Performance evaluation of Voice over Internet Protocol
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Voice over Internet Protocol (VoIP) was developed to emulate toll services with lower communication cost. In VoIP applications, voices are digitized and packetized into small blocks. These voice blocks are encapsulated in a sequence of voice packets using the Real-time Transport Protocol (RTP) and delivered by the User Datagram Protocol (UDP). To help VoIP applications deal with unpredictable network performance, the Real-time Transport Control Protocol (RTCP) is developed to monitor the performance of RTP packets and provide feedback to the VoIP applications. The feedback on packet delay, jitter, and loss rate enables the applications to adapt to network conditions to maintain a certain level of voice quality. With this architecture, the quality of service of VoIP relies on the effectiveness of the RTCP network performance report mechanism. This research collects RTCP performance reports from live traffic over real networks and compares their values with the statistics derived from direct measurements of RTP packets to evaluate the effectiveness of RTCP. The live experiments were conducted on networks resembling respectively, Local Area Network (LAN), Wide Area Network (WAN), campus network, and encrypted wireless LAN. Results from these experiments show that RTCP is effective for low delay networks but RTCP performance reports can be inaccurate for networks with large, volatile delays.
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