[Live-devel] RTP player in browser

[Hujka Petr]

Hi all,
i need to make player of streamed content, which will work in IE, firefox
and opera. I don't want to have it as plugin, because of safety reasons. So
i think the best solution is to make it as an Java applet. I would like to
use combination of JMF+Live (if you know better solution, please write it).
I tried JMStudio, but it's not working, it looks like there is some problem
in RTP implementation on Sun's library (RTSP seems to be working). I have
this conclusion from JMStudio log:

#
# JMF Version 2.1.1e
#

## Platform: Windows XP, x86, 5.1
## Java VM: Sun Microsystems Inc., 1.5.0

## Player created: com.sun.media.content.rtsp.Handler at 110c31
##   using DataSource: com.sun.media.protocol.rtsp.DataSource at 13f991

## outgoing msg:
## DESCRIBE rtsp://172.31.149.238/test.mp3 RTSP/1.0
CSeq: 377
Accept: application/sdp
User-Agent: JMF RTSP Player Version 2.1.1e


## incoming msg:
## RTSP/1.0 200 OK
CSeq: 377
Date: Tue, Jan 16 2007 11:01:52 GMT
Content-Base: rtsp://172.31.149.238/test.mp3/
Content-Type: application/sdp
Content-Length: 302

v=0
o=- 5813127402 1 IN IP4 172.31.149.238
s=MPEG-1 or 2 Audio
i=test.mp3
t=0 0
a=tool:LIVE555 Streaming Media v2006.11.15
a=type:broadcast
a=control:*
a=range:npt=0-300.769
a=x-qt-text-nam:MPEG-1 or 2 Audio
a=x-qt-text-inf:test.mp3
m=audio 0 RTP/AVP 14
c=IN IP4 0.0.0.0
a=control:track1

## outgoing msg:
## SETUP rtsp://172.31.149.238/test.mp3/track1 RTSP/1.0
CSeq: 378
Transport: RTP/AVP;unicast;client_port=46674-46675
User-Agent: JMF RTSP Player Version 2.1.1e


## incoming msg:
## RTSP/1.0 200 OK
CSeq: 378
Date: Tue, Jan 16 2007 11:01:52 GMT
Transport: RTP/AVP;unicast;destination=172.31.149.238
;client_port=46674-46675;server_port=6972-6973
Session: 4


## outgoing msg:
## PLAY rtsp://172.31.149.238/test.mp3 RTSP/1.0
CSeq: 379
Range: npt=0.0-
Session:
User-Agent: JMF RTSP Player Version 2.1.1e


## incoming msg:
## RTSP/1.0 200 OK
CSeq: 379
Date: Tue, Jan 16 2007 11:01:52 GMT
Range: npt=0.000-
Session: 4
RTP-Info:
url=rtsp://172.31.149.238/test.mp3/track1;seq=35049;rtptime=1277791466


## RTP audio socket buffer size: 378 bytes.

XX Failed to realize: Server is not responding
$$ Profile: instantiation: 0 ms

## Player created: com.sun.media.content.unknown.Handler at 747fa2
##   using DataSource: com.sun.media.protocol.rtp.DataSource at 1be16f5

$$ Profile: parsing: 16 ms

## Building flow graph for: null

## Building Track: 0
## Input: mpegaudio/rtp, 44100.0 Hz, 16-bit, Stereo, LittleEndian, Signed


## Here's the completed flow graph:
  com.sun.media.parser.RawBufferParser at 1ae90c
     connects to: com.sun.media.codec.audio.mpa.DePacketizer at ba4211
     format: mpegaudio/rtp, 44100.0 Hz, 16-bit, Stereo, LittleEndian, Signed
  com.sun.media.codec.audio.mpa.DePacketizer at ba4211
     connects to: com.sun.media.codec.audio.mpa.NativeDecoder at 47a0d4
     format: mpegaudio, 44100.0 Hz, 16-bit, Stereo, Signed
  com.sun.media.codec.audio.mpa.NativeDecoder at 47a0d4
     connects to: com.sun.media.renderer.audio.DirectAudioRenderer at 55bb93
     format: LINEAR, 44100.0 Hz, 16-bit, Stereo, BigEndian, Signed


$$ Profile: graph building: 125 ms

$$ Profile: realize, post graph building: 16 ms
--------------------------------------------------------------------------------------------------------------------------
Is there anyone who has an idea what must be corrected or is there anyone
who dealed the same problem (browser rtp player+live server) and is able to
share solution?
Thanks for response.
Petr
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