Multimedia traffic is becoming more and more important in the Internet. Users want to have prompt access to interactive multimedia services like video or audio streaming. Existing transport protocols like TCP and UDP/RTP are not able to deliver the required service as needed. In addition to that Internet connections strongly differ regarding available bandwidth, delay and jitter. A new transport protocol is required to cope with these problems.
Four important features of future transport protocols have been identified in the past: tunability, adaptability, compatibility and flexibility. With these requirements in mind, a new transport protocol TPTR (Transport Protocol with Tunable Reliability) is defined. It consists of three sublayers: Application Framing (AF), Windowing, Reliability, Timing and Flow-Control (WRTF), and Congestion Control (CC).
The transport protocol has to mediate between the network characteristics and the application requirements. Each of the sublayers has specific tasks: The AF sublayer adapts data coming from the application to a format suitable for the network, and supports the assignment of priority values. The WRTF sublayer does the main task and decides which packet to send when a transmission is possible; basis of the decision are both importance (priority) and urgency (timestamp) of the data. Last but not least the CC sublayer is responsible for link estimation; i.e. it determines when the next packet can be sent.
In the book concepts are both newly developed, like the decodability mechanisms, and reused, like the packet structure of RTP. For the AF the concept of decodability has been invented. It makes use of structural information of a video stream to determine the priority of its different parts. Its key characteristic is the fact that it does not rely on the availability of the original video stream before encoding because this may not be available.