The Research And Design Of A VoIP System Based On SIP Protocol Framework

As the ever-growing coverage of the Internet and the booming expansion of the network devices are widely taking their places, the means by which data are transferred is now under the way of a radical reform. The traditional circuit switch is gradually giving its way to packet, and many services which were exclusive in the circuit switch are now available in the packet switch network, among which VoIP speech communication service is widely considered as the most typical one. This IP network based voice communication can provide not only very competitive price, but also a wide range of services and better connection to the new web technology.On the basis of much reference on the VoIP architecture and relevant protocols, this paper presents a solution and implementation of a VoIP system in Linux OS.This VoIP adopts SIP as its basic framework and employs several other protocol modules to guarantee the quality of service, such as RTP and SDP.On the whole, the VoIP is functionally designed into two part, namely the client and the server. The server is responsible for user localization and session initialization, the process of which is completed by using open-sourced SIP library OpenSIPS.The design of client also obeys the suggestion of the SIP framework. Light-weight protocols are used in application layer to build up a session and to strengthen the session management, and protocols in transfer layer, including the RTP and RTCP,are responsible for a better system QoS, meanwhile for a lower transfer delay and smaller communication cost, all the voice data are packed into UDP packets, whose connection-free transmission will help the system to realize the expected performance.The VoIP in this paper features the comparative low voice communication traffic and higher sound quality. To accomplish the voice sampling and playback part in the VoIP system, ALSA Linux sound architecture is chosen in the paper and its powerful function and easy-going interaction really make some parts of our coding much easier. And since the voice compressing and depressing algorithm is always of top priority in every VoIP, we finally decide to use the CELP in G.723.1 after a few comparison tests. Contributed much by the advanced sound driver architecture and voice compression method, our VoIP system has a prominent performance in sound effect. Shown by the tests and experiments, the VoIP system in this paper has largely met the requirement of practical use in both the aspects of communication delay and sound effects.