Before moving any video source or graphic overlay to your recording or stream, you usually evaluate it in the preview window. Before broadcasting or distributing your content, previewing RTMP streams is a crucial step in assuring the caliber and accuracy of your streaming setup. Before broadcasting or publishing your content, checking the RTMP stream’s compatibility, quality, and settings in preview mode has several advantages. These advantages include:

Quality Control: You can evaluate your video and audio caliber by previewing your RTMP broadcast. Pixelation, artifacts, audio distortion, and other visual and sound irregularities can all be detected.

Performance optimization: You can adjust several streaming settings while previewing the stream. You can change variables like resolution, bit rate, encoding profiles, and keyframe intervals to achieve the ideal balance between quality and bandwidth usage. 

Testing for Compatibility: You can preview the stream’s compatibility with various hardware, software, and media players. Specific platforms may have problems because not all devices or players handle RTMP streams similarly.

Content Validation: By previewing your stream, you can ensure that the material is what you expected. By verifying that the appropriate video or audio source is being recorded and transmitted, you can be sure that your audience is receiving the desired content.

Issue Detection and Troubleshooting: By previewing the stream, you can locate and address any problems that might occur when streaming. By carefully watching the preview, you can find difficulties like frame dropouts, synchronization problems for the audio, or network-related hiccups. 

Evaluation of the User Experience: By previewing the stream, you may assess the overall user experience by putting yourself in the viewer’s position. You can evaluate elements like buffering, playback smoothness, and stream latency. This aids in avoiding interruptions and maximizing viewer happiness while enabling you to create a flawless and interesting streaming experience for your audience.

You normally require a media player or streaming application that supports RTMP playback to preview an RTMP stream. The general steps needed to preview an RTMP stream are as follows:

Create your RTMP stream:  RTMP stream is a streaming protocol that transfers audio, video, and data over the Internet in real-time. To create an RTMP stream, you must have a working RTMP server or streaming platform to receive your broadcast. A remote streaming service or a local server running NGINX with the RTMP module may offer RTMP ingestion.

You can also use our RTMP Server with easy to use rtmp control panel for your RTMP service, be it for live streaming or ondemand streaming. You can visit the rtmp server, at Hosting Marketers and an experienced rtmp hosting company that since 2005 has been the premier live streaming host.

Set up your streaming software: Streaming software is software that supports a free or paid Internet streaming service. This streaming software needs to be set up. You should use an encoder or streaming program like OBS Studio, Wirecast, or FFmpeg that supports RTMP streaming. You must configure the software with the correct RTMP server information, including the stream key and server URL. 

Start the stream: You can do this by pressing a button on your software or recording device. It will begin transmitting the audio and video data to the designated RTMP server.

Activate a media player: A media player is a piece of hardware or software that can play files or discs. Open a media player that can playback RTMP files. A well-liked option that supports RTMP streams is VLC Media Player.

URL for the stream, enter: Locate the “Open Network Stream” or “Open Media” option in the media player. Enter the RTMP stream URL that your server or platform has provided. Typically, the URL looks like this: rtmp://server-url/stream-key.

Once you’ve input the correct stream URL, the media player should start playing your RTMP broadcast, allowing you to preview it. The media player now displays a preview of your stream.

 These procedures may change depending on the streaming application and media player you use.


An open-source protocol called Real-Time Messaging Protocol (RTMP), which Adobe owns is made for streaming audio and video while preserving low-latency connections. The RTMP protocol broadcasts data, audio, and video across the internet. A server that uses the RTMP protocol to control the streaming of audio and video content from a source to one or more clients is known as an RTMP server.

A video encoder or camera may be the source of the data an RTMP server receives and transmits to one or more connected clients. Web browsers, mobile devices, or other programs receiving RTMP streams can be the clients.

Live video streaming frequently employs RTMP servers, particularly for occasions like sporting events, concerts, and conferences. They are also utilized for video-on-demand (VOD) services, where users can access previously recorded videos as needed.

Adobe Media Server, Wowza Streaming Engine, and Red5 are well-known RTMP servers.


Using an RTMP server for video-on-demand and live-streaming services has various benefits:

1. Low latency: Low latency allows your live video stream to maintain a stable connection and video feed for the viewer, even if the internet connection is unreliable. This gives your viewers fewer “lags” when watching your videos with a shaky internet connection, allowing them to quickly resume the stream once their internet connection stabilizes. RTMP allows for a better viewing experience.

2. High-quality streaming: High-quality streaming with little demand on bandwidth is made possible by RTMP servers’ use of cutting-edge codecs like H.264 for video and AAC for music. By doing this, customers can enjoy pixelated or buffering-free high-quality video and audio.

3. Adaptive bitrate streaming: With adaptive bitrate streaming, the video quality can be automatically adjusted by RTMP servers based on the user’s internet connection speed. Customers can stream uninterrupted, even with a slow internet connection.

4. Scalability: RTMP servers may easily scale up or down to support multiple concurrent users, depending on the demand for streaming services.


The control of audio and video content streaming from a source to one or more clients is the responsibility of an RTMP server. Here is how an RTMP server functions:

  1. A stream of audio and video data is sent to the RTMP server by a video source, such as a camera or a video encoder. Both the H.264 video and AAC audio codecs are used in the encoding of this stream.
  1. The RTMP server receives the data stream and is kept in a buffer. The data is then divided into smaller portions and sent to the clients by the server.
  1. The clients request the audio and video stream by connecting to the RTMP server. Clients can be other apps, mobile devices, or web browsers. The server sends the split data online to the clients.
  1. After receiving the segmented data, the clients decode it with the H.264 video and AAC audio codecs. The user is then given a real-time playback of the audio and video.

Large amounts of streaming data can be handled using RTMP servers, which offer dependable and steady streaming. They employ sophisticated algorithms to maintain audio and video quality and optimize internet data delivery throughout the streaming process.

In general, an RTMP server is a crucial part of any live streaming or video-on-demand service since it enables the effective delivery of audio and video material to users via the internet.

HTTP Live Streaming (HLS) is here to stay. It works everywhere, with high quality and stability, but latency or delays between the stream and the end-user can be 60-90 seconds if you are streaming live events. That can be a very big issue. Imagine sports betting?
But with the Wowza Streaming Engine, it is possible to reduce the latency to 3 to 5 seconds.
It is not good enough, but for now, there is no other solution.

Below we have some adjustments that using the Wowza Streaming Engine Manage, which is the main software we use for the RTMP Server, we can easily improve, although, of course, there are some negative components.

These risks inherent in the configuration adjustments are due to smaller chunk sizes and may result in playback errors if you fail to increase the number of seconds built into the playlist. If a stream is interrupted, and the player requests the next playlist, the stream may be interrupted when the requested playlist doesn’t arrive.

rtmp server for live streaming

Additionally, by increasing the number of segments that are needed to create and deliver low-latency streams, you also increase the server cache requirements. To alleviate this concern, ensure that your server has a large enough cache, or built-in elasticity. You will also need to account for greater CPU and GPU utilization resulting from the increased number of keyframes. This requires careful planning for load balancing, with the understanding that increased computing and caching overhead incurs a higher cost of operation.

Lastly, as chunk sizes are smaller, the overall quality of the video playback can be impacted. This may result in either not being able to deliver a full 4K video to the player, or small playback glitches with an increased risk of packet loss. Essentially, as you increase the number of bits (markers on the chunks), you require more processing power to provide smooth playback; without it, you get packet loss and interruptions.

A- In Wowza Streaming Engine Manager, click the Applications tab and then click live in the contents panel.

On the details page Properties tab, click Cupertino Streaming Packetizer in the Quick Links bar.

In the Cupertino Streaming Packetizer area, click Edit, and change the values for the following properties:

1- cupertinoChunkDurationTarget- Set the duration of each chunk in milliseconds. The default value is 10000, but the chunkDurationTarget must be smaller to reduce the latency of Apple HLS streams. For a one-second chunk size, set the value to 1000.

2- cupertinoMaxChunkCount- Sets the maximum number of chunks Wowza Streaming Engine stores in the available chunk list. The default value is 10, but for reduced latency streaming we recommend storing 50 seconds worth of chunks. For one-second chunks, set the cupertinoMaxChunkCount to 50; if you’re using half-second chunks, the value should be doubled (100).

3-cupertinoPlaylistChunkCount-Sets the number of items returned to the player in the playlist. The default value is 3, but for reduced latency streaming, we recommend returning 12 seconds of data to the player. This prevents the loss of chunks between chunklist requests. For one-second chunks, set the cupertinoPlaylistChunkCount to 12; if you’re using half-second chunks, the value should be doubled (24).

Click Save.

B- On the Properties tab, click Custom in the Quick Links bar.

In the Custom area, click Edit.

Click Add Custom Property, specify the following custom property settings in the Add Custom Property dialog box, and then click Add:

1- In Path, select /Root/Application/LiveStreamPacketizer.

2- In Name, enter cupertinoMinPlaylistChunkCount.

3- In Type, select Integer.

4- In Value, specify the minimum number of chunks required before the Wowza Streaming Engine provides a playlist to the player. The default value is 1, but for reduced latency streaming, we recommend requiring a minimum of 6 seconds’ worth of chunks. For one-second chunks, set the cupertinoMinPlaylistChunkCount to 6; if you’re using half-second chunks, the value should be doubled (12).

Click Save and restart the application to apply the changes.

Done. Try now. Our RTMP server should now stream live with lower latency.

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.

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.

rtmp streaming with gocoder

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