Research And Simulation On Packet Loss Processing Algorithm Based On AMR In VoIP

VoIP (Voice over Internet Protocol) is a general term for a family of transmission technologies for delivery of voice communications over IP networks such as the Internet or other packet-switched networks, which has been widely used as its low price and rich features. An important characteristic of voice communications is that the transmission must satisfy the real-time and interaction requirement, while the IP networks were originally designed for the data communication with the best-effort service. Routing, latency, congestion or other errors in the network can cause the loss, delay or disorder of the voice packets. And the packet loss is the key issue of the voice transmission over IP network, which has a significant effect on the voice quality. Hence, more and more research opportunities in VoIP are emerging in the related issues on anti packet loss and packet loss processing.With the development of speech coding technology, variable bit-rate speech coding provides an efficient means to reduce the packet loss in VoIP. The variable bit-rate speech coding technology can adjust the encoding rate adaptively according to the network conditions, which simultaneously reduce the packet loss and guarantee the voice quality, resulting in a widespread use. The AMR (Adaptive Multi Rate) speech codec, based on the variable bit-rate speech coding technology, has a more intelligent way to solve the problems of resource starvation and network congestion, improving the voice quality obviously. It has an even broader prospect for development.A source and channel rate adaption algorithm based on AMR codec for VoIP is proposed, which carries out packet processing issues in a novel way. The adaption algorithm combines the source and channel coding to reduce the packet loss. According to the network conditions, taking packet loss and delay into account, the proposed algorithm selects the best source and channel coding solution which reaches the highest estimated voice quality. In addition, the channel coding is implemented by a Reed-Solomon code based FEC (Forward Error Correction) scheme, and the estimated voice quality is obtained from the E-Model evaluation considering the packet loss and the delay.An analysis is carried out on the source and channel rate adaption algorithm, and the results show the validity of the algorithm. In the NS2 and PESQ based simulated experiments, a VoIP system which adopts the proposed adaption algorithm is designed. Comparing with the other two VoIP systems which are based on AMR codec and G.729 codec respectively, the adaption algorithm based VoIP system performances better in packet loss, average delay and speech quality.