[英]WebRTC java server trouble
我想我非常接近要讓Java服務器應用程序通過WebRTC與瀏覽器頁面對話,但是我不能完全使它正常工作。 我覺得自己缺少一些小東西,因此希望這里有人可以提出建議。
我仔細研究了WebRTC示例-Java單元測試( org.webrtc.PeerConnectionTest
)和示例Android應用程序( trunk/talk/examples/android
)。 根據所學知識,我編寫了一個Java應用程序,該應用程序使用WebSockets進行信號傳輸並嘗試將視頻流發送到Chrome。
問題是,即使我所有的代碼(Javascript和Java)都按我期望的順序執行,並擊中了所有正確的日志記錄語句,瀏覽器中也沒有視頻。 本地libjingle代碼在控制台日志中有一些可疑的輸出,但是我不確定該怎么做。 我在日志中用以下“ >>”突出顯示了可疑行。 例如,似乎視頻端口分配器在創建后不久就被銷毀了,因此顯然有些錯誤。 另外,“ Changing video state, recv=1 send=0
”似乎也不正確,因為Java端應該發送視頻,而不是接收視頻。...也許我在濫用OfferToReceiveVideo
選項?
如果查看下面的日志,您會發現與瀏覽器的WebSocket通信正常運行,並且能夠將SDP報價成功發送到瀏覽器並從瀏覽器收到SDP答復。 在PeerConnections上設置本地和遠程描述似乎也可以正常工作。 HTML5視頻元素將源設置為BLOB url,這是應該的。 那么,我可能會缺少什么呢? 即使我的客戶端和服務器現在位於同一台計算機上,我也需要對ICE候選者做任何事情嗎?
任何建議將不勝感激!
1.134: Java Offer:
v=0
o=- 5893945934600346864 2 IN IP4 127.0.0.1
s=-
t=0 0
a=group:BUNDLE video
a=msid-semantic: WMS JavaMediaStream
m=video 1 RTP/SAVPF 100 116 117
c=IN IP4 0.0.0.0
a=rtcp:1 IN IP4 0.0.0.0
a=ice-ufrag:dJxTlMlXy7uASrDU
a=ice-pwd:r8BRkXVnc4dqCABUDhuRjpp7
a=ice-options:google-ice
a=mid:video
a=extmap:2 urn:ietf:params:rtp-hdrext:toffset
a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
a=sendrecv
a=rtcp-mux
a=crypto:0 AES_CM_128_HMAC_SHA1_80 inline:yq6wOHhk/QfsWuh+1oOEqfB4GjKZzz8XfQnGCDP3
a=rtpmap:100 VP8/90000
a=rtcp-fb:100 ccm fir
a=rtcp-fb:100 nack
a=rtcp-fb:100 goog-remb
a=rtpmap:116 red/90000
a=rtpmap:117 ulpfec/90000
a=ssrc:3720473526 cname:nul6R21KmwAms3Ge
a=ssrc:3720473526 msid:JavaMediaStream JavaMediaStream_v0
a=ssrc:3720473526 mslabel:JavaMediaStream
a=ssrc:3720473526 label:JavaMediaStream_v0
1.149: Received remote stream
1.150: Browsers Answer:
v=0
o=- 4261396844048664099 2 IN IP4 127.0.0.1
s=-
t=0 0
a=group:BUNDLE video
a=msid-semantic: WMS
m=video 1 RTP/SAVPF 100 116 117
c=IN IP4 0.0.0.0
a=rtcp:1 IN IP4 0.0.0.0
a=ice-ufrag:quzQNsX+ZlUWUQqV
a=ice-pwd:y5A0+7sM8P88AatBLd1fdd5G
a=mid:video
a=extmap:2 urn:ietf:params:rtp-hdrext:toffset
a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
a=recvonly
a=rtcp-mux
a=crypto:0 AES_CM_128_HMAC_SHA1_80 inline:WClNA69OfpjdJy3Bv4ujejk/IYnn4DW8kjrB18xP
a=rtpmap:100 VP8/90000
a=rtcp-fb:100 ccm fir
a=rtcp-fb:100 nack
a=rtcp-fb:100 goog-remb
a=rtpmap:116 red/90000
a=rtpmap:117 ulpfec/90000
對我來說這沒關系。 Java的報價包括我的視頻流。
(帶有“ >>”標記的可疑行)
Camera '/dev/video0' started with format YUY2 640x480x30, elapsed time 59 ms
Ignored line: c=IN IP4 0.0.0.0
NACK enabled for channel 0
NACK enabled for channel 0
Created channel for video
Jingle:Channel[video|1|__]: NULL DTLS identity supplied. Not doing DTLS
Jingle:Channel[video|2|__]: NULL DTLS identity supplied. Not doing DTLS
Session:5893945934600346864 Old state:STATE_INIT New state:STATE_SENTINITIATE Type:urn:xmpp:jingle:apps:rtp:1 Transport:http://www.google.com/transport/p2p
Setting local video description
AddSendStream {id:JavaMediaStream_v0;ssrcs:[3720473526];ssrc_groups:;cname:nul6R21KmwAms3Ge;sync_label:JavaMediaStream}
Add send ssrc: 3720473526
>> Warning(webrtcvideoengine.cc:2704): SetReceiverBufferingMode(0, 0) failed, err=12606
Changing video state, recv=0 send=0
Transport: video, allocating candidates
Transport: video, allocating candidates
Jingle:Net[eth0:192.168.0.0/24]: Allocation Phase=Udp
Jingle:Port[:1:0::Net[eth0:192.168.0.0/24]]: Port created
Adding allocated port for video
Jingle:Port[video:1:0::Net[eth0:192.168.0.0/24]]: Added port to allocator
Jingle:Net[tun0:192.168.128.6/32]: Allocation Phase=Udp
Jingle:Port[:1:0::Net[tun0:192.168.128.6/32]]: Port created
Adding allocated port for video
Jingle:Port[video:1:0::Net[tun0:192.168.128.6/32]]: Added port to allocator
Ignored line: c=IN IP4 0.0.0.0
Warning(webrtcvideoengine.cc:2309): GetStats: sender information not ready.
Jingle:Channel[video|1|__]: Other side didn't support DTLS.
Jingle:Channel[video|2|__]: Other side didn't support DTLS.
Enabling BUNDLE, bundling onto transport: video
Channel enabled
>> Changing video state, recv=1 send=0
Session:5893945934600346864 Old state:STATE_SENTINITIATE New state:STATE_RECEIVEDACCEPT Type:urn:xmpp:jingle:apps:rtp:1 Transport:http://www.google.com/transport/p2p
Setting remote video description
Hybrid NACK/FEC enabled for channel 0
Hybrid NACK/FEC enabled for channel 0
SetSendCodecs() : selected video codec VP8/1280x720x30fps@2000kbps (min=50kbps, start=300kbps)
Video max quantization: 56
VP8 number of temporal layers: 1
VP8 options : picture loss indication = 0, feedback mode = 0, complexity = normal, resilience = off, denoising = 0, error concealment = 0, automatic resize = 0, frame dropping = 1, key frame interval = 3000
WARNING: no real random source present!
SRTP activated with negotiated parameters: send cipher_suite AES_CM_128_HMAC_SHA1_80 recv cipher_suite AES_CM_128_HMAC_SHA1_80
Changing video state, recv=1 send=0
Session:5893945934600346864 Old state:STATE_RECEIVEDACCEPT New state:STATE_INPROGRESS Type:urn:xmpp:jingle:apps:rtp:1 Transport:http://www.google.com/transport/p2p
Jingle:Net[eth0:192.168.0.0/24]: Allocation Phase=Relay
Jingle:Net[tun0:192.168.128.6/32]: Allocation Phase=Relay
Jingle:Net[eth0:192.168.0.0/24]: Allocation Phase=Tcp
Jingle:Port[:1:0:local:Net[eth0:192.168.0.0/24]]: Port created
Adding allocated port for video
Jingle:Port[video:1:0:local:Net[eth0:192.168.0.0/24]]: Added port to allocator
Jingle:Net[tun0:192.168.128.6/32]: Allocation Phase=Tcp
Jingle:Port[:1:0:local:Net[tun0:192.168.128.6/32]]: Port created
Adding allocated port for video
Jingle:Port[video:1:0:local:Net[tun0:192.168.128.6/32]]: Added port to allocator
Jingle:Net[eth0:192.168.0.0/24]: Allocation Phase=SslTcp
Jingle:Net[tun0:192.168.128.6/32]: Allocation Phase=SslTcp
All candidates gathered for video:1:0
Transport: video, component 1 allocation complete
Transport: video allocation complete
Candidate gathering is complete.
Capture delay changed to 120 ms
Captured frame size 640x480. Expected format YUY2 640x480x30
Capture size changed : selected video codec VP8/640x480x30fps@2000kbps (min=50kbps, start=300kbps)
Video max quantization: 56
VP8 number of temporal layers: 1
VP8 options : picture loss indication = 0, feedback mode = 0, complexity = normal, resilience = off, denoising = 0, error concealment = 0, automatic resize = 1, frame dropping = 1, key frame interval = 3000
VAdapt Frame: 0 / 300 Changes: 0 Input: 640x480 Scale: 1 Output: 640x480 Changed: false
>> Jingle:Port[video:1:0::Net[eth0:192.168.0.0/24]]: Port deleted
>> Jingle:Port[video:1:0::Net[eth0:192.168.0.0/24]]: Removed port from allocator (3 remaining)
Removed port from p2p socket: 3 remaining
Jingle:Port[video:1:0::Net[tun0:192.168.128.6/32]]: Port deleted
Jingle:Port[video:1:0::Net[tun0:192.168.128.6/32]]: Removed port from allocator (2 remaining)
Removed port from p2p socket: 2 remaining
>> Jingle:Port[video:1:0:local:Net[eth0:192.168.0.0/24]]: Port deleted
>> Jingle:Port[video:1:0:local:Net[eth0:192.168.0.0/24]]: Removed port from allocator (1 remaining)
Removed port from p2p socket: 1 remaining
Jingle:Port[video:1:0:local:Net[tun0:192.168.128.6/32]]: Port deleted
Jingle:Port[video:1:0:local:Net[tun0:192.168.128.6/32]]: Removed port from allocator (0 remaining)
Removed port from p2p socket: 0 remaining
<html lang="en">
<head>
<title>Web Socket Signalling</title>
<link rel="stylesheet" href="css/socket.css">
<script src="js/socket.js"></script>
</head>
<body>
<h2>Repsonse from Server</h2>
<textarea id="responseText"></textarea>
<h2>Video</h2>
<video id="remoteVideo" autoplay></video>
</body>
</html>
(function() {
var remotePeerConnection;
var sdpConstraints = {
'mandatory' : {
'OfferToReceiveAudio' : false,
'OfferToReceiveVideo' : true
}
};
var Sock = function() {
var socket;
if (!window.WebSocket) {
window.WebSocket = window.MozWebSocket;
}
if (window.WebSocket) {
socket = new WebSocket("ws://localhost:8080/websocket");
socket.onopen = onopen;
socket.onmessage = onmessage;
socket.onclose = onclose;
} else {
alert("Your browser does not support Web Socket.");
}
function onopen(event) {
getTextAreaElement().value = "Web Socket opened!";
}
function onmessage(event) {
appendTextArea(event.data);
sdpOffer = new RTCSessionDescription(JSON.parse(event.data));
remotePeerConnection = new webkitRTCPeerConnection(null);
remotePeerConnection.onaddstream = gotRemoteStream;
trace("Java Offer: \n" + sdpOffer.sdp);
remotePeerConnection.setRemoteDescription(sdpOffer);
remotePeerConnection.createAnswer(gotRemoteDescription, onCreateSessionDescriptionError, sdpConstraints);
}
function onCreateSessionDescriptionError(error) {
console.log('Failed to create session description: '
+ error.toString());
}
function gotRemoteDescription(answer) {
remotePeerConnection.setLocalDescription(answer);
trace("Browser's Answer: \n" + answer.sdp);
socket.send(JSON.stringify(answer));
}
function gotRemoteStream(event) {
var remoteVideo = document.getElementById("remoteVideo");
remoteVideo.src = URL.createObjectURL(event.stream);
trace("Received remote stream");
}
function onclose(event) {
appendTextArea("Web Socket closed");
}
function appendTextArea(newData) {
var el = getTextAreaElement();
el.value = el.value + '\n' + newData;
}
function getTextAreaElement() {
return document.getElementById('responseText');
}
function trace(text) {
console.log((performance.now() / 1000).toFixed(3) + ": " + text);
}
}
window.addEventListener('load', function() {
new Sock();
}, false);
})();
public class PeerConnectionManager {
/**
* Called when the WebSocket handshake is completed
*/
public void createOffer() {
peerConnection = factory.createPeerConnection(
new ArrayList<PeerConnection.IceServer>(),
new MediaConstraints(),
new PeerConnectionObserverImpl());
// Get the video source
videoSource = factory.createVideoSource(VideoCapturer.create(""), new MediaConstraints());
// Create a MediaStream with one video track
MediaStream lMS = factory.createLocalMediaStream("JavaMediaStream");
VideoTrack videoTrack = factory.createVideoTrack("JavaMediaStream_v0", videoSource);
videoTrack.addRenderer(new VideoRenderer(new VideoRendererObserverImpl()));
lMS.addTrack(videoTrack);
peerConnection.addStream(lMS, new MediaConstraints());
// We don't want to receive anything
MediaConstraints sdpConstraints = new MediaConstraints();
sdpConstraints.mandatory.add(new MediaConstraints.KeyValuePair(
"OfferToReceiveAudio", "false"));
sdpConstraints.mandatory.add(new MediaConstraints.KeyValuePair(
"OfferToReceiveVideo", "false"));
// Get the Offer SDP
SdpObserverImpl sdpOfferObserver = new SdpObserverImpl();
peerConnection.createOffer(sdpOfferObserver, sdpConstraints);
SessionDescription offerSdp = sdpOfferObserver.getSdp();
// Set local SDP, don't care for any callbacks
peerConnection.setLocalDescription(new SdpObserverImpl(), offerSdp);
// Serialize Offer and send to the Browser via a WebSocket
JSONObject offerSdpJson = new JSONObject();
offerSdpJson.put("sdp", offerSdp.description);
offerSdpJson.put("type", offerSdp.type.canonicalForm());
webSocketContext.channel().writeAndFlush(
new TextWebSocketFrame(offerSdpJson.toString()));
}
/**
* Called when an SDP Answer arrives via the WebSocket
*/
public void setRemoteDescription(SessionDescription answer) {
peerConnection.setRemoteDescription( new SdpObserverImpl(), answer);
}
}
啊。 沒關系。 很抱歉這個愚蠢的問題。
缺少的部分是瀏覽器和Java服務器之間的ICE候選者交換。 現在,我添加了通過WebSocket進行ICE協商的代碼,一切正常!
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