Performance Improving Technology for VoIP Wireless Transmission Based on Packet-Level Reed-Solomon Codes
|Course||Communication and Information System|
|Keywords||VoIP Wireless transmission Performance Improving Packet-level Reed-Solomon NLMS SER|
VoIP (Voice over IP), also known as Internet telephony, it transmits voice streams over the TCP/IP network. It not only costs inexpensive but also provides better service with good voice quality. With the telecom operators to continuously push forward the construction of the4G network and WiFi network, flow business is expanding. This indicates that the flow business of telecommunications business will be more popular in future, and it is obvious that VoIP meets the trend. With the continuous development of wireless communications technology, VoIP will be bound to be part of our lives.In the wireless network, the interference, time out and congestion usually lead to packet losses. The lost packets usually are not retransmitted due to real-time service requirements of VoIP. Hence, the transmission of voice stream goes through a serious packet loss in VoIP, causing gaps in the IP media stream. Group packet loss of information not only directly reduces the voice quality, but also causes the spread of mistaken frame because of the correlation between the adjacent frames in the phonetic source encoding. The best-effort heterogeneous packet networks and wireless systems during a deep fade have transient shutdown.Firstly Signaling protocols and media transport protocol and other protocols for the VoIP wireless transmission are described and the main factors affecting the voice quality of VoIP and QoS of VoIP are presented. Particularly, the important compression coding scheme G.729and a new compression coding scheme iLBC, in VoIP wireless transmission, are compared.Secondly, by the structure of VoIP wireless transmission, we build a VoIP radio communication system which can be used in the practice:MESH routers constitute the backbone network and access network of VoIP wireless transmission system, and Asterisk servers make up the SIP signaling server and the softphone Linphone serve as the terminal. It is Confirmming that the serious packets loss will cause the decline of the voice quality basing the system, and the simple techniques such as redundant coding with low bit rate and interleaving technique for reducing the packet loss are put forward.Finally, the Girbert model, wireless packet loss channel, is erected for the phenomenon of packet loss in the VoIP wireless transmission system. Basing on the packet loss channel model, the more in-depth analysis and study of forward error correction PRS(Packet-level Reed-Solomon,PRS) codes is given in the Part4. Especially, the algorithm of the PRS codes in this paper not only gives the standard PRS decoding algorithm and the truncated PRS code decoding algorithm. In addition to the algorithm, th e PRS codes selection recommendations are given for the sudden loss channel. Such performance improving programs, the redundant encoding with the low bit rate and interleaving techniques and FEC PRS codes, are not dynamically adjusted according to the channel packet loss in a timely manner. For this reason, the adaptive algorithm of the PRS codes is advised in this paper. Other authors have studied adaptive algorithm such as NLMS for PRS codes. In this paper, SER algorithm is the first time to be introduced for forecast of the PRS code redundancy. Over the same packet loss channel, it is proved by the mean of simulation that PRS code with the SER algorithm outperforms with NLMS algorithm.