Voice over Internet Protocol (VoIP) Performance Analysis, QoS Measures to Minimize Packet Loss, and Identifying Link Failures during Transmission

摘要

VoIP is the recent step of telephony evolution. VoIP stands for Voice over Internet Protocol packets. Voice over the IP network is a general term that refers to any means of converting voice calls into voice data packets that are transmitted over an IP network, either public or private. PSTN is circuit switched and VoIP is packet switched. In circuit switching irrespective of amount information to be sent, full bandwidth is reserved. In VoIP, distance between sender and receiver is not based on geographical distance. The transmission time depends on bandwidth and dynamic routing algorithms adopted. VoIP facilitates voice communication over Internet. One of the problems using VoIP is latency. When packets are sent in different routes, all the packets may not arrive at the same time. This causes latency. To retransmit/download a message all the packets are to be received. Even if one the packets are missing, complete message cannot be formed. This is not the case with PSTN. To overcome this problem, and to make transmission as efficient as PSTN, we adopt dynamic source routing algorithms. Secondly, potentiality decreases with the degradation of QoS. Some of the packets may be lost due to timeout or lack of buffer space. This cause erroneous data received and also requires the retransmission of the same message again and again. In this paper, we present generic algorithms to reduce latency and dynamic routing algorithms to minimize packet loss. Keywords: VoIP, Latency, QoS, Dynamic Source Routing algorithms