This article show the different ways to make calls to Wowza Media Engine from external applications and environments for various purposes such as getting server status , listeners , connections , applications and its streams etc .
HTTP Providers
HTTP Providers are Java classes that are configured on a per-virtual host basis.
Some pre packaged HTTP providers that return data in XML :
1. HTTPConnectionCountsXML
Returns connection information like Vhost , application , application instance , message in bytes rate , message out byte rates etc.
Each HTTP provider can be configured with different request filter and authentication method ( none , basic , digest). We can even create our own substitutes for the HTTP providers as defined in the next section .
Extending HTTProvider2Base
The following code snippet describes the process of creating a Wowza Web services that return a json containing all the values .
Imports to build a HTTP provider
import com.wowza.wms.application.*;
import com.wowza.wms.vhost.*;
import com.wowza.wms.http.*;
import com.wowza.wms.httpstreamer.model.*;
//since we want to return in json format
import org.json.simple.JSONObject;
The class declaration is as folllows
public class DCWS extends HTTProvider2Base
{
....
}
The code to extract application names
public JSONObject listChannels(){
JSONObject obj=new JSONObject();
//get params from virtual host and iterate through it
List<String> vhostNames = VHostSingleton.getVHostNames();
Iterator<String> iter = vhostNames.iterator();
while (iter.hasNext())
{
String vhostName = iter.next();
IVHost vhost = (IVHost)VHostSingleton.getInstance(vhostName);
List<String> appNames = vhost.getApplicationNames();
Iterator<String> appNameIterator = appNames.iterator();
int i=0;
while (appNameIterator.hasNext())
{
String applicationName = appNameIterator.next();
try {
String key = "channel"+ (++i);
obj.put(key, URLEncoder.encode(applicationName, "UTF-8"));
}
catch (UnsupportedEncodingException e) {
e.printStackTrace();
}
}
}
return obj;
}
this article is focused on Wowza RTMP Authentication with Third party Token provider over Tiny Encryption Algorithm (TEA) and is a continuation of the previous post about setting up a basic RTMP Authentication module on Wowza Engine above version 4.
The task is divided into 3 parts .
RTMP Encoder Application
Wowza RTMP Auth module
Third party Authentication Server
The component diagram is as follows :
The detailed explanation of the components are :
1.Wowza RTMP Auth module
The Wowza Server receives a rtmp stream url in the format as :
It considers the username and pass to be user credentials . RTMP auth Module invokes the getPassword() function inside of deployed application class passing the username as parameter. The username is then encrypted using TEA ( Tiny Encryption algorithm)
TEA is a block cipher which is based on symmetric ( private) key encryption . Input is a 64 bit of plain or cipher text with a 128 bit key resulting in output of cipher or plain text respectively.
The code for encryption is
TEA.encrypt( username, sharedSecret );
The code to make a connection to third party auth server is
url = new URL(serverTokenValidatorURL);
URLConnection connection;
connection = url.openConnection();
connection.setDoOutput(true);
OutputStreamWriter out = new OutputStreamWriter(connection.getOutputStream());
out.write("clientid=" + TEA.encrypt( username, sharedSecret ););
out.close();
The sharedsecret is the common key which is with both the Auth server and wowza server . It must be atleast a 16 digit alphanumeric / special character based key . An example of shared secret is abcdefghijklmnop .The value can be stored as property in Application.xml file.
The 3rd party Auth server stores the passwords for users or performs oauth based authentication . It uses a shared secret key to decrypt the token based on TEA as explained in above section .
The code to decrypt the incoming clientId
TEA.decrypt(id, sharedSecret);
Add own custom logic to check files , databases etc for obtaining the password corresponding to the username as decrypted above.
The code to encrypt the password for the user if exists or send invalid response if non exists is
To secure the publishers for a common application through username -password specific for stream names , this post is useful . It uses Module Core Security to prompt back the user for supplying credentials.
The detailed code to check the rtmp query-string for parameters and performs the checks – is user is allowed to connect and is user allowed to stream on given stream name is given below .
Initialize the hashmap containing publisher clients and IapplicationInstance
On app start initilaize the IapplicationInstance object .
public void onAppStart(IApplicationInstance appInstance)
{
this.appInstance = appInstance;
}
Onconnect is called called when any publisher tries to connects with media server. At this event collect the username and clientId from the client. Check if publisherclient contains the userName which client has provided else reject the connection .
AMFDataItem: class for marshalling data between Wowza Pro server and Flash client.
As the event user starts to publish a stream after sucessful connection Onpublishing function is called . It extracts the stream name from the client ( function extractStreamName() )and checks if user is allowed to stream on the given streamname (function isStreamNotAllowed()) .
public void publish(IClient client, RequestFunction function, AMFDataList params)
{
String streamName = extractStreamName(client, function, params);
if (isStreamNotAllowed(client, streamName))
{
sendClientOnStatusError(client, NetStream.Publish.Denied, "Stream name not allowed for the logged in user: "+streamName);
client.rejectConnection();
}
else{
invokePrevious(client, function, params);
}
}
Function when publisher disconnects from server . It removes the client from publisherClients.
public void onDisconnect(IClient client)
{
if(this.publisherClients!=null){
this.publisherClients.remove(client.getClientId());
}
}
To purpose of the article is the use the RTMP Authentication Module in wowza Engine . This will enable us to intercept a connect request with username and password to be checked from any outside source like – database , password file , third party token provider , third party oauth etc. Once the password provided by user is verified with the authentic password form external sources the user is allowed to connect and publish.
Step 1 : Create a new Wowza Media Server Project in Eclipse . It is assumed that user has already integrated WowzaIDE into eclipse .
File -> New -> Wowza Media Server Project
Step 2: Give any project name . I named it as “RTMPAuthSampleCode”.
wowza RTMP Auth
Step 3 : Point the location to existing Wowza Engine installed in local environment .
It is usually in /usr/local/WowzaStreamingEngine/
Wowza RTMP Auth
Step 4 : Proceed with the creation , uncheck the event methods as we are not using them right now .
Step 5: Put the code in class.
The class RTMPAuthSampleCode extends AuthenticateUsernamePasswordProviderBase . Its mandatory to define getPassword(String username ) and userExists(String username). ModuleRTMPAuthenticate will invoke getPassword for connection request from users .
We can add any source of obtaining password for a given username which will be matched to the password supplied by user . If it matches he will be granted access otherwise we can return null or error message .
We may use various ways of obtaining user credentials like databse , password files , third part token provider etc . I will be discussing more ways to do RTMP authenticate esp using a third part token provider which using TEA.encrypt and shared secret in the next blog.
Step 6: Build the project and Run.
Project-> Build the Project
Run -> Run Configurations … -> WowzaMediaServer_RTMPAuthSampleCode
To modules in my ubuntu 64 bit version 14.04 system , I also need to provide
-Dcom.wowza.wms.native.base=”linux” inside of the VM Arguments . Its highlighted in figure below.
Step 7: Click Run to start the wowza Media Engine
Step 8 : Open the Manager Console of Wowza.
web based GUI interface of managing the application and checking for incoming streams . The manager script can be started with
Step 12 : Make Authentication.xml file inside /usr/local/WowzaStreamingEngine/conf folder.
Note that from wowza 4 and later versions the Authentiocation.xml has come bundled with wms-server.jar which is inside of lib folder . However for me , without giving a explicit Authentication.xml file the program froze and using my own simple authentication.xml gave problems with the digest . Hence follow the below process to get a working Authentication.xml file inside conf folder
Expand the archive and inside the extracted folder wms-server copy the file from location wms-server/com/wowza/wms/conf/Authentication.xml to /usr/local/WowzaStreamingEngine/conf.
Step 13 : Restart Wowza Media Engine .
Step 14 : Use any RTMP encoder as Adobe Live Media Encoder or Gocoder or your own app ( could not use this with ffmpeg ) and try to connect to application RTMPAuthSampleCode with username test and password 1234.
Step 15 : Observer the logs for incoming streams and traces from getpassword .
If you want the user test to have permission to publish stream to this application then return 1234 from getPassword else return null .
This blog is in continuation to the attempts / outcomes and problems in building a WebRTC to RTP media framework that successfully stream / broadcast WebRTC content to non webrtc supported browsers ( safari / IE ) / media players ( VLC ).
Attempt 4: Stream the content to a WebRTC endpoint which is hidden in a video call . Pick the stream from vp8 object URL send to a streaming server
This process involved the following components :
WebRTC API : simplewebrtc on Chrome
Transfer mechanism from client to Streaming server: webrtc media channel
Problems : No streaming server is qualified to handle a direct webrtc input and stream it on network .
Attempt 4.1 : Stream the content to a WebRTC endpoint . Do WebRTC Endpoint to RTP Endpoint bridge using Kurento APIs.
Use the RTP port and ip address to input into a ffmpeg or gstreamer or VLC terminal command and out put a live H264 stream on another ip and port address .
This process involved the following components :
API : Kurento
Transfer mechanism : HTML5 webrtc client -> application server hosting java -> media server -> application for webrtc media to RTP media conversation -> RTP player
Screenshots of attempts with Wowza to stream RTP from a IP and port
Problems : The stream was black which means 100% loss.
Lesson learned : RTP is not suitable for over the intgernet transmission especially with firewalls
Attempt 4.2 : Build a WebRTC Endpoint to Http endpoint in kurento and force the video audio encoding to be that of H264 and PCMU.
Code snippet for adding constraints to output media via pipeline and forcing choice of codecs( H264 for video and PCMU for audio ).
MediaPipeline pipeline = kurento.createMediaPipeline();
WebRtcEndpoint webRtcEndpoint = new WebRtcEndpoint.Builder(pipeline).build();
HttpGetEndpoint httpEndpoint=new HttpGetEndpoint.Builder(pipeline).build();
org.kurento.client.Fraction fr= new org.kurento.client.Fraction(1, 30);
VideoCaps vc= new VideoCaps(VideoCodec.H264,fr);
httpEndpoint.setVideoFormat(vc);
AudioCaps ac= new AudioCaps(AudioCodec.PCMU, 65536);
httpEndpoint.setAudioFormat(ac);
webRtcEndpoint.connect(httpEndpoint);
Alternatively one can opt to use gstreamer filter to force the output in raw format.
// basic media operation of 1 pipeline and 2 endpoints
MediaPipeline pipeline = kurento.createMediaPipeline();
WebRtcEndpoint webRtcEndpoint = new WebRtcEndpoint.Builder(pipeline).build();
RtpEndpoint rtpEndpoint = new RtpEndpoint.Builder(pipeline).build();
// adding Gstream filters
GStreamerFilter filter1 = new GStreamerFilter.Builder(pipeline, "videorate max-rate=30").withFilterType(FilterType.VIDEO).build();
GStreamerFilter filter2 = new GStreamerFilter.Builder(pipeline, "capsfilter caps=video/x-h264,width=1280,height=720,framerate=30/1").withFilterType(FilterType.VIDEO).build();
GStreamerFilter filter3 = new GStreamerFilter.Builder(pipeline, "capsfilter caps=audio/x-mpeg,layer=3,rate=48000").withFilterType(FilterType.AUDIO).build();
// connecting all poin ts to one another
webRtcEndpoint.connect (filter1);
filter1.connect (filter2);
filter2.connect (filter3);
filter3.connect (rtpEndpoint);
// RTP SDP offer and answer
String requestRTPsdp = rtpEndpoint.generateOffer();
rtpEndpoint.processAnswer(requestRTPsdp);
End result : The output is still webm based and doesnt work on h264 clients.
Attempt 5 : Use a RTP SDP Endpoint ( ie a SDP file valid for a given session ) and use it to play the WebRTC media over Wowza streaming server
This process involved the following components
WebRTC Stream and object URL of the blob containing VP8 media
Kurento WebRTC Endpoint bridge to generate SDP
Wowza Streaming server
Snippet used for kurento to generate a SDP file from WebRTC to RTP bridge
@RequestMapping(value = "/rtpsdp", method = RequestMethod.POST)
private String processRequestrtpsdp(@RequestBody String sdpOffer)
throws IOException, URISyntaxException, InterruptedException {
//basic media operation of 1 pipeline and 2 endpoinst
MediaPipeline pipeline = kurento.createMediaPipeline();
WebRtcEndpoint webRtcEndpoint = new WebRtcEndpoint.Builder(pipeline).build();
RtpEndpoint rtpEndpoint = new RtpEndpoint.Builder(pipeline).build();
//connecting all poin ts to one another
webRtcEndpoint.connect (rtpEndpoint);
// RTP SDP offer and answer
String requestRTPsdp = rtpEndpoint.generateOffer();
rtpEndpoint.processAnswer(requestRTPsdp);
// write the SDP conector to an external file
PrintWriter out = new PrintWriter("/tmp/test.sdp");
out.println(requestRTPsdp);
out.close();
HttpGetEndpoint httpEndpoint = new HttpGetEndpoint.Builder(pipeline).build();
PlayerEndpoint player = new PlayerEndpoint.Builder(pipeline, requestRTPsdp).build();
httpEndpoint.connect(rtpEndpoint);
player.connect(httpEndpoint);
// Playing media and opening the default desktop browser
player.play();
String videoUrl = httpEndpoint.getUrl();
System.out.println(" ------- video URL -------------"+ videoUrl);
// send the response to front client
String responseSdp = webRtcEndpoint.processOffer(sdpOffer);
return responseSdp;
}
End result : wowza doesnt not recognize the WebRTC SDP and play the video
screenshot of wowza with SDP input
Attempt 5.1 : Use a RTP SDP Endpoint ( ie a SDP file valid for a given session ) and use it to play the WebRTC media over Default Ubuntu media player
End result : wowza doesnt not recognize the WebRTC SDP and play the video : deformed media
screenshot of playing from a SDP file
Attempt 5.2 : Use a RTP SDP Endpoint ( ie a SDP file valid for a given session ) and use it to play the WebRTC media over VLC using socket input
End result : nothing plays
screenshot of VLC connected to play from socket and failure to play anything
Attempt 5.3: Create a WebRTC endpoint and connected it to RTP endpoint via media pipelines . Also make the RTP SDP offer and answering the same . Play with ffnpeg / ffplay / gst playbin
Write the requestRTPsdp to a file and obtain a RTP connector endpoint with Application/SDP .It plays okay with gst playbin ( 10 secs without audio ). Successful attempt to play from a gst playbin
gst-launch -vvv playbin uri=file:///tmp/test.sdp
but refuses to be played by VLC , ffplay and even wowza . The error generated with
End result : This results in “Could not find codec parameter for stream1 ( video:h263, none ) .Other errors types are , Could not write header for output file output file is empty nothing was encoded”
Error screenshots of trying to play the RTP SDP file with ffmpeg
Attempt 6 : Use a WebRTC capable media and streaming server ( eg Kurento ) to pick a live stream of VP8 .
Convert the VP8 to H264 ( ffmpeg / RTP endpoint )
Convert H264 to Mp4 using MP4 parser and pass to a streaming server ( wowza)
End Result : yes it did work on mozilla but with considerable lag
Update : Thankfully the updates to WebRTC standards mandated the support for PCMU and AVC/H264 CB profile in the media stack of the browser thus solving the “from scratch buildup of transcoder between webrtc and non webrtc endpoints”.
Video Codecs : RFC 7742 specifies that all WebRTC-compatible browsers must support VP8 and H.264’s Constrained Baseline profile for video.
Audio Codecs : RFC 7874 specifies that browsers must support at least the Opus codec as well as G.711’s PCMA and PCMU formats.
The latest Webrtc specification lists a set of codecs which all compliant browsers are required to support which includes chrome 52 , Firefox , safari , edge.
References :
RFC7742: WebRTC Video Processing and Codec Requirements
RFC 7874: WebRTC Audio Codec and Processing Requirements
As the title of this article suggests I am going to pen my attempts of streaming / broadcasting Live Video WebRTC call to non WebRTC supported browsers and media players such as VLC , ffplay , default video player in Linux etc.
Some of the high level archietctures for streaming Webrtc Video to multiple endpoints can be viewed in the post below.
Aim : I will be attempting to create a lightweight WebRTC to raw/h264 transcoder by making my own media engine which takes input from WebRTC peerconnection or getusermedia. I am sharing my past experiments in hope of helping someone whose objective may be to acheive the same since many non webrtc supported endpoints ( Rpi , kisosks , mobile browsers ) could benifit heavily from webrtc streaming . Even if your objective is not the same as mine, you may gain some insigh on what not to do when making a media transcoder.
It uses API fromwebrtc-experiment.com. The broadcast is in one direction only where the viewrs are never asked for their mic / webcam permission .
problem : The broadcast is for WebRTC browsers only and doesnt support non webrtc players / browsers
Attempt 1.1: Stream the media directly to nodejs through websocke
window.addEventListener('DOMContentLoaded', function () {
var v = document.getElementById('v');
navigator.getUserMedia = (navigator.getUserMedia ||
navigator.webkitGetUserMedia ||
navigator.mozGetUserMedia ||
navigator.msGetUserMedia);
if (navigator.getUserMedia) {
// Request access to video only
navigator.getUserMedia(
{
video: true,
audio: false
},
function (stream) {
var url = window.URL || window.webkitURL;
v.src = url ? url.createObjectURL(stream) : stream;
v.play();
var ws = new WebSocket('ws://localhost:3000', 'echo-protocol');
waitForSocketConnection(ws, function () {
console.log(" url.createObjectURL(stream)-----", url.createObjectURL(stream))
ws.send(stream);
console.log("message sent!!!");
});
},
function (error) {
alert('Something went wrong. (error code ' + error.code + ')');
return;
}
);
} else {
alert('Sorry, the browser you are using doesn\'t support getUserMedia');
return;
}
});
//Make the function wait until the connection is made...
function waitForSocketConnection(socket, callback) {
setTimeout(
function () {
if (socket.readyState === 1) {
console.log("Connection is made")
if (callback != null) {
callback();
}
return;
} else {
console.log("wait for connection...")
waitForSocketConnection(socket, callback);
}
}, 5); // wait 5 milisecond for the connection...
}
Problem : The video is in form of buffer and doesnot play
Attempt 2: Record the WebRTC media ( 5 secs each ) into chunks of webm format-> transfer them to other end -> append the chunks together like a regular file
This process involved the following components :
Recorder Javascript library : RecordJs
Transfer mechanism : Record using RecordRTC.js -> send to other end for media server -> stitching together the small webm files into big one at runtime and play
Programs :
Code for video recorder
navigator.getUserMedia(videoConstraints, function (stream) {
video.onloadedmetadata = function () {
video.width = 320;
video.height = 240;
var options = {
type: isRecordVideo ? 'video' : 'gif',
video: video,
canvas: {
width: canvasWidth_input.value,
height: canvasHeight_input.value
}
};
recorder = window.RecordRTC(stream, options);
recorder.startRecording();
};
video.src = URL.createObjectURL(stream);
}, function () {
if (document.getElementById('record-screen').checked) {
if (location.protocol === 'http:')
alert('https is mandatory to capture screen.');
else
alert('Multi-capturing of screen is not allowed.Have you enabled flag: "Enable screen capture support in getUserMedia"?');
} else
alert('Webcam access is denied.');
});
Code for video append-er
var FILE1 = '1.webm';
var FILE2 = '2.webm';
var FILE3 = '3.webm';
var FILE4 = '4.webm';
var FILE5 = '5.webm';
var NUM_CHUNKS = 5;
var video = document.querySelector('video');
window.MediaSource = window.MediaSource || window.WebKitMediaSource;
if (!!!window.MediaSource) {
alert('MediaSource API is not available');
}
var mediaSource = new MediaSource();
video.src = window.URL.createObjectURL(mediaSource);
function callback(e) {
var sourceBuffer = mediaSource.addSourceBuffer('video/webm; codecs="vorbis,vp8"');
GET(FILE1, function (uInt8Array) {
var file = new Blob([uInt8Array], {type: 'video/webm'});
var i = 1;
(function readChunk_(i) {
var reader = new FileReader();
reader.onload = function (e) {
sourceBuffer.appendBuffer(new Uint8Array(e.target.result));
if (i == NUM_CHUNKS) mediaSource.endOfStream();
else {
if (video.paused) {
video.play(); // Start playing after 1st chunk is appended.
}
readChunk_(++i);
}
};
reader.readAsArrayBuffer(file);
})(i); // Start the recursive call by self calling.
});
}
mediaSource.addEventListener('sourceopen', callback, false);
mediaSource.addEventListener('webkitsourceopen', callback, false);
mediaSource.addEventListener('webkitsourceended', function (e) {
logger.log('mediaSource readyState: ' + this.readyState);
}, false);
// function get the video via XHR
function GET(url, callback) {
var xhr = new XMLHttpRequest();
xhr.open('GET', url, true);
xhr.responseType = 'arraybuffer';
xhr.send();
xhr.onload = function (e) {
if (xhr.status != 200) {
alert("Unexpected status code " + xhr.status + " for " + url);
return false;
}
callback(new Uint8Array(xhr.response));
};
}
Shortcoming of this approach
The webm files failed to play on most of the media players
The recorder can only either record video or audio file at a time .
Attempt 2.Chunking and media proxy
Since the previous approach failed to support on webrtc endpoinst , the next iteration of this approach was to channel the webrtc media via a nodejs server thus disrupting the peer to peer media strem in favour of centralized / proxied emdia stream. This would enable me to obtain raw media packets form teh stream using low level C based vp8 decoder libraries and then re encode them to h364 or other media formats suitable for endpoints .
In theory media could be reencoded jusing openH264 library and the frame could be then send to players
let mediaSource = new MediaSource();
let sourceBuffer = mediaSource.addSourceBuffer('video/mp4; codecs=vp9',
new VP9Decoder());
let buffer = await loadBuffer();
sourceBuffer.appendBuffer(buffer);
Further extending for uncompressed video
let mediaSource = new MediaSource();
let sourceBuffer = mediaSource.addSourceBuffer('video/raw; codecs=yuv420p');
for (let p in demuxPAckets()) {
let frame = await codec.decode(p);
sourceBuffer.appendBuffer(frame);
}
Atleast that was the plan .
Attempt 2.1: Record the WebRTC media ( 5 secs each ) into chunks of webm format ( RecordRTC.js) > Use Kurento JS script ( kws-media-api,js) to make a HTTP Endpoint to recorded Webm files -> append the chunks together like a regular file at runtime
// UI elements
function getByID(id) {
return document.getElementById(id);
}
var recordAudio = getByID('record-audio'),
recordVideo = getByID('record-video'),
stopRecordingAudio = getByID('stop-recording-audio'),
stopRecordingVideo = getByID('stop-recording-video'),
broadcasting = getByID('broadcasting');
var canvasWidth_input = getByID('canvas-width-input'),
canvasHeight_input = getByID('canvas-height-input');
var video = getByID('video');
var audio = getByID('audio');
// Audio video constraints
var videoConstraints = {
audio: false,
video: {
mandatory: {},
optional: []
}
};
var audioConstraints = {
audio: true,
video: false
};
// Recording and stop recording - to be convrted into real time capture and chunking
const ws_uri = 'ws://localhost:8888/kurento';
var URL_SMALL = "http://localhost:8080/streamtomp4/approach1/5561840332.webm";
var audioStream;
var recorder;
recordAudio.onclick = function () {
if (!audioStream)
navigator.getUserMedia(audioConstraints, function (stream) {
if (window.IsChrome) stream = new window.MediaStream(stream.getAudioTracks());
audioStream = stream;
audio.src = URL.createObjectURL(audioStream);
audio.muted = true;
audio.play();
// "audio" is a default type
recorder = window.RecordRTC(stream, {
type: 'audio'
});
recorder.startRecording();
}, function () {
});
else {
audio.src = URL.createObjectURL(audioStream);
audio.muted = true;
audio.play();
if (recorder) recorder.startRecording();
}
window.isAudio = true;
this.disabled = true;
stopRecordingAudio.disabled = false;
};
Recording and stop recording video inot small media files ( chunks )
recordVideo.onclick = function () {
recordVideoOrGIF(true);
};
stopRecordingAudio.onclick = function () {
this.disabled = true;
recordAudio.disabled = false;
audio.src = '';
if (recorder)
recorder.stopRecording(function (url) {
audio.src = url;
audio.muted = false;
audio.play();
document.getElementById('audio-url-preview').innerHTML = '<a href="' + url + '" target="_blank">Recorded Audio URL</a>';
});
};
function recordVideoOrGIF(isRecordVideo) {
navigator.getUserMedia(videoConstraints, function (stream) {
video.onloadedmetadata = function () {
video.width = 320;
video.height = 240;
var options = {
type: isRecordVideo ? 'video' : 'gif',
video: video,
canvas: {
width: canvasWidth_input.value,
height: canvasHeight_input.value
}
};
recorder = window.RecordRTC(stream, options);
recorder.startRecording();
};
video.src = URL.createObjectURL(stream);
}, function () {
if (document.getElementById('record-screen').checked) {
if (location.protocol === 'http:')
alert('<https> is mandatory to capture screen.');
else
alert('Multi-capturing of screen is not allowed. Capturing process is denied. Are you enabled flag: "Enable screen capture support in getUserMedia"?');
} else
alert('Webcam access is denied.');
});
window.isAudio = false;
if (isRecordVideo) {
recordVideo.disabled = true;
stopRecordingVideo.disabled = false;
} else {
recordGIF.disabled = true;
stopRecordingGIF.disabled = false;
}
}
stopRecordingVideo.onclick = function () {
this.disabled = true;
recordVideo.disabled = false;
if (recorder)
recorder.stopRecording(function (url) {
video.src = url;
video.play();
document.getElementById('video-url-preview').innerHTML = '<a href="' + url + '" target="_blank">Recorded Video URL</a>';
});
};
Broadcasting the chunks to media engine
function onerror(error) {
console.log(" error occured");
console.error(error);
}
broadcast.onclick = function () {
var videoOutput = document.getElementById("videoOutput");
KwsMedia(ws_uri, function (error, kwsMedia) {
if (error) return onerror(error);
// Create pipeline
kwsMedia.create('MediaPipeline', function (error, pipeline) {
if (error) return onerror(error);
// Create pipeline media elements (endpoints & filters)
pipeline.create('PlayerEndpoint', {uri: URL_SMALL}, function (error, player) {
if (error) return console.error(error);
pipeline.create('HttpGetEndpoint', function (error, httpGet) {
if (error) return onerror(error);
// Connect media element between them
player.connect(httpGet, function (error, pipeline) {
if (error) return onerror(error);
// Set the video on the video tag
httpGet.getUrl(function (error, url) {
if (error) return onerror(error);
videoOutput.src = url;
console.log(url);
// Start player
player.play(function (error) {
if (error) return onerror(error);
console.log('player.play');
});
});
});
// Subscribe to HttpGetEndpoint EOS event
httpGet.on('EndOfStream', function (event) {
console.log("EndOfStream event:", event);
});
});
});
});
}, onerror);
}
problem : dissecting the live video into small the files and appending to each other on reception is an expensive , time and resource consuming process . Also involves heavy buffering and other problems pertaining to real-time streaming .
Attempt 2.2 : Send the recorded chunks of webm to a port on linux server. Use socket programming to pick up these individual files and play using VLC player from UDP port of the Linux Server
End Result : Small file containers play but slow buffering makes this approach non compatible for streaming files chunks and appending as single file.
Attempt 2.3: Send the recorded chunks of webm to a port on linux server socket . Use socket programming to pick up these individual webm files and convert to H264 format so that they can be send to a media server.
This process involved the following components :
Recorder Javascript library : RecordJs
Transfer mechanism :WebRTC endpoint -> Call handler ( Record in chunks ) -> ffmpeg / gstreamer to put it on RTP -> streaming server like wowza – > viewers
Programs : Use HTML webpage Webscoket connection -> nodejs program to write content from websocket to linux socket -> nodejs program to read that socket and print the content on console
Snippet to transfer the webm recorder files over websocket to nodejs program
// Make the function wait until the connection is made.
function waitForSocketConnection(socket, callback) {
setTimeout(
function () {
if (socket.readyState === 1) {
console.log("Connection is made")
if (callback != null)
callback();
} else {
console.log("wait for connection...")
waitForSocketConnection(socket, callback);
}
}, 5); // wait 5 milisecond for the connection...
}
function previewFile() {
var preview = document.querySelector('img');
var file = document.querySelector('input[type=file]').files[0];
var reader = new FileReader();
reader.onloadend = function () {
preview.src = reader.result;
console.log(" reader result ", reader.result);
var video = document.getElementById("v");
video.src = reader.result;
console.log(" video played ");
var ws = new WebSocket('ws://localhost:3000', 'echo-protocol');
waitForSocketConnection(ws, function () {
ws.send(reader.result);
console.log("message sent!!!");
});
}
if (file) {
// converts to base64 encoded string of the file data
//reader.readAsDataURL(file);
reader.readAsBinaryString(file);
} else {
preview.src = "";
}
}
Program for Linux Sockets sender which creates the socket for the webm files in nodejs
var net = require('net');
var fs = require('fs');
var socketPath = '/tmp/tfxsocket';
var http = require('http');
var stream = require('stream');
var util = require('util');
var WebSocketServer = require('ws').Server;
var port = 3000;
var serverUrl = "localhost";
var socket;
/*----------http server -----------*/
var server = http.createServer(function (request, response) {});
server.listen(port, serverUrl);
console.log('HTTP Server running at ', serverUrl, port);
/*------websocket server ----------*/
var wss = new WebSocketServer({server: server});
wss.on("connection", function (ws) {
console.log("websocket connection open");
ws.on('message', function (message) {
console.log(" stream recived from broadcast client on port 3000 ");
var s = require('net').Socket();
s.connect(socketPath);
s.write(message);
console.log(" send the stream to socketPath", socketPath);
});
ws.on("close", function () {
console.log("websocket connection close")
});
});
Program for Linux Socket Listener using nodejs and socket . Here the socket is in node /tmp/mysocket
var net = require('net');
var client = net.createConnection("/tmp/mysocket");
client.on("connect", function() {
console.log("connected to mysocket");
});
client.on("data", function(data) {
console.log(data);
});
client.on('end', function() {
console.log('server disconnected');
});
Output 1: Video Buffer displayed
Output 2 : Payload from Video displayed that shows the pipeline works but no output yet.
ffmpeg format of transfering the content from socket to UDP IP and port
ffmpeg -i unix://tmp/mysocket -f format udp://192.168.0.119:8083
problems of this approach : The video was on a passing stage from the socket and contained no information as such when tried to play / show console.
Attempt 3 : Use existing media engine like kurento to do the transocding for me
Send the live WebRTC stream from Kurento WebRTC endpoint to Kurento HTTP endpoint then play using Mozilla VLC web plugin