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Transport Layer Considerations

Planning the contribution feed is of utmost importance to the creation of a stable and high-quality user experience in the live streaming environment. While layer-1 and layer-2 network services are typically bought in from a network services operator, there are many choices that the live streaming engineer (often called a “webcaster” in the context of streaming purely on the Internet) can autonomously make in the use of the layer-3 IP network services that he or she buys.

Continuing our journey “up the stack" we move up from the simple IP routing on layer 2 into the transport layer (layer 3), and we find the Transmission Control Protocol (TCP) and its twin the User Datagram Protocol (UDP).

UDP has a variety of good uses in audio and video streaming, and for many years UDP-based transport protocols for audio and video streaming were developed on the assumption that UDP would become the standard way to transmit audio and video.

UDP has no automatic retransmission process. So, if a packet fails to make it over the network link, it is up to the programmer to define when (or indeed if ) this should be corrected. In the case of a large quantity of data sent in an audio or video transmission, a few lost packets in a stream are generally not missed. The end user does not notice a few pixels of data not shown in a moving image - the eye and brain work together to correct this (known as “perceptual” audio and video encoding).

For many years the early streaming protocols were thus engineered with UDP transport in mind, and this led to a range of custom servers, called “media servers" appearing that specialized in packetizing encoded video into UDP datagrams. Often these media servers had separate control protocols that enabled client applications to communicate with the media servers to establish unique user sessions and subsequently control which video or audio stream is played. These application control protocols offered features such as pause, stop, rewind, and play.

 
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