Requirement Analysis, Design and Implementation Plan
Voice over internet protocol (VoIP)
VoIP is a communication protocol that is
available over a network. The IP network allows clients to make telephone calls
by using VoIP technology. VoIP is the transmission of voice signals over
Internet or private networks by means of the Internet Protocol (IP). More
simply, VoIP, which flows over the internet, converts user’s voice into digital
signal. Over the past few years, VoIP and Internet usage have risen
substantially. The concentration of the latest web facilities is on speech
interaction over IP packets. VoIP is accessible on all networks “that use
IPs such as Internet, Local Area (LAN) networks. Customers can interact with
the internet and use their Internet connection to generate phone calls.Flexibility
and cost efficiency are the main elements that encourage companies to move on
to VoIP. Some safety issues may occur because of the extensive VoIP
implementation. Voice over IP (VoIP) offers interactive communications
facilities such as video conferencing and voice interaction.VoIP supports
transfer of data that is difficult to transfer over cable and wireless
Table of Contents
Voice over Internet Protocol (VoIP) is a transfer protocol allowing anyone the access for generate calls via telephone across the web.Access to the usually VoIP allows users to call others receiving Internet calls. Users can also create and receive calls with VoIP interconnections, from and to the standard fixed line charges. Certain VoIP services requiring a computer and a dedicated VoIP telephone have the requirement of a type of adapter. VoIP can also be defined as a separate alternative that allows speech signals to be transferred via web instead of a traditional cell line.The VoIP applications currently test and encode analog speech signals via codec, then contained in an IP packet and transferred via information wires or the Internet infrastructure via information packets.Previously, VoIP needed a headset to be connected to the computer and only other persons who had similar set-up could be encountered from the speaker and receiving person. They must tell each other in advance, to advise the client at the other end of the call and the call period.
The VoIP technology was not fully created in its early phases and numerous loopholes existed. It is possible to conclude that technical scarcity prevented any significant developments or modifications in VoIP. The marketing framework and technological reality were very much different. Nevertheless, technological and viable progress has been made recently by VoIP. In order to set up and decommission calls, to transmit information required for location and negotiation customers, signalling protocols are used. One of the primary benefits of VoIP is that it makes long-haul calls at small rates including calls to other nations that are able to use the same figures in different fields around the globe.
The VoIP system is equipped with many advanced features, making the VoIP system the best approach to a standard network (Singh et al., 2014). These features attract VoIP telephony from the communication industry and the business community. The short description of the advantages of VoIP telephony are listed as below:
As more calls are made on the traditional network, VoIP network can integrate with the traditional PSTN scheme. Signalling protocols such as H.323 and the initiation session protocol allow very easy communication from a VoIP scheme to PSTN.
The rise in connectivity in traditional devices, which implies offering more ports to link phones, contributes to price increases. However, no additional cost to boost connectivity would be covered in the VoIP system, as costs relating to functionality improvements increase or decrease (Luhach et al., 2019). Because VoIP is based on software, any network update add-on is therefore easy. The cost of hardware is also very low because it is PC-based
The main benefit of the VoIP system is the low cost. Long-haul calls may be made via the VoIP system at very low costs, as the access to the public Internet and the cost-effective broadband links enhance the adaptation of the VoIP system in the worlds of small business. The computer system and Internet are the existing facilities for providing communication with no additional cable access costs (Shaw & Sharma, 2016). VoIP addresses voice as well as any other data, enabling users to attach voice messages documents and make videoconferences.
2.4 Recovering disaster
Networking in VoIP is much more aware of the inability of the call trajectory as numerous parts, such as routers, buttons, etc., are linked to each other following the physical route. It eliminates failures and offers alternative forms of connectivity maintenance.
Virtual private network (VPN) is a bandwidth assigned to the Internet where encryption prevents public access. Utilization of VPN makes the network more secure (Kambourakis et al., 2015).
In an implementation loss network it is not possible to read disturbance packets because of the demand for the organized data transmission and TCP transmission mechanism. This increases the length and jitter of speech information, causes audio packets to lose their standard play time and causes a delay and call interval to dramatically impact communications performance. A fresh transmission approach that enables unbalanced TCP data transmission is thus very important in order to decrease retransmission time and ensure speech performance in a low-loss packet frequency network (Kai et al., 2016).In addition, since TCP is a string-oriented byte protocol, the limits of the user data packets cannot be identified. As a result, a disorderly TCP approach may contribute to a packet boundary failure and the recipient cannot recover the initial spoken command. The decoder doesn’t work that way.Therefore, an algorithm must also be created to identify voice signals acquired by data flow and then delivered on time to the voice audio handling unit.
Two of the main methods of improving bandwidth use are VoIP: compression of the packet’s header and multiplexing of VoIP packets (Abualhaj et al., 2019). Compression of the VoIP packets is the 40 byte RTP/UDP/IP header reduction method dependent on certain proprieties of the protocol areas of the header. Multiplexing VoIP packets is the method by which packets distributed in one header are combined.
Compressed header method compressing the RTP / UDP / IP header of 40 bytes in 2 or 4 bytes only. To accomplish this elevated header compression, the utilization of two RTP / UDP / IP characteristics is carried out. First of all, some of the RTP / UDP / IP protocol areas have been corrected and all the packets have the same effect. These areas are therefore sent and deleted from all packets during the session build-up. The other feature is that the proceeding parts increase certain of the fields of RTP/UDP/IP protocols by fixed value. Therefore, the differential coding is used for compressing these fields.
In addition to header compression, a multiplexing method has been proposed combininga number of packets of VoIP within a single header of UDP/IP, at the same time the protocol concerned with the RTP of each of the packetis unchanged (Abualhaj et al., 2016). The suggested technique multiplies packets from a particular VoIP gateway and which are intended for the same VoIP gateway.
Jung et al. (2017) also suggested a techniques which is a combination of two separate methodologies/techniques, a VoIP packet header compressor, and VoIP packet header multiplexing. Multiplication of packets that originate from a specific VoIP password and are designed to be used on the same VoIP gateway in one UDP / IP header takes place.
The International Telecommunication Union (H.323) Protocol contains the family of protocols used for call, terminations, call recording, authentication and other functions, as recommended by the International Telecommunications Union (Shaw & Sharma 2016). The transmission of the Family of H.323 protocols takes place through the utilization of connections of UDP or TCP. The protocol family H.323 consists of H.225, for recording, acceptance and signing of calls. The use of T.120 is carried out in holding a conference in which a common whiteboard application has been applied, to organisation and monitor sessions in the media. H.323 audio codecs is defined in G.7xx series and video codecs are defined in the H.26x series. For media transportation, H.323 uses RTP and RTCP uses for control of RTP sessions.
“Protocol concerned with the SIP to initiate an interactive modifying,user sessionas well as session termination involving video, voicemail, online playbacks, instant messages, and multimedia of other terminals.
Components, developed by the Institute of Engineering in Electronics and Telecommunications (IETE). Audio and video conferencing interactive sessions and interactive gaming can be established via IP networks. The basic IP services are integrated with Web and chat services by these enabled services providers.SIP usually uses Port 5060 as the default TCP or UDP protocol (Shah & Dave, 2016). SIP can be construed as an authorization protocol for voice / telephony and video-over-IP (VoIP) services. It is similar to the HTTP web protocol as messages contain headers and a body of messages.
Audio / Video Conference and Interactive Gaming sessions can be established by SIP over IP networks. These enabled service suppliers incorporate Web and Chat facilities into fundamental IP telephone facilities.
SIP allows the communication of voice-data in real time through two RTP / RTCP protocols and the utilization of SDP is carried out for negotiating participant capabilities, the type of codification, etc. (Gongjian, 2016) A solution to the complicated protocols H.323 is provided by the use of SIP.SIP is more common because of its simpler nature than the H.323 protocol family.
The Media Gateway Control Protocol MGCP is used to interact between the distinct VoIP gateway modules. The modern digital service of communication offers an internet telephony service provider service. This specific service is utilized within the several protocols such as the Media Gateway Control Protocol (MGCP), Session Initiation Protocol (SIP), H.323 protocol and so forth (Paul et al., 2017). In addition, the MGCP is an additional SIP and H.323 protocol. It is more often referred to as the “call agent” is compulsory, since it handles and promotes the facilities offered in MGCP calls and meetings. The endpoint of the MG (the media gateway) is unaware of calls and lectures and therefore does not keep the call states. MG orders are executed which the call officials send. MGCP implies that the call officers are synchronized with each other and send consistent control orders to MGs.
The major questions of QoS related to implementation and safety of VoIP technology are: bandwidth, network delays, delays and loss of traffic. “A medium frequency transfer (optical fiber, coaxial cables etc.) describes its capacity for data transmission in bits/s. The path of network which comprises of a number of LAN and wireless connections is the equal bandwidth to the slowest route. A bottleneck is usually referred as the bottom bandwidth network in the network path. Bottlenecks in the network cause congestion that leads to QoS problems related to speech traffic (Kolhar et al., 2018). Congestion should be avoided for a correct transmission of voice over a network of IP, thus making the improvements for using a better system of VoIP.This could be carried out in different ways, including by increasing bandwidth, prioritizing traffic and traffic compression.
Network delays are the numbers of times that the packet needs for travelling with the beginning point of source via the network to a destined location, known as latency. Latency is the most important part of its processing time, length, and time to serialize and spread time. (Estepa et al., 2018).
The Router requires time for receiving a filethrough the interface of input and enter into the control panel’s queue. The handling period primarily relies on the router’s design and the processing time of the router.
The time a packet has been placed in the yield queue of the router. The queue time is entirely dependent upon the rate of processing, the load bounded, output range and the mechanism of queuing which occurred because of the bottleneck.
The time to place a bundle for transport on the physical medium.
The amount of time a signal takes for a media to transit. It relies on the medium form and the signal form that the information transports (Ismail et al., 2017).
Various approaches were regarded to minimize network delays via an IP network so that IP technology can help application with stringent delays in real time. Network delay with the same policies for raising the bandwidth accessible can be minimised.
In the advent of transmitted packets, jitter is described as a variation. The packets are transmitted on the sending side with packet spacing evenly in a continuous stream. This continuous flow can become irregular under bottleneck circumstances, since the distance between each packet differs in lieu of being continuous (Roy et al., 2018). In the face of the annoying effects of Jitter, de-jitter or delay buffering QoS mechanism was considered.The de-jitter buffer system, implemented on the input interface for the receiving end, depends on a particular buffer recognized as a de-jitter buffer to slow down the obtained packets and place them appropriately before they are placed in a constant stream such as that transferred. While the de-jitter system helps to avoid the jitter effect, it impacts the general network delay.
Network congestion is the primary cause for packet loss via an IP network. Retransmission may retransmit the lost data packets. However, the speech packs lost by retransmission cannot be restored because of the need to play voice traffic in real time. QoS processes should therefore be regarded to minimize voice traffic loss.
VoIP is a description of several vulnerabilities that arise from both VoIP and the facilities (network, working scheme, etc.), these vulnerability may be utilized in carrying out security breaches of several kinds which include safety assaults including accessibility, confidentiality and privacy attack.
In safety of the scheme and network, the vulnerability is a fault or deficiency that an intruder could exploit to perform a safety offense. Two kinds of vulnerability have been identified by VoIP(Kolhar et al., 2018). One is the hereditary vulnerability from the infrastructure used to deploy VoIP apps (network, working system, web server, etc.). The other vulnerability is the result of VoIP protocols and systems, including IP phones, voices, media servers, signals, etc.
Hackers may use the VoIP vulnerabilities to perform various safety assaults. An attacker can disrupt a media service by the process of flooding, collect information regarding privacy through seizing signs of calls or calling contents, call hijacking through servers or clients, and generate calls of fraud via identity spoofing
For the deployment of secured IP data networks efficient measures have been proposed. The primary ones are firewalls, interpretation of the network address and IPSec protocol traffic encryption. For safe VoIP networks, conventional IP network security dimensions can be used. However, it makes several VoIP issues more complicated and affects the QoS (Surasak&Huang, 2019). To help implement an integrated IP network that permits data and voice traffic to be transmitted while taking into account QoS and safety requirements, adjustments were made to key IP data network security standards to support safeguards in the new rapidly expanding world of VoIP.
Under the scheme C/S architecture layout, it can also be split into two components, namely the server and the customer side, with regard to particular growth and execution. The distributed architecture technique on the server hand is taken to deploy the linear extension without influencing the system implementation “in the future updating. The customer portion focuses mainly on the protocol pile, which complies with the RFC necessity and performs the job of speech communication in real time (Gongjian, 2016).
Fig. 1 Structure of implementation; Source:https://dl.acm.org/citation.cfm?id=3028861
Currently the industry includes Asterisk, Sipxec, FreeSWITCH, OpenSIPS, YATE etc. For the servers scheme which utilizes the SIP Protocol. The scheme utilizes a Linux-based Asterisk scheme that is fully free open source as a server design alternative following thorough evaluation and comparisons (Gongjian, 2016). We benefit from the abundance of Asterisk plug-in modules and embrace the architecture for growth of public subscriptions. In addition, C language implements the “SIP redirect, proxy service of SIP, registration of SIP services and value added services of SIP.
The registration server recognizes the terminal’s application for registering and enters the terminal’s SIP and address of IP. The method of REGISTER is utilized” for registering. UA gives registration details first to the server, then the registered server checks customer validity, and returns the user authentication details.
Fig. 2 Flow process of server of Registrer; Source: https://dl.acm.org/citation.cfm?id=3028861
The obligation of the proxy server lies in the task of sending the message of call towards the UAS, the call side, and for transferring message of response to UAC.
Fig. 3 Flow chart for module of proxy server; Source: https://dl.acm.org/citation.cfm?id=3028861
On the terminal equipment of the user, SIP UA (User Agent) is operated (Gongjian, 2016).
“Division is based upon the servers called User account control (UAC) and User Agent Server (UAS). UAC” has the responsibility to call the servers or requested sites; UAS receives the caller’s signal or the status feedback and returns the data to the customer.
Fig. 4 Process of flow for Module of SIP user agent; Source: https://dl.acm.org/citation.cfm?id=3028861
The speech handling unit is primarily used for voice information capture and playback, encoding and decoding features.
Fig. 5 Process of flow for module of voice processing; Source: https://dl.acm.org/citation.cfm?id=3028861
The speech transmission module aims mainly at transmitting voiced information, allowing both parties to acknowledge speech information through the same protocol. Voice information are obtained in the same manner as the method of communication.
Fig. 6 Process of flow of module of voice transmission; Source: https://dl.acm.org/citation.cfm?id=3028861
This document was a short survey of the protocols used to promote VoIP technology, the potential threats to VoIP communications and the safety steps adopted to protect against these risks. VoIP system safety should enforce tangible embedded network safety. It should be safeguarded from harmful network attacks and any threats to the current network. The network and the servers concerned should accommodate the usage of the VoIP scheme.
Loss of packets, delay and performance lead to decreased speech performance. Furthermore, network congestion may happen in any part of the network at any moment. In order to facilitate integration with the Internet, IMS builds on SIP. The future of VoIP therefore seems bright because a device requires IPv6 and SIP only.The IMS network can also be connected via gateways to conventional telephony services, H.323 and other VoIP technologies. Despite the benefits of VoIP, QoS guarantees for speech communication through IP network are one of the remaining issues.
Information management System (IMS): It is a database and transaction management system introduced in 1968 by IBM.
Public Switched Telephone Network (PSTN):PSTN is a global collection of interconnected voice-oriented public telephone networks. PSTN means the public-connected telephone network or traditional circuit-connected telephone network.
Session Initiation Protocol (SIP): The Session Initiation Protocol is a signalling protocol used to initiate, maintain, modify and terminate real-time sessions involving video, voice, message, other applications and services in communications between two or more IP network endpoints.
Virtual Private Network (VPN):A virtual private network provides a secure, encrypted connection over a less secure network, for example public Internet.
over Internet Protocol (VoIP):Voice over Internet
Protocol (VoIP) is a technology that enables voice calls to be carried out
using a broadband Internet connection rather than regular (or analog) telephone
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