Design And Implementation Of VoIP Video Server

VoIP (Voice over Internet Protocol) is the business which IP network is responsible for transmission of digitalized voice signals. With the booming development of the Internet, it’s rising rapidly since the last decades. Then VoIP has been widely applied all over the world, which is also having a great potential for development. The biggest advantage of VoIP is that it provides better and more services via the vast Internet. Consequently, IP telephony comes into being and has become the direction of "next-generation phones". And with the increasing needs of users, VoIP has no longer been limited to voice business, but also gradually with video business added. However, the Internet-based transmission of multimedia has a definitely strict demand on real-time, which traditional Internet transmission business can’t afford. Under such circumstances, IETF (Internet Engineering Task Force) has developed a protocol named RTP (Real-time Transport Protocol), which is used for completing bi-directional multimedia communication much better by interacting with other transmission protocols. RTP can also avoid the shortcomings of the traditional real-time voice communication, reduce network delays and improve voice transmission effects at the same time.In this context, this paper relies on author’s involvement in the practice of "Multimedia emergency dispatching platform". According to the actual needs of the customers, the author did lots of work about the expansion of the video capabilities on the basis of the original voice functions. In the theory-study stage, the author did lots of research about session establishment based on SIP (Session Initiation Protocol), RTP encapsulation of H.264, network transmission of video data, network congestion control, and video storage etc. In the system-design stage, the author adopted a thought of module partition and divided the video sever into five independent parts to finish designing. In the system-implement stage, the author finished all of the functions by coding. In the system-testing stage, the author finished both unit-testing and integrate-testing, finally evaluated the functions and performance of the server.This article’s main work is as follows:1. Introduction and analysis of VoIP-related protocols, including SIP and RTP. And analyzed their characteristics in depth.2. Designed and implemented video server according to customer’s requirements in Linux environment. And designed a congestion control mechanism especially for disaster-monitoring and other special occasions.3. Tested the functions and performance of the video server. And figured out some indicators of the server through the statistics and analysis.4. Proposed the lack of this paper and outlook on future research at the end.