VLC RTSP直播到Android

对于我的应用程序，我必须从甲板连接卡流到Android应用程序（我必须是一个实时流，所以HLS或RTSP似乎是很好的解决方案，因为我的应用程序的目标是Android 3+ ）。我使用decklink sdk重新编译了VLC，并且能够通过网络直播到另一台电脑（但它仅与RTSP一起工作60秒）。VLC RTSP直播到Android

这里是我的尝试：

HTTP流：

./vlc -vvv decklink:// --sout 
'#transcode{vcodec=mp4v,acodec=mpga,vb=56,ab=24,channels=1} 
:standard{access=http{use-key-frames},mux=ts,dst=:3001/stream.mpeg}'

它是工作在Android的VLC 0.0.11，但只有在无线网络，而不是3G。而且我无法使用VideoView在我的应用中播放它。这里是我使用的代码和对应的错误消息：

String url = "http://134.246.63.169:5554/stream.mpeg"; 

VideoView videoView = (VideoView) this.findViewById(R.id.videoView); 
videoView.setVideoURI(Uri.parse(url));   
videoView.setMediaController(new MediaController(this)); 
videoView.requestFocus(); 
videoView.start();

错误消息：

04-08 15:26:46.272: D/MediaPlayer(16349): Couldn't open file on client side, trying server side 
04-08 15:26:46.272: V/ChromiumHTTPDataSource(7680): connect on behalf of uid 1080867789 
04-08 15:26:46.272: I/ChromiumHTTPDataSource(7680): connect to http://134.246.63.169:8554/ @0 
04-08 15:26:46.302: I/AwesomePlayer(7680): AwesomePlayer::AwesomePlayer()in 
04-08 15:26:46.302: I/AwesomePlayer(7680): AwesomePlayer::AwesomePlayer()aftermClient.connect() 
04-08 15:26:46.302: I/AwesomePlayer(7680): setDataSource_l('http://134.246.63.169:5554/') 
04-08 15:26:46.302: W/MediaPlayer(16349): info/warning (701, 0) 
04-08 15:26:46.302: V/ChromiumHTTPDataSource(7680): connect on behalf of uid 10067 
04-08 15:26:46.302: I/ChromiumHTTPDataSource(7680): connect to http://134.246.63.169:5554/ @0 
04-08 15:26:46.342: I/ActivityManager(272): Displayed fr.ifremer.testrtsp/.MainActivity: +183ms 
04-08 15:26:46.382: I/MediaPlayer(16349): Info (701,0) 
04-08 15:27:07.592: E/MediaPlayer(16349): error (1, -2147483648) 
04-08 15:27:07.592: E/MediaPlayer(16349): Error (1,-2147483648)

RTSP：

我用this page由谷歌建议的编码选项，例如：

视频编解码器：H264
音频编解码器：AAC
视频比特率：56
音频比特率：24个
音频通道：1

尺寸：176×144

./vlc -vvv decklink:// --sout-ffmpeg-strict=-2 --sout 
'#transcode{width=176,height=144,vcodec=h264,acodec=mp4a,vb=56,ab=24,channels=1} 
:rtp{dst=134.246.63.169,port-video=5554,port-audio=5556,sdp=rtsp://134.246.63.169:5554/stream.sdp}'

我能够在VLC桌面播放流，但不能在Android中播放（即使在Android VLC版本或默认的Goo中gle视频播放器：/）。如果我没有指定muxer，我也可以用它来播放QuickTime（如果我指定muxer，无论是ts还是ps，我都没有视频，如果我尝试另一个muxer，VLC告诉我，我只允许使用TS或RTP PS）

如果我尝试用谷歌视频播放器，我得到的LOCAT这些消息：

04-08 15:32:45.792: D/MediaPlayer(13688): Couldn't open file on client side, trying server side 
04-08 15:32:45.802: W/MediaPlayer(13688): info/warning (701, 0) 
04-08 15:32:45.812: I/MediaPlayer(13688): Info (701,0) 
04-08 15:32:45.812: D/MediaPlayer(13688): getMetadata 
04-08 15:32:45.812: E/MediaPlayerService(7680): getMetadata failed -38 
04-08 15:32:45.852: I/MyHandler(7680): connection request completed with result 0 (Success) 
04-08 15:32:45.882: I/ARTSPConnection(7680): status: RTSP/1.0 200 OK 
04-08 15:32:45.882: I/MyHandler(7680): DESCRIBE completed with result 0 (Success) 
04-08 15:32:45.882: I/ASessionDescription(7680): v=0 
04-08 15:32:45.882: I/ASessionDescription(7680): o=- 15352003113363922923 15352003113363922923 IN IP4 to63-169.ifremer.fr 
04-08 15:32:45.882: I/ASessionDescription(7680): s=Unnamed 
04-08 15:32:45.882: I/ASessionDescription(7680): i=N/A 
04-08 15:32:45.882: I/ASessionDescription(7680): c=IN IP4 134.246.63.169 
04-08 15:32:45.882: I/ASessionDescription(7680): t=0 0 
04-08 15:32:45.882: I/ASessionDescription(7680): a=tool:vlc 2.0.5 
04-08 15:32:45.882: I/ASessionDescription(7680): a=recvonly 
04-08 15:32:45.882: I/ASessionDescription(7680): a=type:broadcast 
04-08 15:32:45.882: I/ASessionDescription(7680): a=charset:UTF-8 
04-08 15:32:45.882: I/ASessionDescription(7680): a=control:rtsp://134.246.63.169:5554/stream.sdp 
04-08 15:32:45.882: I/ASessionDescription(7680): m=audio 5556 RTP/AVP 96 
04-08 15:32:45.882: I/ASessionDescription(7680): b=AS:24 
04-08 15:32:45.882: I/ASessionDescription(7680): b=RR:0 
04-08 15:32:45.882: I/ASessionDescription(7680): a=rtpmap:96 mpeg4-generic/48000 
04-08 15:32:45.882: I/ASessionDescription(7680): a=fmtp:96 streamtype=5; profile-level-id=15; mode=AAC-hbr; config=118856e500; SizeLength=13; IndexLength=3; IndexDeltaLength=3; Profile=1; 
04-08 15:32:45.882: I/ASessionDescription(7680): a=control:rtsp://134.246.63.169:5554/stream.sdp/trackID=0 
04-08 15:32:45.882: I/ASessionDescription(7680): m=video 5554 RTP/AVP 96 
04-08 15:32:45.882: I/ASessionDescription(7680): b=AS:56 
04-08 15:32:45.882: I/ASessionDescription(7680): b=RR:0 
04-08 15:32:45.882: I/ASessionDescription(7680): a=rtpmap:96 H264/90000 
04-08 15:32:45.882: I/ASessionDescription(7680): a=fmtp:96 packetization-mode=1;profile-level-id=64000b;sprop-parameter-sets=Z2QAC6zZQsTv/AC0ALBAAAADAEAAAAyjxQplgA==,aOvssiw=; 
04-08 15:32:45.882: I/ASessionDescription(7680): a=control:rtsp://134.246.63.169:5554/stream.sdp/trackID=1 
04-08 15:32:45.982: I/ARTSPConnection(7680): status: RTSP/1.0 200 OK 
04-08 15:32:45.982: I/MyHandler(7680): SETUP(1) completed with result 0 (Success) 
04-08 15:32:45.982: I/MyHandler(7680): server specified timeout of 60 secs. 
04-08 15:32:45.992: W/MyHandler(7680): Missing 'source' field in Transport response. Using RTSP endpoint address. 
04-08 15:32:45.992: I/APacketSource(7680): dimensions 176x144 
04-08 15:32:46.012: I/ARTSPConnection(7680): status: RTSP/1.0 200 OK 
04-08 15:32:46.022: I/MyHandler(7680): SETUP(2) completed with result 0 (Success) 
04-08 15:32:46.022: I/MyHandler(7680): server specified timeout of 60 secs. 
04-08 15:32:46.022: W/MyHandler(7680): Missing 'source' field in Transport response. Using RTSP endpoint address. 
04-08 15:32:46.022: W/MyHandler(7680): 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. 
04-08 15:32:46.082: I/ARTSPConnection(7680): status: RTSP/1.0 200 OK 
04-08 15:32:46.082: D/dalvikvm(13688): GC_FOR_ALLOC freed 303K, 7% free 9289K/9927K, paused 35ms, total 36ms 
04-08 15:32:46.092: I/MyHandler(7680): PLAY completed with result 0 (Success) 
04-08 15:32:46.092: I/MyHandler(7680): This is a live stream 
04-08 15:32:48.262: D/AudioHardware(7680): AudioHardware pcm playback is going to standby. 
04-08 15:32:48.262: D/AudioHardware(7680): closePcmOut_l() mPcmOpenCnt: 1 
04-08 15:32:56.092: W/MyHandler(7680): Never received any data, switching transports. 
04-08 15:32:56.112: I/ARTSPConnection(7680): status: RTSP/1.0 200 OK 
04-08 15:32:56.122: I/MyHandler(7680): TEARDOWN completed with result 0 (Success) 
04-08 15:32:56.122: I/MyHandler(7680): connection request completed with result 0 (Success) 
04-08 15:32:56.152: I/ARTSPConnection(7680): status: RTSP/1.0 200 OK 
04-08 15:32:56.152: I/MyHandler(7680): DESCRIBE completed with result 0 (Success) 
04-08 15:32:56.152: I/ASessionDescription(7680): v=0 
04-08 15:32:56.152: I/ASessionDescription(7680): o=- 15352003157473632156 15352003157473632156 IN IP4 to63-169.ifremer.fr 
04-08 15:32:56.152: I/ASessionDescription(7680): s=Unnamed 
04-08 15:32:56.152: I/ASessionDescription(7680): i=N/A 
04-08 15:32:56.152: I/ASessionDescription(7680): c=IN IP4 134.246.63.169 
04-08 15:32:56.152: I/ASessionDescription(7680): t=0 0 
04-08 15:32:56.152: I/ASessionDescription(7680): a=tool:vlc 2.0.5 
04-08 15:32:56.152: I/ASessionDescription(7680): a=recvonly 
04-08 15:32:56.152: I/ASessionDescription(7680): a=type:broadcast 
04-08 15:32:56.152: I/ASessionDescription(7680): a=charset:UTF-8 
04-08 15:32:56.152: I/ASessionDescription(7680): a=control:rtsp://134.246.63.169:5554/stream.sdp 
04-08 15:32:56.152: I/ASessionDescription(7680): m=audio 5556 RTP/AVP 96 
04-08 15:32:56.152: I/ASessionDescription(7680): b=AS:24 
04-08 15:32:56.152: I/ASessionDescription(7680): b=RR:0 
04-08 15:32:56.152: I/ASessionDescription(7680): a=rtpmap:96 mpeg4-generic/48000 
04-08 15:32:56.152: I/ASessionDescription(7680): a=fmtp:96 streamtype=5; profile-level-id=15; mode=AAC-hbr; config=118856e500; SizeLength=13; IndexLength=3; IndexDeltaLength=3; Profile=1; 
04-08 15:32:56.152: I/ASessionDescription(7680): a=control:rtsp://134.246.63.169:5554/stream.sdp/trackID=0 
04-08 15:32:56.152: I/ASessionDescription(7680): m=video 5554 RTP/AVP 96 
04-08 15:32:56.152: I/ASessionDescription(7680): b=AS:56 
04-08 15:32:56.152: I/ASessionDescription(7680): b=RR:0 
04-08 15:32:56.152: I/ASessionDescription(7680): a=rtpmap:96 H264/90000 
04-08 15:32:56.152: I/ASessionDescription(7680): a=fmtp:96 packetization-mode=1;profile-level-id=64000b;sprop-parameter-sets=Z2QAC6zZQsTv/AC0ALBAAAADAEAAAAyjxQplgA==,aOvssiw=; 
04-08 15:32:56.152: I/ASessionDescription(7680): a=control:rtsp://134.246.63.169:5554/stream.sdp/trackID=1 
04-08 15:32:56.222: I/ARTSPConnection(7680): status: RTSP/1.0 461 Unsupported transport 
04-08 15:32:56.222: I/MyHandler(7680): SETUP(1) completed with result 0 (Success) 
04-08 15:32:56.222: I/APacketSource(7680): dimensions 176x144 
04-08 15:32:56.242: I/ARTSPConnection(7680): status: RTSP/1.0 461 Unsupported transport 
04-08 15:32:56.252: I/MyHandler(7680): SETUP(2) completed with result 0 (Success) 
04-08 15:32:56.272: E/MediaPlayer(13688): error (1, -2147483648) 
04-08 15:32:56.272: E/MediaPlayer(13688): Error (1,-2147483648) 
04-08 15:32:56.272: D/VideoView(13688): Error: 1,-2147483648

我想这个问题被指出与“状态：RTSP/1.0 461不支持运输“，但我看不到我可以改变什么：我已经打开了我使用的端口，而且我确实收到了另一台计算机上的视频。

在Android手机上，我可以播放一些我在网上找到的rtsp流，例如：rtsp：//184.72.239.149/vod/mp4：BigBuckBunny_115k.mov。所以它应该是可能的。

如果有人能帮助...！

来源

2013-04-05 Tim Autin

问题是否解决？ – 2013-05-27 12:55:58

我也想开发一个显示IP摄像机直播的应用程序。我已经验证了android的videoview不支持的RSTP url。我也想记录RSTP ...你能指导我如何实现这样的功能吗？ – 2013-05-27 12:57:12

最后，这是一个网络问题，我通过MacBook WiFi共享连接我的设备，似乎它阻塞了RTSP流。现在我正在使用路由器，它在RTSP中工作（我仍然无法在Android VideoView中接收HTTP流）。不过，我仍然有一个超时问题：RTSP流在60秒后停止，因为VideoView不发送保持活动的消息。我会尝试自己做...

来源

2013-04-15 09:25:39

所以错误味精：状态：RTSP/1.0 461不支持的传输真的意味着它的网络问题？ – Robert 2013-12-17 07:56:23

我用openRTSP命令测试了我的rtsp服务器。

这是UDP端口被阻塞。

如果访问RTSP没有-t：

-> $ openRTSP <rtsp_url>

我日志告诉我：

// omit lots of lines.. 
Created receiver for "video/H264" subsession (client ports 63346-63347) 
Sending request: SETUP rtsp://61.218.52.250:554/live/ch00_0/trackID=0 RTSP/1.0 
CSeq: 4 
User-Agent: openRTSP (LIVE555 Streaming Media v2013.12.16) 
Transport: RTP/AVP;unicast;client_port=63346-63347 

Received 47 new bytes of response data. 
Received a complete SETUP response: 
RTSP/1.0 461 Unsupported Transport 
CSeq: 4 

Failed to setup "video/H264" subsession: 461 Unsupported Transport

因此改变与对TCP：

-> $ openRTSP -t <rtsp_url>

它启动成功接收数据。

// omit lots of lines.. 
Opened URL "rtsp://61.218.52.250:554/live/ch00_0", returning a SDP description: 
v=0 
o=- 1 1 IN IP4 127.0.0.1 
s=Ubiquiti Live 
i=UBNT Streaming Media 
c=IN IP4 0.0.0.0 
t=0 0 
m=video 0 RTP/AVP 99 
b=AS:50000 
a=framerate:25 
a=x-dimensions:1280,720 
a=x-vendor-id:ubnt,a521 
a=x-rtp-ts:4617405454576779984 
a=rtpmap:99 H264/90000 
a=fmtp:99 profile-level-id=42A01E;packetization-mode=1;sprop-parameter-sets=Z0IAKOkAoAt1xIAG3dAAzf5gDYgQlA==,aM4xUg== 
a=control:trackID=0 

Sending request: SETUP rtsp://61.218.52.250:554/live/ch00_0/trackID=0 RTSP/1.0 
CSeq: 4 
User-Agent: openRTSP (LIVE555 Streaming Media v2013.12.16) 
Transport: RTP/AVP/TCP;unicast;interleaved=0-1 


Received 107 new bytes of response data. 
Received a complete SETUP response: 
RTSP/1.0 200 OK 
CSeq: 4 
Transport: RTP/AVP/TCP;unicast;interleaved=0-1 
Session: E090B5503236A1BFB7CE 


Setup "video/H264" subsession (client ports 54884-54885) 
Sending request: PLAY rtsp://61.218.52.250:554/live/ch00_0/ RTSP/1.0 
CSeq: 5 
User-Agent: openRTSP (LIVE555 Streaming Media v2013.12.16) 
Session: E090B5503236A1BFB7CE 
Range: npt=0.000- 


Received 159 new bytes of response data. 
Received a complete PLAY response: 
RTSP/1.0 200 OK 
CSeq: 5 
Session: E090B5503236A1BFB7CE 
Range: npt=now- 
RTP-Info: url=rtsp://61.218.52.250:554/live/ch00_0//trackID=0;seq=41402;rtptime=0 


Started playing session 
Data is being streamed (signal with "kill -HUP 96432" or "kill -USR1 96432" to terminate)... 
Received 47 new bytes of response data. 
Received 1424 new bytes of response data. 
Received 1424 new bytes of response data. 
Received 1424 new bytes of response data. 
Received 1424 new bytes of response data. 
Received 1448 new bytes of response data. 
Received 1448 new bytes of response data.

Ref to openRTSP basics.

现在我得如何自动切换找出该TCP的Android系统。

来源

2013-12-17 08:48:00 Robert

请尝试VLC：

VLC some_file.mp4 -I HTTP --sout“#transcode {soverlay，AB = 128，采样率= 44100，通道= 2，acodec = MP4A，了vcodec = H264，宽度= 480，高度= 270，VFILTER = “画布{宽度= 480，高度= 270，纵横= 16：9}”，FPS = 25，VB = 800，VENC = X264 {水平= 12，无CABAC，次亚= 20，threads = 4，bframes = 0，min-keyint = 1，keyint = 50}}：gather：rtp {mp4a-latm，sdp = rtsp：//0.0.0.0:5554/stream.sdp}“

和Android代码：

@Override 
    protected void onCreate(Bundle savedInstanceState) { 
     super.onCreate(savedInstanceState); 
     setContentView(R.layout.activity_main); 

     final VideoView vidView = (VideoView)findViewById(R.id.myVideo); 

     MediaController vidControl = new MediaController(this); 
     vidControl.setAnchorView(vidView); 
     vidView.setMediaController(vidControl); 

     vidView.setVideoPath("rtsp://137.110.92.231:5554/stream.sdp"); 

     vidView.start(); 
     }

来源

2015-01-04 04:02:26

使用的MediaPlayer它支持HTTP和RTSP网络协议。 http://developer.android.com/guide/topics/media/mediaplayer.html#mediaplayer http://developer.android.com/guide/appendix/media-formats.html#recommendations

来源

2015-08-06 05:01:29

VLC RTSP直播到Android

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