP/1.0 200 OK
W/MyHandler( 131): Missing 'source' field in Transport response. Using RTSP endpoint address.
W/MyHandler( 131): Server picked an odd RTP port, it should've picked an even one, we'll let it pass for now, but this may break in the future.
I/ARTSPConnection( 131): status: RTSP/1.0 200 OK
I/MyHandler( 131): PLAY completed with result 0 (Unknown error: 0)
I/MyHandler( 131): received BYE on track index 0
I/MyHandler( 131): stream ended? aborting.
I/ARTSPConnection( 131): status: RTSP/1.0 200 OK
I/MyHandler( 131): TEARDOWN completed with result 0 (Unknown error: 0)
W/TimedEventQueue( 131): Event 62 was not found in the queue, already cancelled?
I/MyHandler( 131): TEARDOWN completed with result -107 (Transport endpoint is not connected)

Richard S.

[skrishna]

Hi Richard,

I am facing the same issue. The stream stops after few seconds and the logcat shows received BYE on track index 0. Were you able to figure out the problem?

Thanks,
Krishnan.

[sln]

Right now, I'm also sitting trying to find out how it works. Maybe I can solve your problem. When you are at the destination setup, you can add several destinations. Every time you choose a new one, remember to push the button Add. It's not enough just to choose it in the drop down.

[haribo]

In order to disable 60 seconds RTSP session timeout go to: Tools -> Preferences -> Change "Show settings" to "All" -> Stream output -> Sout stream -> RTP -> Change "RTP session timeout (s)" from 60 to 0.

Hope this helps

[Rémi Denis-Courmont]

And now, if a client crashes or looses network without disconnecting cleanly, your server will continue streaming forever, clogging your network.

Disabling timeouts is not exactly a very good idea... You really need to fix the RTSP receiver.