gst-launch-0.10 -vvv videotestsrc ! queue ! x264enc byte-stream=true bitrate=300 ! rtph264pay ! udpsink port=5000 host=127.0.0.1 sync=false
v=0
o=- 1188340656180883 1 IN IP4 127.0.0.1
s=Session streamed by GStreamer
i=server.sh
t=0 0
a=tool:GStreamer
a=type:broadcast
m=video 5000 RTP/AVP 96
c=IN IP4 127.0.0.1
a=rtpmap:96 H264/90000
Here's the debug log (relevant parts):
DEBUG server comment - cmd: play
INFO server comment - MediaStreamMediaCasterPlay: startPlay
INFO server comment - RTPMediaCaster.create
INFO server comment - RTPMediaCaster.init
DEBUG server comment - cmd: setBufferTime
INFO server comment - RTPMediaCaster.Reconnector: start
INFO server comment - RTPSessionDescriptionDataProviderBasic.getStreamInfo: /usr/local/WowzaMediaServerPro/content/myStream.sdp
DEBUG server comment - sdp: v=0
DEBUG server comment - sdp: o=- 1188340656180883 1 IN IP4 127.0.0.1
DEBUG server comment - sdp: s=Session streamed by GStreamer
DEBUG server comment - sdp: i=server.sh
DEBUG server comment - sdp: t=0 0
DEBUG server comment - sdp: a=tool:GStreamer
DEBUG server comment - sdp: a=type:broadcast
DEBUG server comment - sdp: m=video 5000 RTP/AVP 96
DEBUG stream setbuffertime myStream.sdp -
DEBUG server comment - sdp: c=IN IP4 127.0.0.1
DEBUG server comment - sdp: a=rtpmap:96 H264/90000
DEBUG server comment - onFlushNotifyClients: false
DEBUG server comment - flushInterval: 0
DEBUG server comment - verboseDebug: false
INFO stream create - -
DEBUG server comment - RTPDePacketizerRFC3984H264.init
INFO stream publish myStream.sdp -
INFO server comment - UDPTransport.bind: /127.0.0.1:5000
DEBUG server comment - config: session: setReuseAddress: from:false to:true
DEBUG server comment - config: session: setReceiveBufferSize: from:55296 to:65000
DEBUG server comment - config: session: setSendBufferSize: from:55296 to:55296
DEBUG server comment - config: session: setTrafficClass: from:0 to:0
INFO server comment - UDPTransport.bind: /127.0.0.1:5001
DEBUG server comment - config: session: setReuseAddress: from:false to:true
DEBUG server comment - config: session: setReceiveBufferSize: from:55296 to:65000
DEBUG server comment - config: session: setSendBufferSize: from:55296 to:55296
DEBUG server comment - config: session: setTrafficClass: from:0 to:0
INFO server comment - RTPMediaCaster.Reconnector: stop
DEBUG server comment - rtp[video:1024] {80 60 11 af 25 e9 6f d6 e3 2f 49 73 1c 89 30 00 }
DEBUG server comment - myStream.sdp: 17:36:15: waitforend: dropped:1
INFO server comment - UDPTransport.firstPacket: /127.0.0.1:5000
DEBUG server comment - rtp[video:286] {80 60 11 b0 25 e9 6f d6 e3 2f 49 73 1c 49 c7 bf }
DEBUG server comment - myStream.sdp: 17:36:15: waitforend: dropped:2
DEBUG server comment - rtp[video:1024] {80 60 11 b1 25 e9 7b 8f e3 2f 49 73 1c 89 10 00 }
DEBUG server comment - myStream.sdp: 17:36:15: waitforend: dropped:3
DEBUG server comment - rtp[video:990] {80 60 11 b2 25 e9 7b 8f e3 2f 49 73 1c 49 de 0a }
DEBUG server comment - myStream.sdp: 17:36:15: waitforend: dropped:4
DEBUG server comment - rtp[video:1024] {80 60 11 b3 25 e9 87 46 e3 2f 49 73 1c 89 30 00 }
DEBUG server comment - myStream.sdp: 17:36:15: waitforend: dropped:5
DEBUG server comment - rtp[video:148] {80 60 11 b4 25 e9 87 46 e3 2f 49 73 1c 49 49 9f }
DEBUG server comment - myStream.sdp: 17:36:15: waitforend: dropped:6
DEBUG server comment - rtp[video:1024] {80 60 11 b5 25 e9 92 fe e3 2f 49 73 1c 89 30 00 }
DEBUG server comment - myStream.sdp: 17:36:15: waitforend: dropped:7
DEBUG server comment - rtp[video:177] {80 60 11 b6 25 e9 92 fe e3 2f 49 73 1c 49 7a 2f }
Of course this won't be an easy task, yet i'm wondering what this log mean ? Why are all frames dropped ?
Cheers
Florent
Answer by Charlie Good · Jul 16, 2008 at 04:07 PM
Answer by Charlie Good · Aug 13, 2008 at 04:04 AM
Answer by Charlie Good · Nov 26, 2008 at 08:19 AM
Answer by Charlie Good · Dec 22, 2008 at 08:48 AM
gst-launch-0.10.exe -v filesrc location=videotestsrc_h264.avi ! avidemux ! x264enc byte-stream=true bitrate=300 ! rtph264pay mtu=1438 ! udpsink host=10.218.35.135 port=1234
Answer by Charlie Good · Jan 06, 2009 at 05:01 AM
gst-launch-0.10 -vvv videotestsrc ! queue ! x264enc byte-stream=true bitrate=300 ! rtph264pay ! udpsink port=5000 host=127.0.0.1 sync=false
Answer by Charlie Good · Jan 06, 2009 at 07:28 AM
Answer by Charlie Good · Jan 06, 2009 at 09:07 AM
Answer by Charlie Good · Jan 19, 2009 at 09:53 AM
Answer by Charlie Good · Jan 26, 2009 at 04:17 AM
Answer by Charlie Good · Jan 26, 2009 at 12:34 PM
1) why do i need to restart the server after stopping and restarting the (publisher) streaming process ? Is there a workaround ? If i do not, the flash player only shows a black screen
2) audio plays back faster
3) my audio tests samples (voice/claps) seem to have a direct effect on drifting speed ! If i speak longer, the desync comes faster !
This is a tough one. I just don't have the resources to dig into gstreamer and debug something like this. I know there are plenty of RTP based encoders where there is no issue with AV sync. So it is most likely not a Wowza Pro issue. There may be rounding issues in GStreamer. I have seen this problem in other encoders where they don't deal with timecodes properly and they calculate time differentials which insert small rounding errors that add up over time.
Charlie
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