The Implementation And Optimization Of Voice Phone System Based On Mobile Internet

With the continuous integrated development of the mobile communication and mobile Internet,mobile service is becoming more and more close to our life.People may acquire information from the Internet anytime and anywhere.The total number of mobile Internet users in China has reached 946 million by the end of August 2015,which accounts for 69.5%of the mobile telephone users.The number of Internet users on mobile phones scales up to 900 million.A large number of applications and services based on mobile Internet are related to from the entertainment to daily social life.With the development of network technology and the increasing of mobile data traffic,voice communication service breaks through the limitation of traditional telephone service,and envolves towards the direction of high quality and low cost Voice over Internet Protocol(VoIP).But its performance depends heavely on the transmission condition and network load.Therefore,there are still problems under resolved.In this paper,the voice phone system based on mobile Internet is studied.The communication function of Multi-Rate Adaptive(AMR)is realized based on the open source software Spdroid.At the same time the telephone exchange server is set up.Then the end to end voice call system is built up.On the basis of this,improvements are proposed to ehance the performance of the communication system.An encoding rate adaptive adjustment algorithm is proposed to adapt to the real-time changes of the network conditions.Finally,tests have been carried to verify the performance of the voice phone system and the propsed encoding rate adaptive adjustment algorithm.The specific work is as follows:First,this thesis summarizes the current research status of voice phone technologies based on mobile Internet.Then the proper terminal software is chozed for development,as well as the voice phone system server.The principle of AMR encoding and decoding is also studied.Secondly,this thesis analyzes the design needs of the teminal software and architecture.Add AMR encoding and decoding module to Sipdroid,to achieving AMR voice call with Sipdroid under the Android platform.The voice phone switch server is Asterisk,which can support Session Initiation Protocol.The server is set up on the Ubuntu system,and a patch evaluated AMR codec into asterisk.Then the end to end voice phone system is successfully constructed.After that,the test cases and test plans are designed,to evaluate the performance of the propsed voice phone system based mobile Internet.Various tests have been accomplished,including the basic functions of the software,user-friendly,resource consumption,stability and voice performance testing.The results and analysis are shown to prove the voice quality.Thirdly,the existing AMR rate adjustment adaptive algorithm is studied.Then,the threshold-based encoding rate adaptive adjustment strategy of AMR is proposed,with parameters delay and packet loss rate.The encoding rate could be adjusted in accordance with the status of the network.Besides,other improvement to the terminal software is also realized,such as adding packet time,which can save the bandwidth during the call.Finally,the performance of the propsed AMR rate adjustment adaptive algorithm is tested and evaluated.