For now, i havent found anything that doesnt work… There are few small issues with VLC when it is used as encoder, but i guess it will be repaired soon… The only thing that is a bit new for me is a “text” mode configuration… Is there any kind of GUI for Wowza or something like that?
Thank you guys… 
Once more… I really think that Wowza is one of the best streaming solutions that i have tried so far!
Sorry for my bad English!
I try to re-stream an rtsp stream and i got these warn log
2010-02-26 13:52:17 CST comment server WARN 200 - Failed to find an appropriate depacketizer for this track (X-KATA): You may not have the proper software license to use this feature. Please contact sales@wowza.com for more information. defaultVHost rtplive definst 20.031 - - - - - - - - - - - - - - - - - - - - - - - - -
2010-02-26 13:52:17 CST comment server WARN 200 - Failed to find an appropriate depacketizer for this track (H264): You may not have the proper software license to use this feature. Please contact sales@wowza.com for more information. defaultVHost rtplive definst 20.031 - - - - - - - - - - - - - - - - - - - - - - - - -
2010-02-26 13:52:17 CST comment server WARN 200 - Failed to find an appropriate depacketizer for this track (AMR): You may not have the proper software license to use this feature. Please contact sales@wowza.com for more information. defaultVHost rtplive definst 20.031 - - - - - - - - - - - - - - - - - - - - - - - - -
and i use evaluation verison.
Does this license problem ??? and how can i resolve this??
thx
It sounds like yoiu are trying to stream a video/audio format that we do not support. Please describe in detail the source of the video/audio and the format/codecs that are being used. What is the make/model of the camera or encoder? What settings are you using?
Charlie
my stream source is h.264/amr narrorband
Wowza and Flash does not support AMR.
Charlie
Can I ignore audio ?
Do I have to install a RTSP Broadcaster software or Wowza just do that?
I need to Re-stream a smil playlist to Iphone and other mobile devices.
Thanks
Hi guys…
I have Multiple IP Cameras and wants to build a server side module to get the streams from multiple IP cameras using RTSP and publish it on different rtmp URLs so that multiple clients can connect to it…
I could not find rtplive server programming example? are there any example of RTSP to RTMP server side module?
Thanks in advance.
Prince
Hi Charlie…
Thanks…
I was able to use it for one camera… what i was looking for is some kind of server side module example… so that i can create my on module for rtsp to rtmp…
Thanks…
Prince
I need help combining an audio and video stream from my restreamed RTSP ip camera. Yeah yeah I know Wowza doesn’t support this, but it’s such a common problem, there must be a solution!
I have an IP camera without audio (format is not important, I’m re-encoding with VLC). I would like to combine a live audio source from my sound card and stream it with the video to the RTPLive application (for example). I can stream either the video or the audio, both work perfectly, just not together (the vlc.sdp file gets overwritten and the stream gets switched, on the fly, to whatever instance of VLC I start last, without even rebuffering! IMPRESSIVE!).
Here is what I run for VLC to get the video feed:
vlc -vvv rtsp://192.168.0.61:554 --noaudio --sout “#transcode{venc=x264{keyint=75,profile=baseline,level=3.0,nocabac,qpmax=36,qpmin=10,me=hex,merange=24,subme=9,qcomp=0.6},vcodec=x264,vb=500,scale=.5,fps=25,acodec=none}:duplicate{dst=rtp{dst=127.0.0.1,port-video=10000,sdp=file://c:/progra~1/wowzam~1/wowzam~1.3/content/vlc.sdp}}”
Works amazing! Flawlessly fluid progressive scan (non-interlaced) video, no compression artificats, great resolution (downscaled from 1280x960!!), just amazing!
To get audio I can run:
vlc dshow:// :dshow-vdev=none :dshow-adev=“SoundMAX HD Audio” --sout=#transcode{acodec=mp4a,ab=192,channels=1,samplerate=48000}:rtp{ddst=127.0.0.1,port-audio=10002,sdp=file://c:/progra~1/wowzam~1/wowzam~1.3/content/vlc.sdp}
Both work great, just not together. Is there nothing I can do to fudge the vlc.sdp file to point at both instances to get the audio and video streams? If they are on seperate ports 10000 and 10002 why not?! I can post the contents of each SDP file if that would help. I manually merged them with notepad, nothing played, just buffered forever.
Audio video sync is not important!
I’m cross posting this question in some VLC help forums… If I could just combine my RTSP input and a dshow input together in VLC, my problem would be solved 
Works fine with two dshow inputs (i.e. usb webcam and sound card), just not with RTSP. My attempts at using RTSP directshow filters have failed, even though this camera uses a super simple RTSP string (my guess is it’s super funky RTP packetization).
Screen capture from the actual 500kbps webfeed! http://likestuff.globat.com/boundin.jpg
-Jamie M.
(p.s. Wowza Media Server 2.2.3, VLC 1.05 Golden Eye for audio, VLC 1.0.3 for video, Windows XP 32bit)
Take a look at :input-slave=dshow://… or a variation of that option.
Thanks! I might give er a try if my current solution doesn’t work out… in the mean time, I GOT WOWZA TO MERGE STREAMS! SWEEEEET!
My vlc.sdp file:
o=- 0 0 IN IP4 127.0.0.1
t=0 0
s=Unnamed
a=tool:vlc 1.0.5
c=IN IP4 127.0.0.1
m=video 10000 RTP/AVP 96
b=AS:500
a=rtpmap:96 H264/90000
a=fmtp:96 packetization-mode=1;profile-level-id=42c01e;sprop-parameter-sets=Z0LAHpZiAUB7YCIAAAMAAgAAAwBkHixckA==,aM4yyA==;
m=audio 10002 RTP/AVP 96
b=AS:192
a=rtpmap:96 mpeg4-generic/44100
a=fmtp:96 streamtype=5; profile-level-id=15; mode=AAC-hbr; config=1208; SizeLength=13; IndexLength=3; IndexDeltaLength=3; Profile=1;
My two instances of VLC:
audio…
vlc dshow:// :dshow-vdev=none :start-time=6 :dshow-adev=“SoundMAX HD Audio” --audio-desync=-1277 --sout=#transcode{acodec=mp4a,ab=192,channels=1,samplerate=44100}:rtp{dst=127.0.0.1,port-audio=10002}
video…
vlc -vvv rtsp://192.168.0.61:554 --noaudio --sout “#transcode{venc=x264{keyint=75,profile=baseline,level=3.0,nocabac,qpmax=36,qpmin=10,me=hex,merange=24,subme=9,qcomp=0.6},vcodec=x264,vb=500,scale=.5,fps=25,acodec=none}:duplicate{dst=rtp{dst=127.0.0.1,port-video=10000}}”
Works absolutely amazing. Exactly what I wanted 
I had to set the a/v sync manually, and after messing around I got it close to perfect at -1.227 seconds. I’ll download some a/v sync test videos and fine tune her when I’m more awake.
I purposly lagged the encoding machine (OCCT) till it dropped some serious frames and went all messy, THE AUDIO STAYED IN SYNC PERFECT! I dunno how, but it’s flawless. I live streamed for about 40 mins, perfect beginning to end, unplugged and re-plugged network cable to the ip cam, restarted the VLC sessions, crashed Wowza and restarted, sync sync sync
Wowza and VLC, you guys rock!!
Will post a test video as soon as I figure out how to make Wowza record the stream! (into perfectly split 10 minute chunks if possible)… for another night…
-Jamie M.
I’m using the “client/live.html” file and JW Player for testing. I used “rtmp://209.236../rtplive" as Server and "rtsp://*****.dyndns.org:554/” as Stream. I get this messages again and again:
INFO server comment - RTPMediaCaster.Reconnector[1440018046:rtplive/_definst_:rtsp://*******.dyndns.org:554/]: start: 1 INFO server comment - RTPSessionTracker.add[rtsp://*******.dyndns.org:554/]: 1 INFO server comment - RTPSessionDescriptionDataProviderBasicRTSPWorker.buildSDPData: sessionId:2047359602 sessionTimeout:0 INFO server comment - RTPSessionDescriptionDataProviderBasic.getStreamInfo[rtplive/_definst_]: RTSP/RTP re-streaming. Success, received SDP data. INFO server comment - sortPackets[1]: sortBufferSize:500 INFO stream create - - INFO stream publish rtsp://*******.dyndns.org:554/ - INFO server comment - RTPSessionDescriptionDataProviderBasicRTSPWorker.sessionStart: PLAY: rtsp://*******.dyndns.org:554/ INFO server comment - RTPSessionDescriptionDataProviderBasicRTSPWorker.processResponse: PLAY: rtsp://*******.dyndns.org:554/ INFO server comment - RTPMediaCaster.Reconnector[1440018046:rtplive/_definst_:rtsp://*******.dyndns.org:554/]: done: 1 INFO server comment - LivePlayer.play[rtplive/_definst_/rtsp://*******.dyndns.org:554/]: Dynamic Stream Markers are on.
I know nothing, so don’t listen to anything I say but…
I think you are using incorrect values for the “stream”, my “stream” just says vlc.sdp Are you using the “rtp-live” client playerdemo included with Wowza? When I use it on my wowza this is what it says:
INFO stream publish vlc.sdp -
INFO server comment - RTPUDPTransport.bind[rtplive/_definst_]: /127.0.0.1:10002
INFO server comment - RTPUDPTransport.bind[rtplive/_definst_]: /127.0.0.1:10003
INFO server comment - RTPUDPTransport.bind[rtplive/_definst_]: /127.0.0.1:10000
INFO server comment - RTPUDPTransport.bind[rtplive/_definst_]: /127.0.0.1:10001
INFO server comment - UDPTransport.firstPacket: /127.0.0.1:10002
INFO server comment - RTPMediaCaster.Reconnector[7437713:rtplive/_definst_:vlc.sdp]: done: 1
INFO server comment - UDPTransport.firstPacket: /127.0.0.1:10000
INFO server comment - RTCPHandler.sendFirstRTCPRR[591921152,10003,/127.0.0.1:3836]
INFO server comment - UDPTransport.firstPacket: /127.0.0.1:10003
INFO server comment - LiveStreamPacketizerCupertino.init[rtplive/_definst_/vlc.sdp]: chunkDurationTarget: 10000
INFO server comment - LiveStreamPacketizerCupertino.init[rtplive/_definst_/vlc.sdp]: chunkDurationTolerance: 500
INFO server comment - LiveStreamPacketizerCupertino.init[rtplive/_definst_/vlc.sdp]: audioGroupCount: 3
INFO server comment - LiveStreamPacketizerCupertino.init[rtplive/_definst_/vlc.sdp]: playlistChunkCount:3
INFO server comment - MediaStreamMap.getLiveStreamPacketizer: Create live stream
packetizer: cupertinostreamingpacketizer:vlc.sdp
INFO server comment - CupertinoPacketHandler.startStream[rtplive/_definst_/vlc.sdp]
INFO server comment - LiveStreamPacketizerCupertino.handlePacket[rtplive/_definst_/vlc.sdp]: Video codec:H264 isCompatible:true
INFO server comment - LiveStreamPacketizerCupertino.handlePacket[rtplive/_definst_/vlc.sdp]: Audio codec:AAC isCompatible:true
INFO server comment - MediaStreamMap.getLiveStreamPacketizer: Create live stream
packetizer: smoothstreamingpacketizer:vlc.sdp
INFO server comment - LiveStreamPacketizerCupertino.handlePacket[rtplive/_definst_/vlc.sdp][mp4a.40.2]: AAC Audio info: {AACFrame: size: 0, rate: 44100, channel
s: 1, samples: 1024, errorBitsAbsent: true, profileObjectType: "LC"}
INFO server comment - LiveStreamPacketizerSmoothStreaming.startStream[rtplive/_definst_/vlc.sdp]
INFO server comment - RTCPHandler.sendFirstRTCPRR[591921152,10001,/127.0.0.1:384
1]
INFO server comment - UDPTransport.firstPacket: /127.0.0.1:10001
Notice the “rtplive/definst/vlc.sdp” part, it should point to an sdp file, not to a url/port (that info should be INSIDE the sdp file)…
And I also don’t think your sdp file should say 10.0.0.50 unless Wowza is on the same internal network as your video server?! If Wowza IS on the same internal network as your video server, use wowza’s ip address (10.0.0.x) for testing (using a computer on the same internal network as Wowza and your video server), instead of your dyndns name (to rule out any router port forwarding/firewall issues, etc.).
Remember, I know nothing.
-Jamie M.
Hello Richard, i been reading your thread and i would like to ask a question.
while testing your video streaming can you tell me:
how much fps you get for a video stream,
the resolution size,
dos it take time to connect for the videos stream session?
regards
jan
Ups, í wondered why in the live.html there was no video, it only starts my stream.
Ok i figured out to receive the WOWZA stream in this form:
rtsp://localhost:1935/rtplive/mystream
When i start this with VLC Player, it requires a username and password, how can i disable or set this??
i only wanted to test the stream!
hello
I have open the 552 port
On VLC The stream work with direct link :
localy rtsp://ip local:552/MediaInput/h264 ok
exteriorly rtsp://ipfixe:552/MediaInput/h264 ok with a latency but it work
With wowza application rtplive and a flash example player live :
localy It work fine
exteriorly it doesn work
the console repeat with incrementation :
INFO server comment - RTPMediaCaster.Reconnector[30538754:rtplive/definst:camra2.stream]: done: 1 2 3 4 5 6 …
Hi,
I have a WMS re-streaming a VLC with RTSP, and the WMS is request SETUP of handshake process to VLC in this form:
...
2010-08-30 21:04:49 GMT-03:00 comment server DEBUG 200 - *** RTSPMessageSend ***
SETUP rtsp://127.0.0.1:554/teste/trackID=0 RTSP/1.0
Transport: RTP/AVP/TCP;unicast;interleaved=0-1
CSeq: 2
...
And the VLC response:
2010-08-30 21:04:49 GMT-03:00 comment server DEBUG 200 - uri: RTSP/1.0 461 Unsupported transport
And the stream no function
I see WMS functions normally in another server with: WMS + VLC RTSP with no problems, and I note a diference beetwen the SETUP command:
SETUP rtsp://127.0.0.1:554/teste/trackID=0 RTSP/1.0
Transport: RTP/AVP/UDP;unicast;..
CSeq: 2
The protocol used to SETUP the medis in this server is UDP, not RTP interleaved in TCP. Why i can configure this (or change TCP to UDP)?
Ohh!
(forceInterleaved = false) == NO FORCE INTERLEAVED!!! :D:D:D
Thanks man! I stay many hours try discovery this problem.
Hi charlie,
I was hoping you can help me w/something; I’m remotely logging onto a linux machine and starting up vlc through a terminal in the cmd line [typed “vlc”] and on the [windows machine] desktop starting the local vlc. I have an axis 207mw cam. I can stream live video from the cam from the linux machine through vlc & viewing it on the local vlc. Now on the remote vlc, I use: http://192.168.1.233/axis-cgi/mjpg/v…esolution=2CIF - network protocol window under Media --> Streaming --> Network tab. This cmd [http://192.168.1.233/axis-cgi/mjpg/v…solution=2CIF] now becomes the source w/http as the type. The destination is UDP & to be Displayed locally w/transcoding enabled w/the profile: Video - MPEG-2 + MPGA (TS). All output is now to be a transcoded stream delivered via UDP to the machine address 192.168.1.241 [the local windows machine that vlc will be running] & the port 1234. So after I click “Stream,” I’m prompted for the User name & Password & once I input that, the cam starts to stream live. On the other vlc [on the desktop], I click Media–>Open Network Stream–> and enter the Network Protocol : udp://@:1234, & click “play.” Now the video is being streamed from 1 machine to another, but I want to make it so that I don’t need to bring up vlc on the remote machine, but instead use the vlc-generated stream output string [under the “options” button “Generated stream output string” ] :sout=#transcode{vcodec=mp2v,vb=800,scale=1,acodec =mpga,ab=128,channels=2,samplerate=44100}:duplicat e{dst=std{access=udp,mux=ts,dst=192.168.1.241:1234 },dst=display} & like type that into a vi text editor & executing the file instead, so the only thing I’d see the the vlc live stream on the desktop. Is such a thing possible? Cause I know you can’t save settings on vlc either…what to do?
Thanks a bunch in advance
For some reason I am not seeing them. I will take a closer look. I mainly want the RR reports so I can control encoding rates based on the loss fraction. Interarrival jitter would be nice to have.
Is it possible to create an APP packet in the Wowza API and send out to the encoder RTCP connection?
Thanks. I am not seeing any packets over the RTCP channel. I will debug through my RTP library. More then likely an issue there.
Are there any plans to add statistical information in the RR packets? The current packets do not appear to be RTCP compound packets.