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- Streaming video is a a video/audio data transfer technique to allow steady and continuous processing by a client.
See also: Videoconferencing, Flash video component tutorial
Multimedia (or video) streaming is based on the following technology
- In order to play video immediately as it is downloaded from the Internet (as opposed to storing it in a file on the receiving computer first) special servers and clients are needed.
- Clients can be web browser plug-ins or special stand-alone applications.
- A compression method (also called Codec) is used to compress data. In order to decompress a file in real time, a fast computer is needed.
- A special transport protocol is needed for optimal transfer.
Streaming bandwidth and storage[edit | edit source]
- Storage size
According to Wikipedia, storage size is calculated from streaming bandwidth and length of the media with the following formula:
storage size (in mebibytes) = length (in seconds) · bit rate (in kbit/s) / 8388.608
Real world example:
One hour of video encoded at 300 kbit/s (this is a typical broadband video for 2005 and it's usually encoded in a 320×240 pixels window size) will be:
(3600 s * 300 kbit/s) / 8388.608 = 128.7 MiB of storage
Note: A mebibyte is roughly a megabyte.
Here is an example taken from the ePresence website: “Compressed video is about 200mb per hour per format. If you support all 3 media formats (Windows Media, Real media and QuickTime) you will need about 500-600 mb per hour. Therefore one a 100 GB hard drive can hold approximately 500 hours of video in one format or 150 hours x 3 formats. Please note that these numbers can vary depending on the encoding options.”, retrieved 13:13, 20 March 2007 (MET).
- Bandwidth
Problems arise when lots of people pull the a file. E.g. if the above stream is viewed by by 1000 people, one would need:
300 kbit/s * 1000 = 300'000 kbit/s = 300 Mbit/s of bandwidth
Video compression (codecs) can make a difference. E.g. the more recent MPEG-4 H.264 standard has the following performance according to Apple (retrieved 13:13, 20 March 2007 (MET)):
Use Scenario |
Resolution & Frame Rate |
Example Data Rates |
Mobile Content |
176x144, 10-15 fps |
50-60 Kbps |
Internet/Standard Definition |
640x480, 24 fps |
1-2 Mbps |
High Definition |
1280x720, 24p |
5-6 Mbps |
Full High Definition |
1920x1080, 24p |
7-8 Mbps |
Transport protocols overview[edit | edit source]
See the Wikipedia article on Video streaming. It explains several issues:
- Underlying protocol
- Datagram protocols (UDP) based solutions are simpler and more efficient, but there can be data loss.
- Transmission Control Protocol (TCP) based solutions guarantee correct delivery, however this can lead to timeouts and therefore needs clients to have a sufficient buffer (data stored before displayed)
- Streaming for multiple clients
- Multicast protocols send only one copy of the media stream over any given network connection (this is more difficult to implement).
- Peer-to-peer solutions will have clients sent data that they already have to other clients (that will save a lot of bandwidth but raises other issues).
- HTTP
- is not a streaming protocol, but can be used to distribute small files. E.g. as in podcasting
- Real-time Streaming Protocol (RTSP)
- allows a client to remotely control a streaming media server, issuing VCR-like commands such as "play" and "pause", and allowing time-based access to files on a server.
- Supported clients: Realplayer, MPlayer, Windows Media Player, QuickTime, Skype
- Real-time Transport Protocol (RTP) and Real-time Transport Control Protocol (RTCP)
- The Real-time Transport Protocol (or RTP) defines a standardized packet format for delivering audio and video over the Internet. It was developed by the Audio-Video Transport Working Group of the IETF. [1]
- RTCP partners RTP in the delivery and packaging of multimedia data, but does not transport any data itself. It is used periodically to transmit control packets to participants in a streaming multimedia session. Its primary function is to provide feedback on the quality of service being provided by RTP.
- Real Data Transport (RDT)
- RDT is RealNetwork's (RealAudio and RealVideo creator's) proprietary alternative to RTP and often used with the control protocol RTSP
A streamed content is delivered with a multimedia container format, e.g. QuickTime, Real Audio, AVI, ...
General, overviews and discussion[edit | edit source]
Streaming and container formats[edit | edit source]
- Streaming media systems
- BlindSideNetworks started the BigBlueButton open source project in 2007 to enable educational institutions and commercial companies to deliver a high-quality learning experience to remote students. There are integration modules for Moodle, Joomla, Wordpress, ... [2/2011]
- Ampache (wikipedia). Free opensource Web-based Audio file manager.
- ePresence an open source infrastructure for webcasting audio, video and synchronized presentation media and automated creation of navigable and searchable multimedia archives (special hardware and services are commercial)
- QuickTime Streaming Server (QTSS), Supports H.264, MPEG-4 and 3GPP and is included in MacOSX
- Unreal Media Server (commercial streaming media platform to be used for web casting, video conferencing, distance learning, video surveillance, Internet Radio/TV, ....). A free version is limited to 15 concurrent connections.
- Helix Community is a collaborative effort among both leading technology companies and open-source developers to extend the Helix DNA, the first open multi-format digital media platform. It includes Client/Player, Producer, Server software and supports a large range of datatypes (containers and codecs).
- Index pages Streaming media systems
- Bytes and Bibytes