Research And Simulation Of VoIP Processing Algorithm Based On AMR

Network voice is representative mark of Internet speech development,it can also be called the Internet Speech(VoIP-Voice over Internet Protocol)technology.VoIP realizes the transmission technology of pronctation communication based in IP packet switching networks,because of its cheaper price and kinds of functions,it has been widely applied.Although VoIP offers a variety of advantages in terms of cost savings and service improvements,but because of partly quality problems,the promotion has been less successful.In fact,between the traditional phone system and new VoIP system there are certain fundamental differences,the smooth and clog of the network inevitably cause delay of the speech signal,jitter,packet loss,and echo.The main reason affecting the quality of VoIP network is the packet loss phenomenon.At the receiving end,the real-time playback of the qulity voice of the loss and deterioration is due to the loss of a voice packet loss after the decoded speech.The thesis achieves the solution of the problem of packet loss through Adaptive Multi-Rate(AMR)speech coding technology and adaptive joint coding in VoIP system source channel.This algorithm uses dynamic network conditions to use Evaluation of E-Model to assess the quality of the speech model and then choose a way to achieve the best voice quality source and channel coding rate in selecting the channel coding of the Reed-Solomon of error correction code system.This thesis designs an adaptive algorithm for VoIP system source channel rate,using the AMR speech coding technology and FEC channel coding basing on the RS code.This kind of algorithm judges the quality of the voice from packet loss and delay by E-Model evaluation model,according to the changing network adaptively selecting the optimal voice source and channel coding scheme.For this algorithm,we solve the source-channel rate and forward error correction coding to make the voice quality under different network conitions.If consistent with the assumed outcome,then it proves its effectiveness.The experiment simulates the changes of the network,carrying out in NS2 simulation environment.In VoIP systems,combining with the source and channel rate adaptation algorithms and comparing to VoIP system with a fixed rate G.729 encoder and the AMR encoder of VoIP systems CITIC adaptive source rate.Experiment showed that this algortithm designed for the influence of packet loss and delay are different degrees of decline.Especially when Network Congestion,the effect is more pronounced.