Voice over IP? How does it operate? How does it integrate with other traffic? How do we guarantee performance?

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Voice over IP: Operation, Integration with other Traffic, and Performance Guarantee
Abstract
Voice over IP (VoIP) transmits voice over packet-switched IP networks unlike traditional telephones operating through physical phone lines. The current paper reviews the operation of a VoIP system, how it integrates with other traffic, and how performance can be guaranteed by addressing major issues such as quality of service (QoS).
Introduction
The prevalence of the Internet and the capacity of Internet Protocol (IP) to connect almost all modern devices make Voice over IP (VoIP) a powerful technology for next-generation applications. Unlike traditional phones operating via physical phone lines, VoIP makes use of an IP to convey voice as packets over an IP network, which is often achieved on all major data networks using an IP such as the Internet, Intranet, Wide Area Networks (WAN), Local Area Networks (LAN), and Wireless Local Area Networks (WLAN) (Shaw and Sharma 16). In VoIP, voice signals are compressed, converted from analog to digital IP packets, before they are transmitted over an IP network (Ahmed et al 652). VoIP uses signaling protocols to set up and tear down calls, convey information that locates users, and negotiate capabilities. The technology offers an algorithm to transmit voice over IP networks and telephones and describes devices utilizing IP as the native transport solution for call signaling compared to traditional devices (Packer and Reuschel 48). The purpose of this paper is to discuss the operation of a VoIP system, the integration with other traffic, and how performance can be guaranteed by addressing the major challenges in VoIP communication such as QoS and security.
VoIP Operation
Voice signals can be transmitted through circuit-switched traditional telephones or packet-switched IP telephones, making VoIP one of the most important and attractive services in telecommunication networks, which can be implemented through two main architectures based on Session Initiation Protocol (SIP) and H.323 (Cruz et al 205). Signalling protocols play a critical role while relaying voice over the Internet as these power network components to talk to each other. In IP telephony, calls may be described as multimedia sessions between multiple parties. A connection is achieved when signalling is conjoined with a call and signalling achieves four major functions: session establishment, user location, user management, and session negotiation. VoIP operates through two major protocols as discussed below.
SIP
SIP refers to a communication protocol for signalling and controlling any form of multimedia communication sessions, including instant messaging (IM), online gaming, and an array of services depending on IP networks (Jalendry and Verma 162). The working mechanism of SIP is similar to HTTP web protocol because messages contain headers and the body. However, while the default port for HTTP is 80, SIP uses port 5060 as the default protocol for TCP or UDP and is the authority protocol for all voice and video telephony. SIP communication involves proxy servers, user agents (UAs), redirect servers, and gateways as main network elements. A proxy server is a mediator entity acting as User Agent Server (UAS) and User Agent Client (UAC) to raise requests on behalf of different clients. Proxy servers also perform routing during the transmission of jobs assigned to other entities adjacent to target users. UA generates or receives SIP messages and may act as UAC to transmit SIP messages when the receiver acts as UAS. Redirect servers, on the other hand, connect SIP session invitations to other outside domains while gateways connect SIP networks to different networks such as the public switched telephone network (PSTN) that employ different technologies and protocols (Jalendry and Verma 162-163). Together with SIP, H.323 makes VoIP systems operational.
H.323
H.323 are protocols, elements, and procedures that are specified by H.323 standard providing multimedia communication (video and data) across packet-based networks. Point-to-point communication services or multipoint multimedia is made possible by the H.323 system through the collaboration of four main elements, including multipoint control units (MCUs), terminals, gateways, and gatekeepers. MCUs manage multipoint conferences comprising of three or more H.323 terminals while terminals are real-time bidirectional multimedia communications enabled by terminals can be stand-alone devices or PCs running H.323 and other multimedia applications or software. H.323 gateways establish connections of H.323 networks to variant networks, including PSTN, Integrated Services Digital Network (ISDN), and H.323 systems while gatekeepers serve to address resolution and control admission (Jalendry and Verma 162). Different standards govern the functioning of the H.323 suit with the minimum requirement for the audio application being the G.711 or the 64-kbps channel and G.723 or 5.3 and 6.3 kbps channels for voice codec standards. For multimedia communication, H.245 control protocol is used during the initial deal between machines in finding the terminal capabilities, media channels, and audio encoding algorithm. Within the H.323...