| IP telephone is widely used in the domestic and internal long distance call currently and became the main service of telecommunication service company. But IP telephone has much problems in communication quality. So how to improve the quality of IP fragment voice correspondence is a hot research direction. This paper is written based on the correlative technology of gripped in my work and the founding practical problem of IP telephone, and reading much VoIP technology and theory books. The aim is to use new technology theory creating new transmission pattern of real time data transmission in internet and increasing the quality of transmission , solving the quality problem of IP telephone at last.QOS is a subjective measure, different people has different reflection to telephone tone .In order to calculate a quality of technology, the main method used is MOS appraise. MOS≥4.0 is high quality audio frequency, MOS≥3.5 is communication quality. The VoIP telephone network Qos of China Satellite Communications has such problems 1) echo 2) dithering 3) intermittence 4) falling line , these problem affect the QOS of Service.This paper analyse these problems:digital voice coding, data fragment,IP packet encapsulating, IP data packet transmitting in Internet, transmition circuit, voice impropriating bandwidth, the max bandwidth of ChangChun, Though these getting the result of influence Qos. Some policies about voice coding, data fragment and IP encapsulation are not fit for VoIP data transmission in the Internet, but these are not main reasons .The main reason influence Qos of VoIP in China Satellite Communication are delay ,packet loss and route disposal in network. IP network initial aim is to offer a Best Effort service.To UDP protocol is a connectless system. This mean that Internet is not to create router table to save connecting state because it is not to create connect at all. In telephone network the caller and the user who is called should create a fixed gateway to connect. However,Internet is a data network of connectionless and adapting route itself. So delay and packet loss must happen. So this Qos is not to be assured.Delay is very important, because it affect voice quality of telephone directly. When the delay exceeds 250ms,communication quality become very bad. Too long delay can bring on voice overlap, echo, dithering, not synchronization and information loss. because late packet can not be used by digital- simulation instrument.Fragment loss has great affect to voice quality. When packet loss percent less than 10% at current voice coding instruction, the simply way is to insert final received packet at the interval between the loss packets. Through this way, still can resume accepted voice semaphore. Otherwise, voice semaphore would be damaged and cause to appear very bad Qos.The solved method of DiffServ absorb the virtues of IntServ in technology and discard the disadvantage of IntServ, it also classifiedly dispose to fragment. Because DiffSer is still in the development course, so research it can reduce the difference of computer network theory and technology between our nation and foreign, and has important meaning. Creating PHB at each router. Detailed policy is:To multimedia and video frequency data transmission, adopting RPT bearing technology differentiate multimedia video frequency packet and sound frequency, using upper priority level assure Qos of multimedia sound data packet, and according to function value in PT domain can distinguish the type of packet.Commutate(reshape) data stream on the gateway equipment of sending terminal with fixed-size speech packet the same as AAL 2 in ATM network,confirm the size of speech packet(10 bytes) according the size of compressed IP/UDP/RTP packet header (9 bytes). There are 19 bytes with compressed packet header in total, 41 bytes less than primary 60 bytes, of which voice packet has decreased 10 bytes.A half bytes of IP/UDP/RTP packet header are unaltered bytes during the whole period of connecting, only with different rank... |
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