Design And Implementation Of Cross-Platform Voice Communication Terminal Based On SIP

With rapid development and popularity of internet, people have more and more demands for internet communication services. Multimedia applications take a leading role in network communication services, in which VoIP (Voice over IP) technology is particularly prominent. Signaling is one of the key techniques for VoIP. The most popular VoIP signaling protocol is Session Initialization Protocol (SIP) and H.323. Session Initiation Protocol, which is defined as signaling protocol in 1999 by Multiparty Multimedia Session Control (MMUSIC) workgroup of Internet Engineering Task Force (IETF), is widely used for controlling multimedia communication sessions such as voice and video calls over Internet Protocol (IP). The protocol can be used for creating, modifying and terminating two-party (unicast) or multiparty (multicast) sessions consisting of one or several media streams. In recent years, SIP, with its simplicity, open, flexibility, extensibility, and other advantages, has been more and more widely used in VoIP applications.The thesis has a deep research and analysis about SIP protocol and related extensible protocols according to RFC3261, at the same time, it has researched various modules such as SIP transaction message parser of PJSIP open source SIP protocol stack. With the principle of cross-platform software design we implement a cross-platform SIP protocol stack based on PJSIP. This protocol compiles into the corresponding Python module according to different operating system, and it has responsibility for the low level functions such as SIP protocol parsing and SIP transaction processing. Combined with Python virtual machine and PyQt4 user interface framework, we design and implement cross-platform SIP voice communication system. The design of client has used the typical MVC (Model-View-Controller) design pattern so that every module of the system has more clear level and has good extensibility.The functions of this SIP voice communication system include SIP account registration, voice communication, instant message coummunication, state presence, management for account and buddy and so on. The user interface of system has implemented the function of adding account and buddy. The system has capacity for swithing among different accounts. Software terminal can choose the online state from five states. The main interface of software terminal can display the list of buddy. One can have a call or send instant message by choosing a buddy of the list. The status bar of the main interface displays the state infornation of the current user, such as registration successed,registration failed and wheather is not in the calling state. Combined with extensive applications and services of internet email, the SIP voice communication system integrates feature service for sending messages and file attachments to buddys via email besides voice communications and instant messaging service, which generates a new SIP+Email internet communication service mode.Tests under Ubuntu Linux, Windows XP and Windows 7 operating systems demonstrate that the SIP voice communication system works stablely on many operating system platforms. The system also has good extensibility and we can add more functions to it in the future. Another voice communication test with other SIP clients such as Yate Client shows that the SIP voice communication system has good compatibility as well.