Category Archives: Live Streaming and Broadcasting

crtmpserver + ffmpeg

This post will show the process of installing , running and using crtmpserver on ubuntu 64 bit machine with gstreamer .

gcc and cmake

We shall build gstreamer directly from sources . For this we first need to determine if gcc is installed on the machine .

If not installed then  run the following command

GNU Compiler Collection (GCC) is a compiler system produced by the GNU Project supporting various programming languages( C, C++, Objective-C, Fortran, Java, Ada, Go etc).

sudo apt-get install build-essential

once it is isnatlled it can be tested with printing the version

Screenshot from 2016-06-09 11-24-33.png

cmake is a software compilation tool.It uses compiler independent configuration files, and generate native makefiles and workspaces that can be used in the differemt compiler environment .

Crtmpserver

To get the source code from git install git first . Then clone the project from https://github.com/j0sh/crtmpserver

sudo apt-get git
git clone https://github.com/j0sh/crtmpserver.git
cd crtmpserver/builders/cmake

Next we create all makefile’s using cmake .

cmake .

Output should look as follows

Screenshot from 2016-06-09 11-47-05

Run make to do compilation

make

Screenshot from 2016-06-09 11-57-19

Run using following command . If should print out a list of ports and their respecting functions

./crtmpserver/crtmpserver crtmpserver/crtmpserver.lua

+—————————————————————————–+
| Services|
+—+—————+—–+————————-+————————-+
| c | ip | port| protocol stack name | application name |
+—+—————+—–+————————-+————————-+
|tcp| 0.0.0.0| 1112| inboundJsonCli| admin|
+—+—————+—–+————————-+————————-+
|tcp| 0.0.0.0| 1935| inboundRtmp| appselector|
+—+—————+—–+————————-+————————-+
|tcp| 0.0.0.0| 8081| inboundRtmps| appselector|
+—+—————+—–+————————-+————————-+
|tcp| 0.0.0.0| 8080| inboundRtmpt| appselector|
+—+—————+—–+————————-+————————-+
|tcp| 0.0.0.0| 6666| inboundLiveFlv| flvplayback|
+—+—————+—–+————————-+————————-+
|tcp| 0.0.0.0| 9999| inboundTcpTs| flvplayback|
+—+—————+—–+————————-+————————-+
|tcp| 0.0.0.0| 6665| inboundLiveFlv| proxypublish|
+—+—————+—–+————————-+————————-+
|tcp| 0.0.0.0| 8989| httpEchoProtocol| samplefactory|
+—+—————+—–+————————-+————————-+
|tcp| 0.0.0.0| 8988| echoProtocol| samplefactory|
+—+—————+—–+————————-+————————-+
|tcp| 0.0.0.0| 1111| inboundHttpXmlVariant| vptests|
+—+—————+—–+————————-+————————-+

If you the following types of errors while pushing a stream to crtmpserver , they just denote they your pipe is not using the correct format.

/home/altanai/crtmpserver/sources/thelib/src/netio/epoll/tcpacceptor.cpp:154 Client connected: 127.0.0.1:55524 -> 0.0.0.0:8080
/home/altanai/crtmpserver/sources/thelib/src/netio/epoll/iohandlermanager.cpp:119 Handlers count changed: 11->12 IOHT_TCP_CARRIER
/home/altanai/crtmpserver/sources/thelib/src/protocols/http/basehttpprotocol.cpp:281 Headers section too long
/home/altanai/crtmpserver/sources/thelib/src/protocols/http/basehttpprotocol.cpp:153 Unable to read response headers: CTCP(16) <-> TCP(13) <-> [IHTT(14)] <-> IH4R(15)
/home/altanai/crtmpserver/sources/thelib/src/netio/epoll/tcpcarrier.cpp:89 Unable to signal data available
/home/altanai/crtmpserver/sources/thelib/src/netio/epoll/iohandlermanager.cpp:129 Handlers count changed: 12->11 IOHT_TCP_CARRIER
/home/altanai/crtmpserver/sources/thelib/src/protocols/protocolmanager.cpp:45 Enqueue for delete for protocol [IH4R(15)]
/home/altanai/crtmpserver/sources/thelib/src/application/baseclientapplication.cpp:240 Protocol [IH4R(15)] unregistered from application: appselector
/home/altanai/crtmpserver/sources/thelib/src/netio/epoll/tcpacceptor.cpp:154 Client connected: 127.0.0.1:44964 -> 0.0.0.0:9999
/home/altanai/crtmpserver/sources/thelib/src/netio/epoll/iohandlermanager.cpp:119 Handlers count changed: 11->12 IOHT_TCP_CARRIER
/home/altanai/crtmpserver/sources/thelib/src/protocols/ts/inboundtsprotocol.cpp:211 I give up. I'm unable to detect the ts chunk size
/home/altanai/crtmpserver/sources/thelib/src/protocols/ts/inboundtsprotocol.cpp:136 Unable to determine chunk size
/home/altanai/crtmpserver/sources/thelib/src/netio/epoll/tcpcarrier.cpp:89 Unable to signal data available
/home/altanai/crtmpserver/sources/thelib/src/netio/epoll/iohandlermanager.cpp:129 Handlers count changed: 12->11 IOHT_TCP_CARRIER
/home/altanai/crtmpserver/sources/thelib/src/protocols/protocolmanager.cpp:45 Enqueue for delete for protocol [ITS(17)]
/home/altanai/crtmpserver/sources/thelib/src/application/baseclientapplication.cpp:240 Protocol [ITS(17)] unregistered from application: flvplayback
/home/altanai/crtmpserver/sources/thelib/src/netio/epoll/tcpacceptor.cpp:154 Client connected: 127.0.0.1:37754 -> 0.0.0.0:1935
/home/altanai/crtmpserver/sources/thelib/src/netio/epoll/iohandlermanager.cpp:119 Handlers count changed: 11->12 IOHT_TCP_CARRIER
/home/altanai/crtmpserver/sources/thelib/src/protocols/rtmp/inboundrtmpprotocol.cpp:77 Handshake type not implemented: 85
/home/altanai/crtmpserver/sources/thelib/src/protocols/rtmp/basertmpprotocol.cpp:309 Unable to perform handshake
/home/altanai/crtmpserver/sources/thelib/src/netio/epoll/tcpcarrier.cpp:89 Unable to signal data available
/home/altanai/crtmpserver/sources/thelib/src/netio/epoll/iohandlermanager.cpp:129 Handlers count changed: 12->11 IOHT_TCP_CARRIER
/home/altanai/crtmpserver/sources/thelib/src/protocols/protocolmanager.cpp:45 Enqueue for delete for protocol [IR(19)]
/home/altanai/crtmpserver/sources/thelib/src/application/baseclientapplication.cpp:240 Protocol [IR(19)] unregistered from application: appselector
/home/altanai/crtmpserver/sources/thelib/src/netio/epoll/tcpacceptor.cpp:154 Client connected: 127.0.0.1:48368 -> 0.0.0.0:6666
/home/altanai/crtmpserver/sources/thelib/src/protocols/liveflv/inboundliveflvprotocol.cpp:51 _waitForMetadata: 1
/home/altanai/crtmpserver/sources/thelib/src/netio/epoll/iohandlermanager.cpp:119 Handlers count changed: 11->12 IOHT_TCP_CARRIER
/home/altanai/crtmpserver/sources/thelib/src/protocols/liveflv/baseliveflvappprotocolhandler.cpp:45 protocol CTCP(16) <-> TCP(20) <-> [ILFL(21)] registered to app flvplayback
/home/altanai/crtmpserver/sources/thelib/src/protocols/liveflv/inboundliveflvprotocol.cpp:102 Frame too large: 6324058
/home/altanai/crtmpserver/sources/thelib/src/netio/epoll/tcpcarrier.cpp:89 Unable to signal data available
/home/altanai/crtmpserver/sources/thelib/src/netio/epoll/iohandlermanager.cpp:129 Handlers count changed: 12->11 IOHT_TCP_CARRIER
/home/altanai/crtmpserver/sources/thelib/src/protocols/protocolmanager.cpp:45 Enqueue for delete for protocol [ILFL(21)]
/home/altanai/crtmpserver/sources/thelib/src/protocols/liveflv/baseliveflvappprotocolhandler.cpp:58 protocol [ILFL(21)] unregistered from app flvplayback

ffmpeg

Download and install ffmpeg from git

 git clone https://git.ffmpeg.org/ffmpeg.git ffmpeg
cd ffmpeg

Once the source code is obtained we need to configure , make and make install it .
We need to have following plugins for muxing and ecoding like libx264 for h264parse , so we configure with the following options

./configure \
  --prefix="$HOME/ffmpeg_build" \
  --pkg-config-flags="--static" \
  --extra-cflags="-I$HOME/ffmpeg_build/include" \
  --extra-ldflags="-L$HOME/ffmpeg_build/lib" \
  --bindir="$HOME/bin" \
  --enable-gpl \
  --enable-libass \
  --enable-libfreetype \
  --enable-libopus \
  --enable-libtheora \
  --enable-libvorbis \
  --enable-libx264 \
  --enable-libx265 \
  --enable-nonfree

the make and make install

make
sudo make install

Screenshot from 2016-06-09 16-59-49

Incase of errors  on ffmpeg configure command , you need to install the respective missing / not found library

libass

sudo apt-get install libass-dev

lamemp3

sudo apt-get install libmp3lame-dev

libaacplus

sudo apt-get install autoconf
sudo apt-get install libtool

wget -O libaacplus-2.0.2.tar.gz http://tipok.org.ua/downloads/media/aacplus/libaacplus/libaacplus-2.0.2.tar.gz
tar -xzf libaacplus-2.0.2.tar.gz
cd libaacplus-2.0.2
./autogen.sh --with-parameter-expansion-string-replace-capable-shell=/bin/bash --host=arm-unknown-linux-gnueabi --enable-static

make
sudo make install

libvorbis
compressed audio format for mid to high quality (8kHz-48.0kHz, 16+ bit, polyphonic) audio and music at fixed and variable bitrates from 16 to 128 kbps/channe. It is from the same reank as MPEG4 AAC

wget http://downloads.xiph.org/releases/vorbis/libvorbis-1.3.2.tar.bz2
tar -zxvf libvorbis-1.3.2.tar.bz2
cd libvorbis-1.3.2
./configure && make && make install

libx264
encoding video streams into the H.264/MPEG-4 AVC compression format, and is released under the terms of the GNU GPL.

git clone git://git.videolan.org/x264
cd x264
./configure --host=arm-unknown-linux-gnueabi --enable-static --disable-opencl
make
sudo make install

libvpx
libvpx is an emerging open video compression library which is gaining popularity for distributing high definition video content on the internet.

sudo apt-get install checkinstall
git clone https://chromium.googlesource.com/webm/libvpx
cd libvpx
./configure
make
sudo checkinstall --pkgname=libvpx --pkgversion="1:$(date +%Y%m%d%H%M)-git" --backup=no     --deldoc=yes --fstrans=no --default

librtmp
librtmp provides support for the RTMP content streaming protocol developed by Adobe and commonly used to distribute content to flash video players on the web.

sudo apt-get install libssl-dev
cd /home/pi/src
git clone git://git.ffmpeg.org/rtmpdump
cd rtmpdump
make SYS=posix
sudo checkinstall --pkgname=rtmpdump --pkgversion="2:$(date +%Y%m%d%H%M)-git" --backup=no --deldoc=yes --fstrans=no --default

Reference:
http://www.videolan.org/developers/x265.html
https://trac.ffmpeg.org/wiki/CompilationGuide/RaspberryPi
http://wiki.serviio.org/doku.php?id=howto:linux:install:raspbian
http://lame.sourceforge.net/

Additionally “pkg-config –list-all” command list down all the installed libraries.


RTMP streaming

1.start the stream from linux machine using ffmpeg

ffmpeg -f video4linux2 -s 320x240 -i /dev/video0 -f flv -s qvga -b 750000 -ar 11025 -metadata streamName=aaa "tcp://<hidden_ip>:6666/live";

Screenshot from 2016-06-11 17-50-02

2.view the incoming packets and stats on terminal at crtmpserver

Screenshot from 2016-06-11 17-53-22

3.playback the livestream from another machine

using ffplay
ffplay -i rtmp://server_ip:1935/live/ccc

Screenshot from 2016-06-09 15-43-58

RTSP streaming

1.start the rtsp stream from linux machine using ffmpeg

here using resolution 320×240 and stream name test

ffmpeg -f video4linux2 -s 320x240 -i /dev/video0 -an -r 10 -c:v libx264 -q 1 -f rtsp -metadata title=test rtsp://server_ip:5554/flvplayback

crtmp2

2.view the incoming packets and stats on terminal at crtmpserver

3.playback the livestream from another machine using

ffplay

ffplay rtsp://server_ip:5554/flvplayback/test

Screenshot from 2016-06-09 18-17-07

VLC

vlc rtsp://server_ip:5554/flvplayback/test

 

 

GStreamer-1.8.1 rtsp server and client on ubuntu

GStreamer is a streaming media framework, based on graphs of filters which operate on media data.

Gstreamer is constructed using a pipes and filter architecture.
The basic structure of a stream pipeline is that you start with a stream source (camera, screengrab, file etc) and end with a stream sink (screen window, file, network etc). The ! are called pads and they connect the filters.

Data that flows through pads is described by caps (short for capabilities). Caps can be though of as mime-type (e.g. audio/x-raw, video/x-raw) along with mime-type (e.g. width, height, depth).

Source Code

Download the latest archives from https://gstreamer.freedesktop.org/src/

Source code on git : https://github.com/GStreamer

Primarily 3 files are required

  1. gstreamer-1.8.1.tar.xz
  2. gst-plugins-base-1.8.1.tar.xz
  3. gst-rtsp-server-1.8.1.tar.xz

If the destination machine is a ec2 instance one can also scp the tar.xz file there

To extract the tar.xz files use tar -xf <filename> it will create a folder for each package.

Prerequisites

build-essentials

sudo apt-get install build-essentials

bison

flex

GLib >= 2.40.0

GLib package contains low-level libraries useful for providing data structure handling for C, portability wrappers and interfaces for such runtime functionality as an event loop, threads, dynamic loading and an object system.

sudo apt-get install libglib2.0-dev

gstreamer

Installing gstreamer 1.8.1 . Gstreamer create a media stream with elements and properties as will be shown on  later sections of this tutorial .

cd gstreamer-1.8.1
./configure
make
sudo make install

Screenshot from 2016-05-19 16-51-29.png

Screenshot from 2016-05-19 16-55-27.png

Screenshot from 2016-05-19 16-56-05.png

after installation  export the path

export LD_LIBRARY_PATH=/usr/local/lib

then verify the installation of the gstreamer by

gst-inspect-1.0

provides information on installed gstreamer modules ie print out a long list ( about 123 in my case ) plugin that are installed such as coreelements:

capsfilter: CapsFilter ximagesink: ximagesink: Video sink videorate: videorate: Video rate adjuster typefindfunctions: image/x-quicktime: qif, qtif, qti typefindfunctions: video/quicktime: mov, mp4 typefindfunctions: application/x-3gp: 3gp typefindfunctions: audio/x-m4a: m4a typefindfunctions: video/x-nuv: nuv typefindfunctions: video/x-h265: h265, x265, 265 typefindfunctions: video/x-h264: h264, x264, 264 typefindfunctions: video/x-h263: h263, 263 typefindfunctions: video/mpeg4: m4v typefindfunctions: video/mpeg-elementary: mpv, mpeg, mpg typefindfunctions: application/ogg: ogg, oga, ogv, ogm, ogx, spx, anx, axa, axv typefindfunctions: video/mpegts: ts, mts typefindfunctions: video/mpeg-sys: mpe, mpeg, mpg typefindfunctions: audio/x-gsm: gsm

gst plugins

Now build the plugins

cd gst-plugins-base-1.8.1
./configure
make
sudo make install

 

gst plugins good

cd gst-plugins-good-1.8.1.tar
./configure
 make
sudo make install

RTSP Server

Now make and install the rtsp server

cd gst-rtsp-server-1.8.1
./configure

last few lines from console traces

Configuration
Version : 1.8.1
Source code location : .
Prefix : /usr/local
Compiler : gcc -std=gnu99
CGroups example : no

make

It will compile the examples .

sudo make install

 

stream video test src

~/mediaServer/gst-rtsp-server-1.8.1/examples]$./test-launch --gst-debug=0 &quot;( videotestsrc ! video/x-raw,format=(yuv),width=352,height=288,framerate=15/1 ! x264enc ! rtph264pay name=pay0 pt=96 )&quot;
stream ready at rtsp://127.0.0.1:8554/test

Ref:

Manual for developers : https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-rtsp-server/html/index.html


Simplest pipeline

gst-launch-1.0 fakesrc ! fakesink

➜ ~ gst-launch-1.0 fakesrc ! fakesink Setting pipeline to PAUSED ... Pipeline is PREROLLING ... Pipeline is PREROLLED ... Setting pipeline to PLAYING ... New clock: GstSystemClock
To stop press ctrl +c ^
Chandling interrupt. Interrupt: Stopping pipeline ... Execution ended after 0:00:48.004547887 Setting pipeline to PAUSED ... Setting pipeline to READY ... Setting pipeline to NULL ... Freeing pipeline ... [/sourcecode ] or to display to a audiovideosink gst-launch-1.0 videotestsrc ! autovideosink
Screenshot from 2016-05-20 12-31-18.png To capture webcam
gst-launch v4l2src ! xvimagesink

Screenshot from 2016-05-20 13-06-56.png

Wowza RTMP Authentication with Third party Token provider over Tiny Encryption Algorithm (TEA)

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 .

  1. RTMP Encoder Application
  2. Wowza RTMP Auth module
  3. Third party Authentication Server

The component diagram is as follows :

Copy of Publisher App iOS

The detailed explanation of the components are :

1.Wowza RTMP Auth module

The Wowza Server receives a rtmp stream url in the format as :

rtmp://username:pass@wowzaip:1935/Application/stteamname

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.

<Property>
<Name>secureTokenSharedSecret</Name>
<Value><![CDATA[abcdefghijklmnop]]></Value>
</Property>

<Property>
<Name>serverTokenValidatorURL</Name>
<Value>http://127.0.0.1:8080/TokenProvider/authentication/token</Value&gt;
</Property>

The values of serverTokenValidatorURL is the third party auth server listening for REST POST request .

The code for receiving the incoming  resulting json data is


	ObjectMapper mapper = new ObjectMapper();
	JsonNode node = mapper.readTree(connection.getInputStream()); 
	node = node.get("publisherToken") ;
	String token = node.asText();
        String token2 =TEA.decrypt(token, sharedSecret);

2.Third party Authentication Server

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


        try {

            String clientID = TEA.decrypt(id, sharedSecret);
            
            String token= findUserPassword(clientID);
            
             token = TEA.encrypt(token, sharedSecret); 
                        
            return "{\"publisherToken\":\""  + token+ "\"}";
            
        }catch (Exception ex) {

            return "{\"error\":\"Invalid Client\"}";
        }

The final callflow thus becomes :

Copy of Publisher App iOS (1)

Screenshots :

Screenshot_2015-09-16-20-22-37Screenshot_2015-09-17-18-36-23Screenshot_2015-09-16-20-22-42Screenshot_2015-09-16-20-23-30

Wowza RTMP Authenticate Module

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

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

Wowza RTMP Auth

Step 4 : Proceed with the creation , uncheck the event methods as we are not using them right now .

Screenshot from 2015-09-17 13:10:24

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 .

Screenshot from 2015-09-17 13:11:58

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.

Screenshot from 2015-09-17 13:12:23

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

sudo ./usr/local/WowzaStreamingEngine/manager/bin/startmgr.sh

The console can be opened at http://127.0.0.1:8088

Screenshot from 2015-09-17 13:53:58

Also you can see that RTMPAuthSampleCode.jar would have been copied to /usr/local/WowzaStreamingEngine/lib folder.

Step 9: Add module to applications

Add folder “RTMPAuthSampleCode” inside /usr/local/WowzaStreamingEngine/applications folder .

Step 10 : Add conf

Add folder “RTMPAuthSampleCode” inside /usr/local/WowzaStreamingEngine/conf  folder

Copy paste Application.xml from conf folder inside RTMPAuthSampleCode folder and make the following changes .

Add the ModuleRTMPAuthenticate module to Modules

<Module> <Name>ModuleRTMPAuthenticate</Name> <Description>ModuleRTMPAuthenticate</Description> <Class>com.wowza.wms.security.ModuleRTMPAuthenticate</Class> </Module>

and comment ModuleCoreSecurity

<!--    <Module>
     <Name>ModuleCoreSecurity</Name>
     <Description>Core Security Module for Applications</Description>
     <Class>com.wowza.wms.security.ModuleCoreSecurity</Class>
</Module> -->

Step 11: Add property usernamePasswordProviderClass to Properties .

usualy present inside Application at the bootom of Application.xml file

<Property>
<Name>usernamePasswordProviderClass</Name>
<Value>com.wowza.wms.example.authenticate.RTMPAuthSampleCode</Value>
</Property>

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 .

References :

  1. Media security overview
    http://www.wowza.com/forums/content.php?115-MediaSecurity-AddOn-Package-(SecureToken-RTMP-RTSP-Authentication-and-more
  2. How to integrate Wowza user authentication with external authentication systems (ModuleRTMPAuthenticate)
    http://www.wowza.com/forums/content.php?236-How-to-integrate-Wowza-user-authentication-with-external-authentication-systems-%28ModuleRTMPAuthenticate%29
  3. How to enable username/password authentication for RTMP and RTSP publishing
    http://www.wowza.com/forums/content.php?449-How-to-enable-username-password-authentication-for-RTMP-and-RTSP-publishing
  4. configuration ref 4.2 http://www.wowza.com/resources/WowzaStreamingEngine_ConfigurationReference.pdf

WebRTC Live Stream Broadcast

WebRTC has the potential to drive the Live Streaming broadcasting area with its powerful no plugin , no installation , open standard  policy . However the only roadblock is the VP8 codec which differs from the traditional H264 codec that is used by almost all the media servers , media control units , etc .

This post is first in the series of building a WebRTC based broadcasting solution. Note that a p2p session differs from a broadcasting session as Peer-to-peer session applies to bidirectional media streaming where as broadcasting only applies unidirectional media flow.

Scalable Broadcasting and Live streaming alternatives

1. WebRTC multi peers

Since WebRTC is p2p technology , it is convenient to build a  network of webrtc client viewers which can pass on the stream to 3 other peers in different session. In this fashion a fission chain like structure is created where a single stream originated to first peer is replicated to 3 others which is in turn replicated to 9 peers etc .

WebRTC Scalable Streaming Server -WebRTC multi peers

WebRTC Scalable Streaming Server -WebRTC multi peers

Advantages :

  1. Scalable without the investment of media servers
  2. No additional space required at service providers network .

Disadvantage :

  1. The entire set of end clients to a node get disconnected if a single node is broken .
  2. Since sessions are dynamically created , it is difficult to maintain a map with fallback option in case of service disruption from any single node .
  3. Client incur bandwidth load of 2 Mbps( stream incoming peer ) incoming and 6 Mbps ( for 3 connected peers ) outgoing data .

2. Torrent based WebRTC chain

To over come the shortcoming of previous approach of  tree based broadcasting , it is suggested to use a chained broadcasting mechanism .

WebRTC Scalable Streaming Server v1 (4)

WebRTC Scalable Streaming Server- Single chain connection

To improvise on this mechanism for incresing efficieny for slow bandwidth connections we can stop their outgoing stream converting them to only consumers . This way the connection is mapped and arranged in such a fashion that every alternate peer is connected to 2 peers  for stream replication. The slow bandwidth clients can be attached as independent endpoints . WebRTC Scalable Streaming Server v1 (3)

3. WebRTC Relay nodes for multiple peers

The aim here is to build a career grade WebRTC stream broadcasting platform , which is capable of using the WebRTC’s mediastream and peerconnection API , along with repeaters to make a scalable broadcasting / live streaming solution using socketio for behavior control and signalling .

Algorithm :

At the Publisher’s end

1. GetUserMedia
2. Start Room “liveConf”
3. Add outgoing stream to session “liveConf “ with peer “BR” in 1 way transport .

1 outgoing audio stream -> 1 MB in 1 RTP port
1 outgoing video -> 1 MB 1 more RTP port
Total Required 2 MB and 2 RTP ports

At the Repeater layer (high upload and download bandwidth )

4. Peer “BR” opens parallel room “liveConf_1” , “liveConf_2” with 4 other peers “Repeater1 “, “Repeater2” , so on
5. Repetare1 getRemoteStream from “liveConf_1” and add as localStream to “liveConf_1_1”

Here the upload bandwidth is high and each repeater is capable of handling 6 outgoing streams . Therefore total 4 repeaters can handle upto 24 streams very easily

At the Viewer’s end

6. Viewer Joins room ”liveConf_1_1”
7. Play the incoming stream on WebRTC browser video element”

WebRTC Relay nodes for multiple peers

WebRTC Relay nodes for multiple peers

Advantages:

  1. As 6 viewers can connect to 1 repeater for feed , total of 24 viewers will require only 4 repeaters.
  2. Only 2 MB consumption at publishers end and 2MB at each viewer’s end.

4. WebRTC  recorder to Broadcasting Media Server VOD

This process is essentially NOT a live streaming solution but a Video On Demand type of implementation for a recorded webRTC stream .

Figure shows a WebRTC node which can record the webrtc files as webm . Audio and video can be together  recorded on fireox . With chrome one needs to merge a separately recorded webm ( video) and wav ( audio ) file to make a single webm file containing both audio and video and them store in VOD server’s repo.

WebRTC Scalable Streaming Server  - WebRTC Chunk recorder to Broadcasting Media Server VOD

WebRTC Chunk recorder to Broadcasting Media Server VOD

Although inherently Media Server do not support webm format but few new age lightweight media servers such as Kurento are capable of this .

Advantages :

  1. Can solve the end goal of broadcasting from a webrtc browser to multiple webrtc browsers without incurring extra load on any client machine ( Obviously assuming that  Media Server handles the distribution of video and load sharing automatically )

Disadvantages:

  1. It is not livestreaming
  2. For significantly longer recorded stream the delta in delay of streaming increases considerably .  Ideally this delta should be no more than 5 minutes .