Research And Implementation Of RTP-based Real-Time Voice Transmission

In recent years, Internet has been widely used. Besides http,ftp and Email services, it has many new applications, such as multimedia communication. The common feature of these new services is the strong requirement of real-time reality. But the design principle for classical transport protocol decides that they can be competent for discrete media transmission, such as text,picture etc, but can't be used for real-time services such as audio,video and so on. Thus, in order to transmit real-time media with high performance over IP network, on November 22, 1995, RTP was approved by the IESG as an Internet proposed standard.First, the paper introduces the development of the real-time data transmission and the current research in China and abroad, then takes research on the characteristics of voice transmission over packet networks. In the second part, after introducing the fundamental features of RTP and its standard elements, the paper presents the analysis of RTP/RTCP in detail, including the protocol packets formats and various fields meaning. After that the paper introduces how to implement the system of real-time audio transmission based on RTP suite in Windows in detail, including the basic frame structure and the main data structures of the system. The main parts of this system are presented, including: how to collect and playout audio data; how to send and receive RTP packets; how to compensate for relative clock skew between sender and receiver, compensate for variation in interpacket timing caused by network queuing jitter and route changes, so to calculate the correct playout time. The system adjusts the playout buffering delay at proper time, to adapt network behavior change, this decreases the network jitter, and trys to find the trade-off between the voice delay and packet loss, to improve the quality of voice communication.To ensure certain QoS, the paper takes adaptive control algorithm for network audio transmission, which can reduce the network congestion. It is realized by estimating and monitoring the status of packets loss over a RTP-based network, and adjusting the rate of the output stream of the senders.