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Live streaming technology is often employed to relay live events such as sports, concerts and more generally TV and Radio programmes that are output live. Often shortened to just streaming, live streaming is the process of transmitting media 'live' to computers and devices. This is a fairly complex and nascent subject with a lot of variables, so in this article, we'll introduce you to the subject and let you know how you can get started.

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To broadcast a stream over the network, we need to click the Stream option: In the Open Media dialog, select the media to stream. We can select any media. Hit the Stream button after selection. In this example, we will use a file for the media: The Stream Output window will appear. We just want to click Next to continue. Live Liga NOS streaming may be available on the Portuguese free-to-air broadcaster's over-the-top (OTT) platform RTP Play for international users. Content disclaimer: The published listings of live, re-aired, and on-demand match and program events published on this website are broadcast by the official rights holders.

The key consideration when streaming media to a browser is the fact that rather than playing a finite file we are relaying a file that is being created on the fly and has no pre-determined start or end.

Key differences between streamed and static media

In this case, we are using static media to describe media that is represented by a file, whether it be an mp3 or WebM file. This file sits on a server and can be delivered — like most other files — to the browser. This is often known as a progressive download.

Live streamed media lacks a finite start and end time as rather than a static file, it is a stream of data that the server passes on down the line to the browser and is often adaptive (see below). Usually, we require different formats and special server-side software to achieve this.

Adaptive streaming

One of the main priorities for live streaming is to keep the player synchronized with the stream: adaptive streaming is a technique for doing this in the case of low bandwidth. The idea is that the data transfer rate is monitored and if it looks like it's not keeping up, we drop down to a lower bandwidth (and consequently lower quality) stream. In order to have this capability, we need to use formats that facilitate this. Live streaming formats generally allow adaptive streaming by breaking streams into a series of small segments and making those segments available at different qualities and bit rates.

Streaming Audio and Video on Demand

San manuel casino age to gamble. Streaming technology is not used exclusively for live streams. It can also be used instead of the traditional progressive download method for Audio and Video on demand:

There are several advantages to this:

• Latency is generally lower so media will start playing more quickly
• Adaptive streaming makes for better experiences on a variety of devices
• Media is downloaded just in time which makes bandwidth usage more efficient

Streaming Protocols

While static media is usually served over HTTP, there are several protocols for serving adaptive streams; let's take a look at the options.

HTTP

For now, HTTP is by far the most commonly supported protocol used to transfer media on demand or live.

RTMP

Real Time Messaging Protocol (RTMP) is a proprietary protocol developed by Macromedia (now Adobe) and supported by the Adobe Flash plugin. RTMP comes in various flavours including RTMPE (Encrypted), RTMPS (Secure over SSL/TLS) and RTMPT (encapsulated within HTTP requests).

RTSP

Real Time Streaming Protocol (RTSP) controls media sessions between endpoints and is often used together with Real-time Transport Protocol (RTP) and with Real-time Control Protocol (RTCP) for media stream delivery. Using RTP with RTCP allows for adaptive streaming. This is not yet supported natively in most browsers, but be aware that Firefox OS 1.3 supports RTSP.
Note: some vendors implement propriety transport protocols, such as RealNetworks and their Real Data Transport (RDT).

RTSP 2.0

RTSP 2.0 is currently in development and is not backward compatible with RTSP 1.0.

Important: Although the and tags are protocol agnostic, no browser currently supports anything other than HTTP without requiring plugins, although this looks set to change. Protocols other than HTTP may also be subject to blocking from firewalls or proxy servers.

Using streaming protocols

The process of using the various protocols is reassuringly familiar if you are used to working with media over HTTP.

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Media Source Extensions (MSE)

Media Source Extensions is a W3C working draft that plans to extend HTMLMediaElement to allow JavaScript to generate media streams for playback. Allowing JavaScript to generate streams facilitates a variety of use cases like adaptive streaming and time shifting live streams.
For example, you could implement MPEG-DASH using JavaScript while offloading the decoding to MSE.

Note: Time Shifting is the process of consuming a live stream sometime after it happened.

Video Streaming File Formats

A couple of HTTP-based live streaming video formats are beginning to see support across browsers. Free slots for fun no download or registration.

Note: You can find a guide to encoding HLS and MPEG-DASH for use on the web at Setting up adaptive streaming media sources.

MPEG-DASH

DASH stands for Dynamic Adaptive Streaming over HTTP and is a new format that has recently seen support added to Chrome, and Internet Explorer 11 running on Windows 8.1. It is supported via Media Source Extensions which are used by JavaScript libraries such as DASH.js. This approach allows us to download chunks of the video stream using XHR and 'append' the chunks to the stream that's played by the element. So for example, if we detect that the network is slow, we can start requesting lower quality (smaller) chunks for the next segment. This technology also allows an advertising segment to be appended/inserted into the stream.

Note: you can also use WebM with the MPEG DASH adaptive streaming system.

HLS

HLS or HTTP Live Streaming is a protocol invented by Apple Inc and supported on iOS, Safari and the latest versions of Android browser / Chrome. HLS is also adaptive.
HLS can also be decoded using JavaScript, which means we can support the latest versions of Firefox, Chrome and Internet Explorer 10+. See this HTTP Live Streaming JavaScript player.
At the start of the streaming session, an extended M3U (m3u8) playlist is downloaded. This contains the metadata for the various sub-streams that are provided.

Streaming File Format Support

[1] Via JavaScript and MSE

[2] Via JavaScript and a CORS Proxy

Video Fallbacks

Between DASH and HLS we can cover a significant portion of modern browsers but we still need a fallback if we want to support the rest.
One popular approach is to use a Flash fallback that supports RTMP. Of course, we then have the issue that we need to encode in three different formats.

Audio Streaming File Formats

There are also some audio formats beginning to see support across browsers.

Opus

Opus is a royalty-free and open format that manages to optimize quality at various bit-rates for different types of audio. Music and speech can be optimized in different ways and Opus uses the SILK and CELT codecs to achieve this.
Currently, Opus is supported by Firefox desktop and mobile as well as the latest versions of desktop Chrome and Opera.

Note: Opus is a mandatory format for WebRTC browser implementations.

MP3, AAC, Ogg Vorbis

Most common audio formats can be streamed using specific server-side technologies.

Note: It's potentially easier to stream audio using non-streaming formats because unlike video there are no keyframes.

Server-side Streaming Technologies

In order to stream live audio and video, you will need to run specific streaming software on your server or use third-party services.

GStreamer

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GStreamer is an open source cross-platform multimedia framework that allows you to create a variety of media-handling components, including streaming components. Through its plugin system, GStreamer provides support for more than a hundred codecs (including MPEG-1, MPEG-2, MPEG-4, H.261, H.263, H.264, RealVideo, MP3, WMV, and FLV.)

GStreamer plugins such as souphttpclientsink and shout2send exist to stream media over HTTP. You can also integrate with Python's Twisted framework or use something like Flumotion (open source streaming software). Where was poker invented.

For RTMP transfer you can use the Nginx RTMP Module.

SHOUTcast

SHOUTcast is a cross-platform proprietary technology for streaming media. Developed by Nullsoft, it allows digital audio content in MP3 or AAC format to be broadcast. For web use, SHOUTcast streams are transmitted over HTTP.

Note: SHOUTcast URLs may require a semi-colon to be appended to them.

Icecast

The Icecast server is an open source technology for streaming media. Maintained by the Xiph.org Foundation, it streams Ogg Vorbis/Theora as well as MP3 and AAC format via the SHOUTcast protocol.

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Note: SHOUTcast and Icecast are among the most established and popular technologies, but there are many more streaming media systems available.

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Streaming Services

Although you can install software like GStreamer, SHOUTcast and Icecast you will also find a lot of third-party streaming services that will do much of the work for you.

See also

So you have your webcam or IP camera all hooked up and ready to begin streaming video. Now you just need to figure out how people can watch it. Ideally, it should be simple and universally available; like an internet browser.

Party poker casino. Unfortunately, most IP cameras are RTSP based which is not natively supported in internet browsers. So how do you use an IP camera for live streaming?

One solution is to connect to the RTSP stream and view it in VLC Media Player. However, that requires additional configurations and is not convenient. It would be much easier if you could just view the video in a standard web browser. Plus what if you need to support thousands or millions of viewers?

Fortunately, WebRTC is a protocol widely accepted by internet browsers. All you need to do is convert the RTSP feed coming from your IP camera to WebRTC. This post will explain how that can be done with Red5 Pro.

How Do I Convert RTSP to WebRTC?

The simple answer is to use Red5 Pro. We've built that logic into our Restreamer Plugin so all that work is done automatically once it's configured. We also have failover support for RTMP and HLS for the few clients that don't support WebRTC.

However, I'm sure that some of you reading this are wondering about the specific process we use to convert RTSP to WebRTC. Read on, my inquisitive friends and we will cover that in more detail.

In order to integrate an IP camera (or general RTSP stream) with WebRTC, you first need to achieve media interoperability (a fancy term for making them work together). The media stream sent out by the camera needs to be made compatible with formats supported by browsers and the WebRTC codecs.

What is RTSP used for?

RTSP is a streaming control protocol that is used to control the streaming server, kind of like how a remote control works with a TV (enabling play, pause, etc.). It does not actually transport the stream, rather it defines how the data in the stream should be packaged for delivery. It also defines how both ends of the connection should behave to prepare a pathway for transportation.

This process is also known as signaling. WebRTC uses a different method to handle signaling typically involving WebSockets.

RTP is used for the actual transportation of the data/video stream. Since RTP is a transport protocol it could be said that it is like a runway and flight path between two airports while RTSP is the air traffic controller that makes sure the runway is open and the flightpath is clear of any obstacles. The actual video codecs (covered in more detail below) used to encode the video data would be the airplane itself.

How Do RTSP and WebRTC Work Together?

As WebRTC also uses RTP for its transport protocol, they are very compatible together. The complication comes from how IP cameras behave.

IP cameras directly produce an RTSP stream, acting more like a server. Normally in the case of webcam or cell phone videos, the Red5 Pro (origin) server takes in the broadcasting stream from those clients. In this way, there is a challenge in connecting a Red5 Pro server, to an IP camera that is also acting as a server. You need a client implementation in the mix, in this case an RTSP client to consume the stream from the IP camera. Just like an electrical socket, you can't join two of the same types of connections: the female wall outlet will only take a male plug. Female to female can't link to each other and you can't connect servers. You need a converter.

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To perform the conversions role, Red5 Pro created a Restreamer Plugin to pull the RTSP stream as a client and re-stream it out over WebRTC. The stream can then be delivered over WebRTC to the browser clients.

In Red5 Pro, our primary ingest codecs are H.264 for video and AAC for audio. Normally, the IP cameras use either RTSP or MPEG-TS (the latter not using RTP) to encode media while WebRTC defaults to VP8 (video) and Opus (audio) in most applications. Since most modern browsers accept H.264 it is faster for Red5 Pro to simply pass the H.264 codec straight through WebRTC while transcoding the AAC codec to Opus. Finally, it routes to WebRTC clients using the WebRTC protocol stack.

However, in some use cases older systems might not be able to handle H.264. In those rare cases Red5 Pro converts H.264 to VP8.

Though this post does not focus on MPEG-TS, it bears mentioning that Red5 Pro supports both MPEG-TS and RTSP. MPEG-TS is either multicast or unicast ingest whereas RTSP would be unicast. Both support H.264 video and AAC audio codecs.

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So That's How it All Works

Fortunately, as we covered earlier, Red5 Pro has already done all this work for you. Find out more by visiting our website and documentation pages. Send any questions you have to info@red5pro.com or schedule a call with us.