RTP-Based Implementation Of Real-time Speech Transmission System On Linux

VoIP is a new fast-growing technology to transfer real-time voice over IP networks. It has the greatest advantage in fully merging into the global IP interconnect environment to supply various kinds of service in low cost. This technology is one of the great supplements for the traditional telephone service which further prove that not only data service like text, picture, but also audio and video service can be transferred through internet.As the IP network is initially designed to transfer data like text and picture, it seems inefficient to transfer media like audio and video data real-timely. It has turned to be a critical point on how to transport real-time media data in high QoS through internet.The speech signal is transmitted by digitising tiny pieces of it at regular intervals and sending these to the destination where an analogue signal is reconstructed. For good quality communication, the overall delay should be below 200 ms. Delay variance or jitter should be eliminated through buffering. Speech communication is fairly tolerant to lost or corrupted packets.As part of the research work—Implementation of Intellectual IP Telephone Terminal on Linux, the author has deeply analyzed the related aspects like delay, jitter, and packet lost in the real-time transfer of voice on IP networks, brought forward some measures to improve on each aspects and also realized a real-time multiple voice system based on Linux.The article first introduced some background knowledge of real-time voice transfer, then point out the necessity of RTP/RTCP protocol for transferring voice due to the inefficiency of the TCP counterparts. Then it gave an exhaustive description on RTP/RTCP protocol. Afterward it provided the software impletementation of the voice transmission system through RTP/RTCP protocols upon Linux. The system is implemented through the technology of multi-thread, multiple ring-buffer and UDP Socket transport technique. Finally the author provided the test results, which demonstrates that the system is rightly designed and runs well for transferring real-time voice in high quality. At the last of the article, the author gave some perspective views tailor to problems which exists on large-scale RTP/RTCP application scenery.