Deploying VoIP in Existing IP Networks

Salah, Khaled

doi:10.1201/9781420070217.pt1

Cited by 2 publications

(2 citation statements)

References 23 publications

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“…In this research a performance characteristic for encoding and decoding the side of PCM channels applicable to voice frequencies (G.714) is deployed. It requires a 150 ms maximum end-to-end total one-way packet delay for VoIP applications [18,31,32]. It was decided to put up with delays of up to 200 ms.…”

Section: Voip Traffic Haracteristics Conditions and Assumptionsmentioning

confidence: 99%

Performance Evaluation of VoIP Analysis and Simulation

Adjardjah,

Kumassah,

Abdallah

et al. 2023

JERR

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The use of technology has impacted on communication in so many ways. The advent of voice over internet protocols (VoIP) has made the world a global village where one can reach out to any part of the universe. But a challenge exists as to how to make communication and data transmission faster, the volume of traffic, bandwidth and latency in networks, that has to be transmitted between the sender and the receiver. The overall customer experience can be improved by the use of technology, which also makes it simpler to collect client information. Data packets are addressed and routed by the Internet Protocol (IP). This research aimed at deploying jitter, throughput, network traffic delay and bandwidth (JiTTraB) as a performance metrics to measure voice over internet protocols (VoIP) to measure the Quality of Service (QoS) of networks. This method prioritizes network traffic going through a router and providing acceptable service to most users in a quest to address VoIP concerns. In comparison to an existing university campus network (UCN), JiTTraB has a higher throughput, more data transmission at a given time with a minimal delay of about 0.001 seconds. However, UCN queues data with a high rate of delay before transmission. Data seem to crash considerably on the UCN due to network overload above its throughput margin and growth capacity. Thus, the proposed network design based on JiTTraB outperforms UCN in terms of transmission rate, throughput, bandwidth, delay and jitter. In addition, the JiTTraB communication network enhances QoS by performing better in close-quarters conversations than in far-quarters conversations. With an expansion reserve of 30%, the JiTTraB can handle up to 350 conversations securely while maintaining VoIP QoS standards without degrading the functionality of other network services or applications.

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Section: Voip Traffic Haracteristics Conditions and Assumptionsmentioning

confidence: 99%

Performance Evaluation of VoIP Analysis and Simulation

Adjardjah,

Kumassah,

Abdallah

et al. 2023

JERR

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“…However, VoIP has time-sensitive traffic, which exhibits a bounded quality of service requirement, such as delay, jitter, and packet loss [9]. It is crucial to verify that adding an extra layer of security on the VoIP packet does not degrade the QoS requirements.…”

Section: Introductionmentioning

confidence: 99%

VoIP Codec Performance Evaluation on GRE with IPsec over IPv4 and IPv6

Alausa¹,

Arekete²,

Odim³

et al. 2021

Adv. sci. technol. eng. syst. j.

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Scientists succeeded in implementing conventional public switch telephone network (PSTN) into internet protocol by launching H.323 IP telephony. The irrelevant and unknown captions in H.323, computer scientists have replaced H.323 by Session Initiation Protocol (SIP) for Voice-over-IP (VoIP). However, the security of voice communication over IP is still a major concern. Besides, security and performance contradict features. VoIP exhibits a quality-of-service requirements that are sensitive to time. Example of such QoS requirements are delay, jitter, and packet loss. Integrating Internet Protocol Security (IPsec) with Generic Routing Encapsulation (GRE) encrypts and authenticate packets from the sender to receiver, but that raises the question of performance as VoIP is time sensitive. Consequently, three codecs were evaluated to determine the efficiency of each on GRE and IPsec implementation on Internet Protocol version 4 and Internet Protocol version 6 (IPv4 and IPv6), respectively. The topology design and device configuration in this study adopted Graphic Network Simulator 3 (GNS3) and Distributed Internet Traffic Generator (D-ITG) to generate VoIP traffic. The evaluation revealed that the G.723.1 codec achieved better results on IPv4 and IPv6 over GRE with IPsec than other codecs used in the experiment. Furthermore, the codec of choice is a major factor in IPsec VoIP deployment, as also revealed in this study.

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Deploying VoIP in Existing IP Networks

Cited by 2 publications

References 23 publications

Performance Evaluation of VoIP Analysis and Simulation

Performance Evaluation of VoIP Analysis and Simulation

VoIP Codec Performance Evaluation on GRE with IPsec over IPv4 and IPv6

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