我正在尝试将音频流从A点发送到Kurento Media Server,并在B点接收该音频流,并带有GSTREAMER。我要实现的目标应该是这样:
(点A(----与GSTREAMER一起发送音频流--->(Kurento(----音频流----->(点B( - - !
到目前为止,我写了代码:
function createOutGoingAudioStream() {
var sdpOffer = " v=0rn"
+ "o=- 0 0 IN IP4 0.0.0.0rn"
+ "c=IN IP4 0.0.0.0rn"
+ "t=0 0rn"
+ "m=audio 5005 RTP/AVP 0rn"
+ "a=rtpmap:0 PCMU/8000rn";
var pipeline;
console.log();
console.log("Starting Audio Stream from Command Post.....");
// get kurento client
getKurentoClient(function(error, kurentoClient) {
if (error) {
return callback(error);
}
// create media pipe line
kurentoClient.create('MediaPipeline', function(error, pipeline) {
if (error) {
return callback(error);
}
// create first rtpEndpoint for the incoming audio stream
pipeline.create('RtpEndpoint', function(error, rtpEndpoint) {
if (error) {
pipeline.release();
return callback(error);
}
console.log('audio RTP Endpoint created successfully!');
rtpEndpoint.processOffer(sdpOffer, function(error, sdpAnswer) {
if (error) {
pipeline.release();
return callback(error);
}
console.log(sdpAnswer);
console.log();
// Start a gstreamer audio stream over the audio port that we got from the kurento server
var jsonSdpAnswer = transform.parse(sdpAnswer);
var port = jsonSdpAnswer.media[0].port;
console.log("Starting audio stream to the kurento server: ");
console.log('sh gstreamer.sh ' + port + ' > log.txt')
exec('sh gstreamer.sh ' + port + ' > log.txt', function(err, stdout, stderr) {
if (err) {
console.error(err);
return;
}
//if all is ok nothing wil prompt to the console
console.log(stdout);
});
});
// create second rtpEndpoint for the outgoing to the odroid's audio stream
pipeline.create('RtpEndpoint', function(error, outRtpEndpoint) {
if (error) {
pipeline.release();
return callback(error);
}
console.log('second RTP Endpoint created successfully!');
rtpEndpoint.connect(outRtpEndpoint, function(error){
if(error) return onError(error);
});
outRtpEndpoint.generateOffer(function(error,offerSdp){
if(error) return onError(error);
console.log("@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@")
console.log(offerSdp);
});
});
});
});
});
}
我从kurento服务器获得untpendpoint的SDPOFFER,看起来像这样:
SDP优惠
为了聆听该流而不想工作,我要做什么都没关系。我究竟做错了什么 ?
我真的很感谢任何帮助。
谢谢!
我能够解决这个问题。我从点A的浏览器(Webrtcendpoint(获得了源音频流,并将该端点连接到rtpendpoint,从那里我将流到PointB。(点A -Web浏览器 -> WebrtcendPoint( ->(Kurento-> rtpendpoint( ->(点B- ffplay(。
function createOutGoingAudioStream(sessionId,ws, sdpOffer, callback){
if (!sessionId) {
return callback('Cannot use undefined sessionId');
}
getKurentoClient(function(error, kurentoClient) {
if (error) {
return callback(error);
}
kurentoClient.create('MediaPipeline', function(error, pipeline) {
if (error) {
return callback(error);
}
createMediaElements(pipeline, ws, function(error, webRtcEndpoint) {
if (error) {
pipeline.release();
return callback(error);
}
if (candidatesQueue[sessionId]) {
while(candidatesQueue[sessionId].length) {
var candidate = candidatesQueue[sessionId].shift();
webRtcEndpoint.addIceCandidate(candidate);
}
}
connectMediaElements(webRtcEndpoint, function(error) {
if (error) {
pipeline.release();
return callback(error);
}
webRtcEndpoint.on('OnIceCandidate', function(event) {
var candidate = kurento.getComplexType('IceCandidate')(event.candidate);
ws.send(JSON.stringify({
id : 'iceCandidate',
candidate : candidate
}));
});
webRtcEndpoint.processOffer(sdpOffer, function(error, sdpAnswer) {
if (error) {
pipeline.release();
return callback(error);
}
sessions[sessionId] = {
'pipeline' : pipeline,
'webRtcEndpoint' : webRtcEndpoint
}
return callback(null, sdpAnswer);
});
webRtcEndpoint.gatherCandidates(function(error) {
if (error) {
return callback(error);
}
var sdp = 'v=0';
sdp += 'nc=IN IP4 IP_WHERE_YOU_WANT_TO_GET_THE_STREAM'; // this is the ip where the kurento should stream to
sdp += 'nm=audio 8080 RTP/AVP 0'; // at port 8080 you will have an audio stream
sdp += 'na=rtpmap:0 PCMU/8000';
sdp += 'nm=video 9090 RTP/AVP 101'; // at port 9090 you will have a video stream
sdp += 'na=rtpmap:101 H264/90000';
pipeline.create('RtpEndpoint', function(err, rtpEndpoint){
rtpEndpoint.processOffer(sdp, function(error, sdpAnswer) {
if (error) {
return callback(error);
}
console.log("################################################");
console.log(sdpAnswer);
console.log("################################################");
});
webRtcEndpoint.connect(rtpEndpoint, function(err, rtpEndpoint) { if(err) { console.log("Error!"); } });
});
});
});
});
});
});
}
在您流到的计算机上,您可以使用以下方式收听流
ffplay rtp://IP_FROM_THE_SDP_OFFER_IN_THE_CODE_ABOVE:AUDIO_PORT_FROM_THE_SDP_OFFER_FROM_THE_CODE_ABOVE