Performance Metrics for SIP-Based VoIP Applications Over DMO

Performance Metrics for SIP-Based VoIP Applications Over DMO

Mazin I. Alshamrani (Ministry of Haj and Umra, Saudi Arabia) and Ashraf A. Ali (The Hashemite University, Jordan & University of South Wales, UK)
DOI: 10.4018/978-1-5225-2113-6.ch004
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Abstract

The evaluation studies need to investigate a determined performance metrics to understand and evaluate the examined scenarios. SIP-based Voice over IP (VoIP) applications over MANET, which behaves in a way similar to Direct Mode of Operation (DMO) in mission Critical Communication Systems, have two main performance categories related to the Quality of Service (QoS). The main performance metrics that are considered for the evaluation processes in this research are the SIP end-to-end Performance metrics as defined by the RFC 6076. The main performance metrics are related to the registration, the call setup, and the call termination processes. In this research study, the SIP performance metrics are based on a single SIP proxy server. For voice data, the QoS evaluation is based on two methods: The Objective method and the Subjective method. The Objective method considers the traffic throughput, end-to-end delays, packet loss, and jitter, while the subjective method considers the Mean Opinion Score (MOS), which is mostly related to the end users' experiences during voice calls.
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Voice Codecs

The voice applications used to compress the analog voice signals into digital signals uses different types of voice codecs. Voice codecs are audio data compression algorithms for use for different types of voice based applications (Ganguly & Bhatnagar, 2008) . This basic stage happens on the caller’s side to make the voice data transferable over the PSTN or the Internet for far distances. For wireless based VoIP applications, the voice compression is critical, as the voice signal needs to be compressed as much as possible to fit with the loose nature of wireless communications. This compression effectively reduces the bandwidth consumption and transmission power over wireless network systems. In addition, the voice compression systems create smaller packets, which reduce the packet loss ratio, and end-to-end delays that support the voice quality as the number of received voice packets relatively increase (Sinnreich, & Johnston, 2012). The present researcher studied the SIP-based VoIP applications over MANET using four common voice codecs:

  • G.723.1 (ITU-T Rec. G.723.1, 2006) is one of the most common voice codecs for VoIP applications that operates at 5.3 Kbit/s or 6.3 Kbit/s and is officially known as Dual Rate Speech Codec for Multimedia Communications Transmitting at 5.3 and 6.3 Kbit/s.

  • G.729 (Schulzrinne, Casner, Frederick, and Jacobson, 2003) is another common voice codec for VoIP applications because of its low bandwidth requirements. It operates at 8 Kbit/s and formally known as Coding of Speech at 8 Kbit/s Using Conjugate-Structure Algebraic Code-Excited Linear Prediction Speech Coding (CS-ACELP).

  • The GSM voice codec is developed by the European Telecommunication Standards Institute (ETSI). It is widely used in mobile telecommunications as it operates at 13 Kbit/s and has good performance over CPU demands that support the nodes’ mobility nature (GSM, 2000) .

  • G.728 (ITU-T Rec. G.728, 2012) is a speech coding algorithm which operates at 16 Kbit/s and is described by the International Telecommunication Union – Telecommunication Standardization (ITU-T) as the Coding of Speech at 16 Kbit/s Using Low-Delay Code-Excited Linear Prediction (LD-CELP).

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