Abstract devices for communication services over public internet rather

    Abstract The objective of thecoursework is to implement the Voice over Internet Protocol (VoIP) call using SIP Server (Trueconf) and analysis the resultbased on the quality of service, security and using of different codec (G.

711and G.722) of VoIP call over wired (LAN) and wireless (Wi-Fi) network. Voice over Internet Protocol (VoIP) is a methodology and group oftechnologies for the delivery of voicecommunications and multimedia sessions over Internet Protocol (IP) networks, such as the Internet.

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VoIP allowto make calls free or very low cost. Voice over Internet Protocol telephonyprovides many features and services which are expensive using traditionaltelephony networks.  They are terms likeInternet telephony, broadband telephony and broadband phone service refers tothe devices for communication services over public internet rather than thepublic switched telephone network (PSTN). VoIP having manyadvantages over public switched telephone network (PSTN). Therefore, the reportis about a brief introduction on the basic principles and technologies of SIPbased VoIP networks, their advantages and disadvantages, and analysis of thewired/wireless VoIP calls by using TrueConf SIP Server. Principle of Operation of VoIP SystemsVoice over Internet Protocol (VoIP)is a method for the delivery of voicecommunications and multimedia sessions over Internet Protocol (IP) networks, such as the Internet.

 The basic principle of operation of VoIPtelephone calls are very similar to traditional telephony. The steps involvedare signalling,channel setup, digitization of the analog voice signals, and encoding. Instead of being transmitted over a circuit-switchednetwork, the digital information is packetized,and transmitted as IP packets over a packet-switched network. The media streams are transported using specialmedia delivery protocols that encode audio and video with audiocodecs, and video codecs (The termcodec is the short form for “encoder/decoder”).The media streams are encoded at the transmitter end, sent over the internetand decoded at the receiver end. The codecs must match at each end of thecommunication link.

Various codecs exist that optimize the media stream basedon application requirements and network bandwidth; some implementations relyon narrowband and compressedspeech, while others support high-fidelity stereo codecs. Some popular codecs include µ-law and a-law versions of G.711, G.722, an open source voice codec known as iLBC,and many others (“Voice over IP”, 4th December 2017, para. 1 &2).  Challengesin Implementation of VoIP Systems”Voice over Internet Protocol (VoIP) has become a popularalternative to traditional public-switched telephone network (PSTN) networksthat provides advantages of low cost and flexible advanced digital features.The flexibility of the VoIP system and the convergence of voice and datanetworks brings with it additional security risks” (Butcher, Li & Guo,2007). Therefore, it is necessary to analyse the challenges involved inimplementing the VoIP systems.

Some of the major drawbacks of VoIP systems areas follows:a)    Qualityof Service (QoS) – Quality of service (QoS) is fundamental to a VoIP network’soperation. A VoIP application is much more sensitive to delays than itstraditional data counterparts. Tools such as encryption and firewall protectioncan help secure the network, but they also introduce significant delay. Jitter,referred to nonuniform delays is another QoS issue, it can cause packets toarrive and processed out of sequence. The Real-Time Transport Protocol (RTP),which is used to transport voice media, is based on UDP, so packets receivedout of order can’t be reassembled at the transport level, but must be reorderedat the application level, introducing significant overhead.

To control jitter,network designers can use buffers and implement QoS-supporting network elements(especially routers) that let VoIP packets “play through” when larger datapackets are scheduled ahead of them. Another issue that must be dealt with isthe packet loss. However, VoIP packets are small, containing a payload of only10–50 bytes, or approximately 12.

5–62.5 ms, with most implementations at the shorterend of the range. The loss of such a minuscule amount of speech isindiscernible, or at least unworthy of complaint, by a human VoIP user (Walsh& Kuhn, 2005).b)   Bandwidth/concurrentcall capacity – Voiceover Internet Protocol (VoIP) bandwidth consumption over a network is oneof    the most important factors toconsider when deploying a VoIP infrastructure. Failure to account for VoIPbandwidth requirements will severely limit the reliability of the VoIP systemand place a huge burden on the network infrastructure and VoIP Server inquestion (Soroyewun & Obiniyi, 2015).A link which has high bandwidth may be able to carry large volumes of datapackets, but the network links with low bandwidth can cause packets loss andother QoS problems. Hence, proper bandwidth reservation and allocation isessential to VoIP quality. Sharing data and voice on the same wires, which isone of the great attractions of VoIP, is also a potential challenge forimplementation.

c)    Choiceof CODEC – In VoIP systems, codecs serve as animportant function – they enable the conversion of analog voice signals intodigital packets that can be sent over the Internet. VoIP codecs generallyaccomplish encoding/decoding and compression/decompression. The codec’s job isto convert the human voice into digitized packets that can be routed over theinternet just like any other form of media. These voice packets can be flaggedwith a higher priority through QoS settings to enable them to pass throughfirewalls or another border equipment quicker. This is often necessary toensure that audio does not get dropped during calls. The main differencebetween the various types of VoIP codecs used by providers comes from the waythey compress audio.

Unfortunately, there is no perfect codec that isacceptable for all enterprises under all circumstances. Some high-qualitycodecs require licensing fees whereas others are open source, some offer betterquality while others use less bandwidth. All these aspects will affect the finalchoice of codec selection by a provider, client or organization.d)   Security– Eavesdropping is the major security threat in VoIP systemsespecially when transmitting the data packets in wireless communication. Through eavesdropping, athird party can obtain names, password and phone numbers, allowing them to gaincontrol over voicemail, calling plan, call forwarding and billing information.This subsequently leads to service theft.

VoIP systems consists of softphonesand software that are vulnerable to viruses and malware attacks. Anothersecurity threat is call tampering, for example, the attacker can simply spoilthe quality of the call by injecting noise packets in the communication stream. TrueConf SIP ServerTrueconf is a developer of software andhardware solutions for video conferencing over the internet and in corporatenetworks.

Trueconf provides unique products like video conferencing server.Trueconf server provides cloud service for trueconf online and trueconf mobile,client applications for mobiles devices running on android and iOS. It helps toorganize people and video conferences in anywhere like in the office, at homeare in public places.Trueconf software architecture is basedon Scalable Video Coding (SVC)  Setup and Configuration of TrueConf SIP ServerTrueConf Server is shipped as a softwareinstallation package that contains the server side and client applications forpopular platforms. After the installation package is downloaded, it is launchedto begin the installation. TrueConf Web Manager port is determined during theserver installation. Since we were installing the server behind the firewall,to complete the registration TCP port 4310 was used to access from inside tointernet and port 80 was used between server and client applications.

Tocomplete the registration, the free registration key was obtained. After the server had been successfully registered, ontop of the activation window of web-configurator a special sign appeared on theserver status stating that the server is running and registered. Once theserver has been configured, the TrueConf client software is installed on theclient machine. Then internal and external addresses must be specified. Internal addresses and ports are used for clients tocontact this server. By default, the server uses all IP addresses of machine ondefault TrueConf Server port 4307. When default settings are on, currentconnections are displayed in this column. Externaladdresses are the ports and IP addresses or DNS names, which helpclient applications to connect to the server (“TrueConf Server AdministratorGuide”, n.

d.). TrueConf Server has built-in gateway for SIP protocolsinteroperability. Practical Implementation ofVoIP CallsTheTrueConf Server application has a dedicated softphone for making video calls. Thevideo call between two end-users can be established by just entering the userID or the IP address of the other user in the address bar as shown in figure 1.And figure 2 shows the TrueConf softphone with video call in progress.

Fig.1Fig.2The five main functions of the TrueConfSIP server in implementation of the VoIP calls are first the TrueConf SIPserver determines the user location and the type of end system that will beused by the session. Second function of the SIP server is to determine the useravailability. End users can tell the system that they are available to talk orthat they are busy and do not wish to be disturbed. The third SIP function isthe user capabilities function. This is important because different deviceshave different capabilities. For example, computer can do more things than aphone, therefore, the user capabilities function allows the SIP to determinethe media being used and of the parameters being associated with it.

The fourthSIP function is the session setup. This function is responsible for connectingthe call. It establishes session parameters for both the caller and therecipient of the call. The fifth and final function of the TrueConf SIP serveris the session management.

It allows the users to end a call, transfer a callto someone else, add multiple users to make a conference call or makemodifications to the session parameters (“An introduction to SIP and SIP functions”, n.d.). Results and DrawbacksThe VoIP video calls using TrueConf SIPserver were successfully made through wired connection as well as through thewireless connection via Wi-Fi. The sample packets were captured and analysedwith the observer. Problems with VoIP audio quality are always due to networkdelay, jitter and packet loss. Observer helps in tracking network factors thataffect quality and reports call quality scores, which measure the overall VoIPnetwork health.

Figure 3 shows the traffic summary of the call made throughwired connection using codec G.711.Fig.3Figure4 and figure 5 below, shows the statistics of the overall call quality of thecall made through wired connection using codec G.711.

The total number of VoIPpackets were 32,955. And the sample packetsshow how signalling, call set up/take down, data transfer occur during atypical session, duration of the call and the average jitter rate.  Fig.4 Fig.5Figure 6 shows the traffic summary ofthe call made through wireless connection using codec G.711.

 Fig.6Figure 7 and figure 8 below, shows thestatistics of the overall call quality of the call made through wiredconnection using codec G.711.

Thetotal number of VoIP packets were 34,345.Fig.7Fig.8VoIP enjoys adistinct advantage and supremacy of budget scalability in comparison with thecurrently operational alternative telephony systems. This allows lines to beshared with other users and services thus helping to lower the overall costsover the circuit switched networks. However, being predominantly a networkbased on software – it is exposed to the possibility of being attacked orharmed by the progressively rising threat of cyber-attacks from crackers interms of malware such as viruses and worms. The VoIP traffic, must pass acrossseveral different types of networks — often heterogeneous in nature.

Degradation of Quality of Service (QoS) is thus experienced while the traffictraverses across such assorted networks. The QoS parameter of VoIP trafficvaries, and can be quantified by a range of divergent metrics, such as the:jitter, end-to-end delay and Mean Opinion Score (MOS). The Mean Opinion Score(MOS) has been used to subjectively measure the voice quality in a telephonenetwork (Miraz,Molvi, Ganie, Ali & Hussein, 2017).For the calls made over Wi-Fi networks, astime passes, with the increased level of VoIP traffic due to the higher numberof calls generated, the MOS decreases. Whereas, there are better MOS scores forthe calls made over wired networks. ConclusionVoice overInternet Protocol is the digitized voice traffic intrinsically transmittedusing a data network to make telephone calls. This differs from using atraditional analogue circuit switched public network, as now the data has beensplit into packets.

These packets can take any route to reach the destination.Packetized data travel through a virtual circuit which differs from a circuitswitched network in that the circuit does not need to be reserved for theentire duration of the call between the sender and the receiver with packetswitching. The advantages offered include the multiple routing of the VoIP trafficensuring a cheaper and often free of cost flow of traffic between the differentintra-packet network components such as the routers and switches. On the otherhand, there are major drawbacks as well with regard to the quality of serviceand security, especially for the calls over Wi-Fi networks. The other issue tobe dealt with is the choice of codec. In this project, two types of codecs wereused – G.711 and G.722.

  Codec G.711 offers narrow-band voice with low processingrequirements, and with typical bit-rates of 32-64 kbps. G.711is currently used in a wide domain of applications,it performs best inlocal networks, therefore best suited for wired networks.

Codec G.722 offers wideband voice with low DSP processingrequirement. Itprovides improved speech quality due to a wider speech bandwidth of50–7000 Hz compared to narrowband speech coders like G.711. However, it requires abit-rate of 48-64 kbps per channel.

This codec is best suited for calls overwireless networks.In this coursework, we implemented the VoIP callsusing TrueConf SIP server. The video calls were established both on wired andwireless based networks. The overview of the principle of operation of VoIP,advantages and major drawbacks were presented in this report.

Next the overviewof TrueConf SIP server, its setup and configuration process, and its functionwas presented. Then the analysis of the packets captured during laboratoryinvestigations were presented to show how signalling, call setup/take down anddata transfer occur during a typical session. Finally,problems of making secure video calls atacceptable QoS levels over current Wi-Fi networks based on the lab resultsusing different codecs (G.711 and G.722) were discussed.   References       1       Voice over IP.

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org/wiki/TrueConf       6       TrueConf ServerAdministrator Guide. (n.d.). Retrieved December 19, 2017, from https://trueconf.

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