WHERE CAN I FIND A RTMP TEST STREAM?

What is RTMP?

Real-Time Message Protocol, or RTMP, is a structured streaming system that sends live video and audio data from an encoder to a host platform (server) over the internet. RTMP can be compared to a middleman who facilitates communication between two parties. 

Consider your social media live event streaming strategy. The real process involves connecting the video to the main streaming platform via an encoder. The stream is then simultaneously aired on that social platform. 

What is an RTMP stream?

The transmission of audio-visual waves from an encoding platform like OBS Studio to a video streaming platform like RTMP Server is known as an RTMP stream, which powers live broadcasts over the internet. 

For events and webinars that broadcast live action to an international audience, RTMP streams produce low-latency audio-video transmission, which is a huge advantage for production teams. In addition, RTMP streams offer adaptive-bitrate streaming with little buffering and the capability to fast-forward and rewind videos, providing a satisfying viewing experience.

rtmp server

Here is a detailed explanation of how RTMP for live video streaming functions.

  • Your camera and microphone record your video and sounds.
  • RTMP requires an encoder to function. The encoder processes your camera and microphone’s video and audio data into tiny data packets so they may be transferred to a streaming server more quickly.
  • Processing by streaming servers: After receiving the data from the encoder, streaming servers (such as YouTube’s servers, if you’re streaming to YouTube) prepare the content for viewing.
  • Playback: The information is shown so the viewer can see it, such as a live stream they may access on their device.

Purpose of RTMP Test Streams

Real-Time Messaging Protocol test streams are used to assess and confirm RTMP-based streaming configurations’ efficiency, effectiveness, and compatibility. These test streams act as simulated audio or video streams that can be used to gauge the RTMP connection’s latency, dependability, and overall quality.

Testing RTMP streams ensures the setup’s reliability, quality, and performance are of the highest caliber. To ensure that streaming service providers can give their audiences high-quality streaming experiences, the following issue is discovered and fixed throughout the process of RTMP test streams.

1. Troubleshooting and Debugging

2. Performance Optimization 

3. Measuring Latency

4. Assessing Video Quality

5. Testing RTMP Connectivity

It cannot be easy to locate a precise RTMP (Real-Time Messaging Protocol) test stream because it depends on the particular needs or goals you have in mind. To find test streams, you can search the following sources:

Open-Source Projects: Look at open-source initiatives for media or streaming servers like FFmpeg, OBS Studio, or NGINX. These projects frequently offer example RTMP URLs for testing.

Streaming Platforms: You can test RTMP connections using specific test streams or APIs on some streaming platforms, including YouTube or Twitch. For more details, consult their developer documentation or forums.

Online communities and forums: Participate in online communities and forums that concentrate on streaming technologies. Users can exchange test streams or offer instructions for setting up their test streams for experimenting.

Third-Party Services: RTMP stream testing can be done via third-party services. They might provide tools to create temporary test URLs for evaluation purposes or make publicly accessible test streams available.

The availability of test streams can vary, so remember to abide by any usage guidelines or terms of service set forth by the source you select.

Publishing from our server RTMP to Youtube

It is possible to publish your live streaming services to Youtube using our Stream Targets function on the RTMP Control Panel. Our RTMP Server for streaming can be used in a variety of ways, and one of them is for social media. You can stream from our RTMP Server directly to YouTube and at the same time to your Facebook and even to your website.

Login to the RTMP control Panel, and Enable Youtube Stream Target under the configuration tab >> Wowza Stream Engine >> Stream Target
rtmp to youtube live streaming

Save & Restart the service to apply the changes.
Open Stream Targets

On YouTube, when you try to start live streaming, they have to enable your live streaming, but you will need to be confirmed by phone, and then it takes 24 hours to activate.

You will need to enter the Server URL and Stream Key from https://www.youtube.com/live_dashboard

Select Streaming to software

Now they will give you an rtmp and a key. Copy this to the rtmp control panel.

Enter a name for your target, select Youtube, and then Save Stream Target.

Save Stream Target
Press Reconnect

RTMP Server in the era of HTTP video streaming

Given the growing popularity and support of HTTP video streaming, it may be tempting to consider Real Time Messaging Protocol (RTMP) streaming obsolete. But in many cases, working with RTMP server is still very meaningful. As soon as Macromedia first pronounced RTMP with Flash Player 6 in 2002, the brand new Macromedia Flash Communication Server MX (FCS) required to stream the evolving Flash Video (FLV) format, stream was live or VOD.

However, for FCS, licensing costs were high (up to $5,000 per server), and as a result, CDN costs were high than other proprietary streaming formats. Some manufacturers in the industry have called RTMP streaming costs a “Flash tax”. But Flash Player has been a ubiquitous introduction to desktop browsers for well over a decade, much larger than other plug-ins. Though, streaming server technology has traditionally been harder for web developers to implement, and simpler HTTP delivery has mostly been easier and more cost-effective. In 2003, Macromedia enabled Flash Player 7 to support HTTP delivery of FLV files, allowing integrators to use standard web server technology to deploy online video.

So in many ways, RTMP as a VOD transport delivery has not been a requirement for web video for more than a decade. As a video solution architect, I think of business requirements as the leader in most audio video decisions. And still, in 2014, RTMP is the de facto standard for the following use cases:

Publishing live streams from software/hardware encoders: Almost all streaming devices support RTMP to publish to CDN providers and streaming servers. Some native mobile applications also use RTMP libraries to publish live video from their mobile camera.

Near-instant search/playback: One of the benefits of RTMP streaming is its enhanced search capability. With real-time streaming, the player can search anywhere in the video with less buffering than HTTP delivery. However, to enjoy this feature, you need a Flash Player on your desktop and as such, it is not possible in mobile browsers. HTML5 browsers use HTTP domain requests to facilitate faster searching of VOD files.

Content Protection: RTMP Server can facilitate different levels of content protection, from obfuscation to true DRM. RTMP streams are not stored for playback on a desktop browser.

Adaptive Streaming: One of the preferred uses of RTMP is adaptive streaming playback, where we provide the video player with more bitrates and content resolutions, providing the best resolution at the current network speed. Some HTTP adaptive streaming technologies, such as HLS and MPEG-DASH, allow similar delivery, but RTMP may be more responsive to switching one bitrate to another. Unfortunately, there is no standard for HTTP adaptive streaming between HTML5 video-enabled browsers. As such, the Flash-based display of adaptive streaming is still a requirement for adaptive streaming.

Live Streaming Playback: While live streams do not need to be adaptive, compressed video segments apply the same principle to all live streaming playback. For desktop playback within the browser, virtually every live streaming event requires a video player that supports Flash-based rendering and RTMP playback. Apple Safari on iOS natively supports HLS, and luckily, all modern streaming servers, including the Wowza Streaming Engine, support RTMP and HLS compression.

The requirement for a Live Streaming Flash Player will only change if MPEG-DASH appears everywhere as Flash currently. If Internet Explorer 8, 9 and 10 support is required to install live streaming, then RTMP streaming is still required. Also, if you need a delay close to zero in a live stream, then HTTP streaming will almost certainly not meet your needs while RTMP is capable; HTTP mechanisms require that multiple packets be collected on the server before being sent to the video player.

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In summary, if your video workflow involves live streaming or any kind of packaged video installation, RTMP is the key to a successful video experience. HTTP delivery has already replaced most Flash VOD installs on video, but HTTP video can’t handle all RTMP video use cases.

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Which is better for live streaming, RTMP vs HLS vs WebRTC?

It’s hard to say which one is better, as we’re not comparing apples. Let’s break it up to the strengths and weaknesses of each method.

RTMP used to be the de facto standard for live streaming. Many CDNs offer delivery that scales to the masses. RTMP, however, does not take into consideration that broadcaster and viewers might have internet connections that aren’t always up to the task of transferring the stream at full speed. Smartphones and web browsers are also unable to play back RTMP natively; Browsers used to depend on a flash plugin for playback, which has been phased out over the last couple of years.

HLS was created for using existing HTTP CDNs for delivering live streams. It scales great, but latency can suffer in many implementations. Bitrate can adapt depending on the viewers needs. Most devices can play HLS natively, or through a javascript player. This is probably the most cost-effective way of delivering video.

WebRTC is more focused for one-to-one streaming. Bitrate is adaptable, but not many CDNs support edge delivery over WebRTC – those that do cost quite a bit more than other solutions. WebRTC can be played back in most browsers and smartphones today, using javascript players. Open source solutions for native playback and broadcasting are also available.

Red5 or Wowza Media Server consulting services

Since 2008 we worked with Red5 and Wowza Media Server platform, developing custom modules, helping our clients to integrate a number of systems to control the stats or installing a Wowza control panel into their servers, debugging performance issues, and tuning performance.

Our main areas of expertise are:

  • Live Broadcasts, Video chats Scripts
  • Server: custom module development
  • Integration: HTML, PHP, web services, etc.
  • Load Balancing

And since we know the Media Servers thoroughly, we might also be of help with:

  • Helping you choose the best server for your needs
  • Production environment setup and deployment
  • Server: installation & configuration
  • Wowza Control panel setup
  • Wowza Server: performance tuning

If you’re interested in our services, please contact us and we can further discuss your needs.

How to optimize wowza

This article provides suggestions for tuning Wowza Media Server on your production hardware.

Note: For use with Wowza Media Server® 2 and later.

 

  1. It is generally best to run the most recent Sun Java JDK (Java SE Downloads). It is best to run a 64-bit operating system with the 64-bit Java VM. The 64-bit VM enables Java heap sizes greater than 2GB.
    Note: If you are on a Mac with OS X 10.5.x or greater (Snow Leopard) with 64-bit hardware you can run the Java 6 VM. To change your default Java VM in OSX, open the Java preferences application /Applications/Utilities/Java/Java Preferences and drag the Java SE 6 item to the top of the Java Applications list.
  2. It is best to run the -server version of the Java VM. The -server option is enabled on Linux and Mac OS X by default. It is only on Windows where this option is not enabled by default. To enable the -server version of the Java VM on Windows, do the following:
    1. Install the latest version of the Oracle Java JDK and be sure that the JAVA_HOME environment variable is pointing to this Java installation.
    2. Edit the [install-dir]/bin/setenv.bat file to add the -server option to the startup scripts. For example:
      Code:
      set JAVA_OPTS=%JAVA_OPTS% -server
      Note: Affects both Windows Standalone and Windows Service, uncomment line 7.
  3. The default Java heap size for Windows is set to 768MB, and for Linux is set to 1200MB, which is great for development, but not large enough for high volume production use. If you are running the 64-bit version of the Java VM and have 4GB or more of RAM in your machine, we suggest a Java heap size of between 3000 MB to 5000 MB. If you have 16 GB or more RAM, we suggest a heap size of 8000 MB. We do not suggest a heap size above 10 GB as it can lead to long GC cycles/pauses. If you are running the 32-bit version of the Java VM or have less than 3GB of RAM in your machine, we suggest a Java heap size of 1500MB. Do not try to set a heap size greater than 1800MB when using the 32-bit VM. On many Linux platforms the server will start but will fail with an OutOfMemory error after a short period of time. Memory settings can be adjusted in the following script files (based on platform):
    • [install-dir]/bin/setenv.sh (Linux, Mac OS X and Solaris, around line 4)
      Code:
      JAVA_OPTS="-Xmx3000M"
    • [install-dir]/bin/setenv.bat (Windows Standalone and Windows Service, around line 4)
      Code:
      set JAVA_OPTS=-Xmx3000M
  4. Garbage collection (GC) tuning in Java is tricky. What works best in one server setup may not work in another. Through trial-and-error and customer feedback, we have several suggested approaches:Garbage Collection Options:

    We strongly recommend No Additional Settings. The default -server tuning (see step #2 above) works well for many streaming situations, so in most cases this tuning works great without additional settings.

    Only if you are experiencing problems with the Java Heap, such as a build-up of memory usage, should you try implementing one of the following Garbage Collection options:

    1. Concurrent Collector: The concurrent collector is designed for applications that prefer shorter garbage collection pauses and that can afford to share processor resources with the garbage collector while the application is running. The suggested additional settings are:If heap size is 5000 MB or greater, NewSize=512m
      If heap size is 3000 MB – 5000 MB, NewSize=256m
      If heap size is less than 3000 MB, NewSize=128m

      For example, with 64-bit OS, 8 GB RAM, and heap size of 6000 MB

      Code:
      -XX:+UseConcMarkSweepGC -XX:+UseParNewGC -XX:NewSize=512m
    2. Experimental: Garbage-First garbage collector (G1): In the Sun Java 6 (Update 14 or greater) release, there is a new garbage collector that is in the experimental stage. It is designed for low pause time, high-throughput applications such as Wowza Media Server. It is an option to explore. There is more information below. To enable this garbage collector, the additional settings are:
      Code:
      -XX:+UnlockExperimentalVMOptions -XX:+UseG1GC

    To change the garbage collection settings, modify the following scripts:

    • [install-dir]/bin/setenv.sh (Linux, Mac OS X, and Solaris, uncomment line 10). For example:
      Code:
      JAVA_OPTS="$JAVA_OPTS -XX:+UseConcMarkSweepGC -XX:+UseParNewGC -XX:NewSize=512m"
    • [install-dir]/bin/setenv.bat (Windows Standalone and Windows Service, uncomment line 10). For example:
      Code:
      set JAVA_OPTS=%JAVA_OPTS% -XX:+UseConcMarkSweepGC -XX:+UseParNewGC -XX:NewSize=512m

    There is more detailed information here:

    Java SE 6 HotSpot[tm] Virtual Machine Garbage Collection Tuning
    Java SE 6 Performance White Paper
    The Garbage-First Garbage Collector (G1)

  5. For low-latency chat applications, it is best to use smaller socket buffer sizes (16000 bytes for read and write). The socket buffer sizes are configured in [install-dir]/conf/VHost.xml:
    Code:
    <ReceiveBufferSize>16000</ReceiveBufferSize>
    <SendBufferSize>16000</SendBufferSize>
  6. On Linux, increase the maximum number of file descriptors. This will address the Too many files open error message. To do this, edit the following two files:/usr/local/WowzaMediaServer/bin/wms.sh

    from:

    Code:
    #ulimit -n 20000

    to:

    Code:
    ulimit -n 20000

    /usr/local/WowzaMediaServer/bin/startup.sh

    from:

    Code:
    #ulimit -n 20000

    to:

    Code:
    ulimit -n 20000
    Note: This may fail on some versions of Linux or if you do not have permission to change the descriptor limit. If after making this change Wowza Media Server will not start, consult the documentation for your Linux distribution to see how to increase the descriptor limit.
    Note: On some versions of Linux, there is also a limit in the kernel that may need to be increased. Consult the documentation for your distribution. You may need to add the following line to your /etc/sysctl.conf file:

    Code:
    fs.file-max=20000
  7. On Linux, switch to using the Anticipatory elevator (as) elevator algorithm. It is described here:Choosing an I/O Scheduler for Red Hat
  8. On Linux, mount your disk with the noatime option. This operation will differ based on your Linux distribution. Here is a basic description of the setting:Using noatime Attribute
  9. To tune your server based on the available CPU resources of your server, use the following guidelines:The [total-core-count] refers to the total number of CPU cores in your server. For example, if you have dual quad core processors (two quad core processors), the [total-core-count] is:
    Code:
    2 (processor) x 4 (cores) = 8

    If your server supports hyper-threading, then use the total number of threads. In the above example, if hyper-threading is available, then the total number of threads is:

    Code:
    2 (processor) x 4 (cores) x 2 (threads per core) = 16

    With the number of cores and threads per physical processor continually growing, we suggest a maximum number of threads for each value below:

    In the configuration file [install-dir]/conf/VHost.xml:

    Code:
    HostPort/ProcessorCount: 2x[total-core-count] (maximum of 24) Note: The HostPort/ProcessorCount field in the Admin HostPort (/Port "8086") should not be modified.
    IdleWorkers/WorkerCount: 2x[total-core-count] (maximum of 24)
    
    NetConnections/ProcessorCount: 2x[total-core-count]  (maximum of 24)
    
    RTP/UnicastIncoming/ProcessorCount: [total-core-count]  (maximum of 12)
    RTP/UnicastOutgoing/ProcessorCount: 2x[total-core-count]  (maximum of 24)
    
    RTP/MulticastIncoming/ProcessorCount: [total-core-count]  (maximum of 12)
    RTP/MulticastOutgoing/ProcessorCount: [total-core-count]  (maximum of 12)
    
    HandlerThreadPool/PoolSize: (60x[total-core-count]) (maximum of 480)
    TransportThreadPool/PoolSize: (40x[total-core-count]) (maximum of 320)

    This assumes you have at least 1GB of memory per core and that in the case you have 4 or more total cores you are running the 64-bit Java VM using the suggested memory settings above.

  10. Running Multiple Virtual Hosts:If you are running more than one virtual host, the resource allocations must be distributed between each VHost. The simplest approach is to divide the settings listed above (which are intended for a single VHost) and distribute the resources across each VHost on your system. The settings do not have to be evenly divided, however, the total should equal what you would allocate if you were configuring for asingle VHost. If one of your VHosts will be idle most of the time, you may allocate more memory than the combined total. Be careful with this setting because excessive allocations are risky. An outOfMemory error will occur in the event both VHosts exceed the combined, available resources.

    Alternatively, in [install-dir]/conf/VHost.xml, you can set each VHost/HandlerThreadPool/PoolSize to “0” and VHost/TransportThreadPool/PoolSize to “0” which will cause the [install-dir]/conf/Server.xml settings for these properties to be used instead. This will instruct Wowza Media Server to manage the PoolSize amongst VHosts.

    A mixed approach can also be used with the [install-dir]/conf/VHost.xml file by setting the PoolSize to “0” for idle/minimal-use VHosts while using higher values for busy/high-performance VHosts with higher resource requirements.

  11. If you are not doing low latency streaming and you have a client side buffer of 3 or more seconds (NetStream.bufferTime), you can reduce the CPU load on the machine and handle more concurrent sessions by editing [install-dir]/conf/VHost.xml and changing the following two values:
    Code:
    IdleWorkers/CheckFrequency: 100
    Client/IdleFrequency: 500
  12. If you are having problems with multiple incoming multicast streams interfering with each other on Linux you may need to set the Java property java.net.preferIPv4Stack to true. To do this, edit [install-dir]/bin/setenv.sh and uncomment the following line (line 13):
    Code:
    JAVA_OPTS="$JAVA_OPTS -Djava.net.preferIPv4Stack=true"
  13. This really isn’t tuning but it seems like a good place… When streaming to the Adobe Flash player, it is important to try and avoid RTMPT (tunneling version of RTMP) as much as possible. RTMPT uses a polling mechanism which is very chatty and CPU intensive. We suggest employing a protocol rollover strategy so that only connections that need RTMPT use RTMPT. Is is described here:http://www.wowzamedia.com/forums/content.php?50
  14. On Windows 7, Server 2003 or 2008, it is probably best to turn off TCP auto tuning. There is information about how to do it here:How to disable Windows Vista TCP/IP auto-tuning

How to set up live streaming using the Telestream Wirecast encoder

How to setup and play back a live stream from your Wowza Media Server using Telestream’s Wirecast encoder.