problem about rtsp-Collection of common programming errors
J_D
rtsp
I was under the impression that since a TEARDOWN request releases resources that are normally allocated when a SETUP is made, a TEARDOWN request was only necessary after the SETUP request.However, I just had an Android device that sent a TEARDOWN immediately after receiving the response to a DESCRIBE request (before the SETUP request, the Session: parameter of the request was empty).This was unexpected, and I was not able to have a confirmation, even by re-reading the RFC, if this is legit or no
realjin
java c jvm sip rtsp
I want to implement a high performance rtsp server which is to handle vod request — it only handles signaling request, it does not need to streaming the media file. I have accomplish a version that is written in Java basing on the Mina networking framework, and the performance seems to be not very high. As far as I know, high performance SIP server(e.g. VoIP server) is written in C (e.g. OpenSIPS, Kamailo), should I use C or C++ for my project to get a significant performance improvement?BTW.
LordNeckbeard
h.264 raspberry-pi rtsp libav
I’m using avconv on a Raspberry Pi to try to fetch a still image from an IP camera with an RTSP stream of H.264 video:$ avconv -v verbose -i $url -fflags discardcorrupt -t 00:00:01 -r 0.1 -an -vsync 1 -qscale 1 -f image2 images%09d.jpgI’m having some problems, presumably because the CPU on the Pi is not able to keep up with decoding the video, so sometimes the resulting JPEG is corrupted, for example:Probably 80% of the time I can get a valid image from the above command, but 20% of the time I g
slhck
ffmpeg h.264 nginx flv rtsp
Audio and video works for 900 seconds and then, ffmpeg stops with:video:1889kB audio:1825kB global headers:0kB muxing overhead 2.162866%I stream from an AXIS IP camera through ffmpeg over nginx.ffmpeg -v verbose -i rtsp://root:[email protected]/axis-media/media.amp -itsoffset 00:00:05 -acodec copy -ac 1 -ab 64k -ar 16000 -f flv -r 5 -qscale 1 -b 128kB rtmp://192.168.0.107/myapp/testI tested many commands, but this one gave me the best results.Console output:ffmpeg -v verbose -i rtsp://root:pass
sirlion
streaming video-streaming mp4 rtsp ffserver
I’m trying to stream a mp4 file over RTSP using ffserver with no luck so far. I just want to stream directly from the file, without feeding from ffmpeg (no transcoding involved). But I’ve made it work with mpg video.Here is my ffserver config filePort 8090 BindAddress 0.0.0.0 MaxHTTPConnections 2000 MaxClients 1000 MaxBandwidth 500000 CustomLog – NoDaemonRTSPPort 7654 RTSPBindAddress 0.0.0.0<Stream test1-rtsp>Format rtpFile “/home/g/video_streaming/sample3-mpeg2.mpg” </Stream> <St
Prashant
android media-player inputstream rtsp
I trying to play RTSP live streaming through both VideoView and MediaPlayer… I am able to get the stream both AAC audio and H264 Video. But after 15-20 minutes, RTSP stack is getting fulled and it’s crashing my application and tablet also… So is there any solution to increase the size of RTSP stack in Android source code? (I have whole AOSP ICS code). Any kind of help appreciable. @Ganesh Please find the Application Source Code below.package com.example.testvideo;import java.io.File; import
Emir Akaydin
streaming directshow rtsp
I have my own RTSP Source Filter solution which is still under development but working with H.264/MPEG-4 video streams at the moment. My problem is at initialization stage. If I open graphedit and add RTSP source, decoder and video renderer filters one by one and connect their pins, everything works fine (RTSP Source URL is hardcoded at the moment). But if I try to save the graph and re-open the graph from saved file, the graph crashes. I think the reason is at my RTSP Source Filter’s initializa
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android eclipse client-server videoview rtsp
I am writing rtsp client for android. As a server running vlc. Between computer and Android device connection is. But the video does not play. The server is started so vlc -v ~/paila.mp4 -I dummy –sout “#transcode{vcodec=h264,venc=x264{no-cabac,level=12,vbv-maxrate=384,vbv-bufsize=1000,keyint=75,ref=3,bframes=0},width=320, height=240,acodec=mp4a,ab=64,vb=384}:rtp{dst=,port=8888,sdp=rtsp://46.241.183.194:8888/stream.sdp,mp4a-latm}”. Here is the client code public class MainActivity ext
gezzuzz
vb.net vlc rtsp
I am using VLCPlugin2 in VB.net to watch a live RTSP Stream. I need it to be live as possible so I have been messing with the network-caching commandline option. 250ms crashes very easy and 300ms will crash after a few minutes. While I could increase the cache to 500ms which is on the upper end what I could tolerate I figure over time I could also get a crash. I would like to find a way to detect this crash so that I can restart the stream with a higher cache.. Starting at 300-350ms and if I kee
tirlototo
axis rtsp ip-camera mplayer
I’m using mplayer for playing RTSP streams (Axis IP Camera), and I would like to know how I can prevent mplayer from crashing when the IP cameras are unreachable (network problem, camera is shutdown, etc…). Currently, mplayer crash with this output: MPlayer interrupted by signal 11 in module: free_demuxer – MPlayer crashed by bad usage of CPU/FPU/RAM.Recompile MPlayer with –enable-debug and make a ‘gdb’ backtrace anddisassembly. Details in DOCS/HTML/en/bugreports_what.html#bugreports_crash. –
slhck
vlc video-streaming rtsp
I’m trying to figure out how to use the server capabilities of VLC. More specifically, how to export an SDP file when RTP streaming. In chapter 4 in the section related to RTP Streaming examples for server and client are given:vlc -vvv input_stream –sout ‘#rtp{dst=192.168.0.12,port=1234,sdp=rtsp://server.example.org:8080/test.sdp}’ vlc rtsp://server.example.org:8080/test.sdpIt’s not very clear to me how to make it actually work. I have tried these two commands for server and client using two cm
Androidnewbie
android rtsp
I am testing (RTSP/RTP/RTCP) streaming from my with HTC one X. After finishing the rtsp session i see udp packets reaching ports and after few secs, a fin ack from the mobile (in wireshark logs) for the RTSP port (550 in my case). While debugging with logcat, I see following linesI/ARTSPConnection( 144): status: RTSP/1.0 200 OKI/MyHandler( 144): SETUP(1) completed with result 0 (Success)W/MyHandler( 144): Missing ‘source’ field in Transport response. Using RTSP endpointI/ARTSPConnection( 144
Ben
android localhost media-player rtsp
I’m trying to “spoof” the android player. I want MediaPlayer (I’m using the VideoView wrapper) to connect to localhost so that I may reply to its RTSP requests. The application works fine when the underlying video system is the VisualOn type (VOME) however any device which uses the PacketView system (PVPlayer) the TCP connection is closed after I send the reply to the DESCRIBE request/challenge.I know for a fact the reply is not the issue. When I take the exact code (generating the same replies)
Peter O.
ios rtsp
I am new in the iPhone app development, I wants to do live streaming in the iPhone through my app. Please suggest me how to develop video streaming in the iPhone/iPad. Current, I am integrate dropcam library for rtsp video streaming in the iPhone, but after integration of the Live555, DecoderWrapper, ffmpeg and libswscale library in my project, I have get 15 error:Undefined symbols for architecture i386:”_av_register_all”, referenced from:+[VideoDecoder staticInitialize] in libDecoderWrapper.a(V
Chris Knadler
ffmpeg rtsp
I have an IPCamera on my LAN streaming video using RTSP. I have been able to capture and display it successfully using ffplay command:ffplay rtsp://admin:[email protected]:7070 (with authentication)So I would like to achieve the same using programming in C/C++ using ffmpeg library. I guess this must be possible.So let me phrase two simple questions :How do I receive the stream in a C/C++ program using FFMPEG library (just provide some URL/tutorial, as google was not helpful) How do I display
icktoofay
rtsp gstreamer gst
I’ve installed gst-rtsp-server and I wanted to try a simple code. But on compilation I’m getting the following error: In function `main’: test-launch01.c:(.text+0x64): undefined reference to `gst_rtsp_server_new’ test-launch01.c:(.text+0x74): undefined reference to `gst_rtsp_server_get_media_mapping’ test-launch01.c:(.text+0x7d): undefined reference to `gst_rtsp_media_factory_new’ test-launch01.c:(.text+0x95): undefined reference to `gst_rtsp_media_factory_set_shared’ test-launch01.c:(.text+0xad
slhck
ffmpeg streaming video-streaming rtsp
I am new to ffmpeg, just trying to build an SDP file for streaming. I’m facing an error when I run the command ffmpeg -re -i “sample.mp4” -f rtp rtp://127.0.0.1:10000 -flags +global_header -c:a libvo_aacenc -vn -map 0:1 -f rtp rtp://127.0.0.1:10002 > 10000.sdpThe error is:Unknown encoder ‘libvo_aacenc’How do I resolve this?
user1388555
android streaming rtsp android-wifi rtsp-client
I have been working on application to play different radio rtsp, when I try it out on a tablet which only uses wifi(with Android 4.3), I can actually hear the radio stations fine. However when I test it on a mobile ( with Android 2.3 ), I cannot get any results using its 3G connection. However when I turn on the mobile’s wifi, it works. For instance, this rtsp will not work for both cases: rtsp://movil.mediastream.com.pe/av_rpp/audio01What am I missing? why does it work with wifi and not on 3G?
AakashM
android video streaming rtsp ice-cream-sandwich
i have a problem with playback of stream video.On HTC Desire 2.2 video playing well, no problems but on Nexus One 2.3.4 and Nexus S 4.0.3 i have only audio, but no video displayedHere is block log that have equal problem on N1 and NS02-27 11:42:30.914: W/MyHandler(116): Unsupported format. Ignoring track #2. 02-27 11:42:30.914: I/MyHandler(116): SETUP(2) completed with result -1010 (Unknown error: 1010)full log:02-27 11:42:20.634: I/MyHandler(116): connection request completed with result 0 (Unk
Lance Roberts
java android mediaplayer rtsp
I get the following error12-15 16:54:37.125: ERROR/MediaPlayer(6032): error (-2147483648, 0)when trying to execute this code in a service:MediaPlayer mp = null;@Override public void onCreate() {if (mp == null) {mp = new MediaPlayer();} }@Override public void onStart(Intent intent, int startId) {if (!mp.isPlaying()) {Log.d(DEBUG_TAG, “Not playing, starting stream…”);try {mp.setDataSource(getString(R.string.address_p3));mp.prepare(); } catch (IOException e) {}mp.start();} else { Log.d(DEBUG_TAG,
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