Telecommunication Transmission Channels, Switching Techniques, PSTN And VoIP

This assignment has two parts. Part A consists of a mixture of short and essay-type questions and Part-B requires you to write a short business report based on a casestudy. You are required to complete both parts.

Question 1

Critically discuss the type of impairments that exist on a telecommunication transmission channel and its impact on data rate and bit rate capacity of the channel?

Question 2

Critically compare the four different switching techniques in telecommunication networks, namely circuit-switching, message-switching, packet-switching and ATM switching. Use illustrations where appropriate to substantiate your answers.

Question 3

Describe a Public Switched Telephone Network (PSTN) and Voice over Internet technologies. Critically review the strengths and weaknesses of each of them. Use appropriate illustrations to support your discussion.

Consider an imaginary Internet Service Provider located in your country that caters to customers such as a call center, IT corporations, hospitals, financial institutions etc. The ISP is consulting with a telecommunication consultancy to optimize the use of the existing infrastructure for maximizing theirreturns as well as to plan the deployment of newer infrastructure and technologies to cater to the growing demands of their customers. You are expected to produce a technical report addressing the following questions with regards to the given case study.

There is an emerging demand from the customers to have converged voice/video/data networks and to securely access the internet services offered by the ISP. You are expected to offer a suitable solution to meet this demand. Implementing Voice and Video Over IP offer a great deal of advantages over traditional telephony services. But they still have their own set of limitations and challenges. Critically discuss the communication architecture and protocol (Session Initiation protocol and others) that enables delivery of voice and video over IP network. Particularly, comment on the limitations of such protocol and the impact it has on voice and video quality. Use appropriate illustrations and references to support your discussion.

It is understood that security concerns arise as a consequence of offering voice, video and data over IP network. Critically review the various security issues that may arise in this scenario and describe the protocols and mechanisms that may help mitigate the above concerns.

Impairments in Telecommunication Transmission Channels

The signal received will differ from the other signals that have been transmitted due to the impairments. The types of the impairments are:

1. Attenuation

This includes the signal amplitude that has been decreasing with the transmission medium. It is mainly for the attenuation process where the amplifiers or the repeaters are set at the particular interval for adequately improving and working on the original level approach. The amplifications, as well as attenuation, is for the decibel (dB) which has been for the decibels per unit. (Chen et al., 2014).

1. Delay Distortion

There have been a changing in the component of frequency as well as the signal that tends to arrive at the end of the receiver side, where there are delays that have been resulting in the delay distortion process as well. (Kuo et al., 2017).

With the change in the bit rate, there is a frequency that is also associated with it, which gets delayed and then it tends to interfere as well as gets associated with the later bit. It is mainly resulting in the inter-symbol interference, which is mainly due to the maximized bit rate.

1. Noise

The effect is mainly on the distortion of the transmitted signal as well as the attenuation of the signal that is transmitted. The signal to noise ratio is primarily to quantify the noise where:

Here, S is the average power of the signal

N is the Noise Power.

This includes the quantifying range to check on the noise with the signal. (Temprana et al, 2015). The higher SNR ratio is set with the relative signal to the noise that is mainly for a better quality as well.

The impulse noise has been primarily for checking the data where the energy is for 0.01 seconds that would not be destroyed till there is any voice data setup. This includes the bits for the digital data as well.

Effect of Bit Error Rate: This is the single bit which has been corrupted with the particular time interval. The BER is set to the value 10-5 where the over voice has been set with the graded line and is common. (Lau et al., 2015). The factors that affect the error rate is the bandwidth, S/N, transmission medium, distance, environment and the performance mainly set for the transmitter and the receiver.

Considering the

S is the signal power in Watts

R is the data rate

W is the bandwidth

Comparison of Circuit-switching, Message-switching, Packet-switching and ATM Switching

N is the noise power

Nyquist formula:  is set where the maximum data rate is for the finite set of the bandwidth with the noiseless channels. This includes the information which has been set for the channel capacity.

Maximum data rate with high SNR quality:

It includes the method for the two networking nodes that have been for the communication channels. (Pandya et al., 2014). These are set through the contrasts with the packet transfer delays. The bit delay shall be fixed with the connections where the circuit can easily be degraded with the competing users who are protected from the use of the other callers till the circuit is also released. The virtual circuit switching is the technology that can emulate the connections with the establishment before any packets are transferred.

In this, the message is considered to be the entity which contains the information, and for each, the switch is also decided. (Bangor et al., 2014). As per the network conditions, a proper conversation is made, where the messages are stored with the RAM limits, and they are known as the "Store and forward network. Any delay in the email is allowed till there is any real time transfer of the data in the computer system. It includes the data channels that could easily be shared with the communication devices. (Kou et al., 2017). With this, the messages can also be stored in the temporary form when there is a network congestion and forward it depending upon the priority. (Bangor et al., 2014).

The network has been set with the different methods which include the increased network efficiency as well as enabling the operations on the same network. The packets are for the header and the payload where the information in the header is mainly for the networking hardware which can also be used for the destination with the application software. The routing methods are for telecommunication messages where the research is based on handling the outing and the transfer of the data. (Porter et al., 2013).

It is mainly for the functionality that includes the circuit and the packet switching networks. The ATM can easily handle the asynchronous time division multiplexing where the encoding of the data is in the smaller or the fixed sized packets. The protocol and the Ethernets could be used with the variable sized frames, through the use of connection orientation model that has been for actual exchange in the data points with the dedicated connections as well as configuring the same through the use of signals. (Liu et al., 2014).

Description of PSTN and VoIP

This includes the aggregated of the circuit switched networks which could easily be operated through the public communication. The PSTM works on the telephone lines, cables as well as the transmission links, cellular networks. (Zhang et al., 2014). The network architecture is set to evolve the support with the increased subscribers, calls and the connections, where the concept of telephone exchange is based on hierarchies. The modern networks are set with the cost of the transmission, with lower equipment. (Tanaka et al., 2014). The automation process works on the digital switching where the trunks are mainly to connect the exchanges with the circuits or the channels. Hence,  the two-wire circuits are also used for the connections where the pulse code modulation is necessary for the G.7111.The call can easily be carried over with the 64 kbit/s channel where the digital signals are holding the circuit switching in the telephone exchange. The higher capacity of the communication links is aggregated with the time division multiplexing modules as well. (Arenburg et al., 2016).

It is based on the homes and the business that can be made for the local as well as the longer distance phone calls. The callers can easily speak and could be heard as there is a bidirectional form of setup. (Zhang et al., 2014). The subscribers can also handle the receiving of the services mainly from the local telecommunication company as well. It uses the circuit switching which also allows the users for the making of all the landline telephone calls easily and efficiently.

It is of higher cost than VoIP where there are different trade-offs which are set with the inherent qualities. (Zhang et al., 2016). The telecommunication is necessary with the purpose based on the designing as well as handling the support for the different traffic types. The VoIP technology has been having a significant impact on the system as it can support the larger volume of the traffic. Echo is also an issue. (Farash et al., 2014).

The setup of the VoIP is mainly based on the delivery of the voice communication as well as handling the sessions on the IP address. (Khasnabish et al., 2015). There are special references for the communication which includes the public Internet, as well as the public, switched telephone. (Morsy et al., 2014). It involves the signal process and the setup of the channels through digitization of the analog signals. The VoIP has been necessary for the business models that can handle the different forms of the mix and the matches. The solution will allow for the dynamic interconnection that is set in between the users on the different domains. (Porter, 2014). This holds the computer system with the legacy networks with the benefit of free calls and convenience.

Strengths and Weaknesses of PSTN and VoIP

This can handle the Internet access devices with the calls and the SMS text messages. It is mainly for the direct inbound dialing where the dedicated versions are set over the IP networks through the use of the technology like the Ethernet and the Wi-Fi. (Punreddy et al., 2014).

It includes the telecommunication where the PSTN and the other mobile networks where the foreign VoIP encounter a higher barrier to the registration. There is another issue which is about the calls between the Americans and the foreigners. (Predieri et al., 2013).

The SIP is the communication with signaling and controlling the communication sessions, with the internet telephone. The services and the systems are based on handling and controlling the video calls and instant messaging. (Ramanathan et al., 2015). This is important for the proper setup as well as the termination process of the voice over the VoIP. The essential focus has been on handling the speech communication as well as the implementation of the OSI layer reference model. The protocol is based on messages that are sent mainly between the ending points along with the establishment, termination as well as handling the call to create, modify and terminate the sessions. The HTTP and SMTP protocol are also used for the mail transfer. The conjunction is mainly with the specification of the data with the message specifying the codec and working on the Real-Time Transport Protocol. (Zhang et al., 2014). The motivation is for the SIP where the signaling procedures, as well as the setup of the call, is based on supporting the superset as well as handling the different features of the call configuration and signal.

The processing in this for the text messages has been a high load on the gateways. Here, the router needs to translate and work on the language so that it is easy for the router to understand as well. (Curpen et al., 2016). The SIP is new and tough to implement in the network. Along with this, it lacks the features for the endpoints, vulnerable to the eavesdropping, and registration. The audio injection with the potential decrease in the call quality where the bandwidth is also saturated.

The VoIP is the other protocol which works for the multimedia sessions as well as handling the communication services over the public internet. (Perello et al., 2013). This includes the signaling, setup of the channel as well as handling the digitization of the analog voice signals. The circuit switched networks where the transmission occurs as the IP packets are sent over the packet switched network. The audio streams also use the media delivery protocols which can be used for the encoding of the audio as well as the video processes to optimize the media stream.

Case Study Technical Report

Hence, ISP is important for catering to the customers where the telecommunication consultancy is based on optimizing the infrastructure with maximized return for planning a better deployment. The Video over the IP has been best as PSTN only includes the interconnection of the telephone networks under the commercial and the government owned form. (Liu et al., 2014). The aggregation is based on the circuit switching that has been evolving with the central local telephone office to the user. PSTN works on the long distance infrastructure, and ISP tends to share the circuit mainly among the users through the use of packet switching. Here, the internet users can easily avoid the payment for the usage of heavy tolls than any other ISP.

Without the connection to the internet, it does not support the system. Along with this, the quality of the sound and the video can also be very poor which can be due to the higher traffic with the people who are trying to use the same bandwidth. Along with this, there is a major process of compressing or decompressing the data over the IP network, which leads to the loss or the delay.

The Real-Time Transport Protocol is also able to provide the transport like for the video as well as the audio streams. All the services include the reconstruction of the time, with the detection of the loss, security as well as the other identification of the content. This has been designed for the multicast users which can also be used for the unicast forms. The RTP is designed over with the conjunction that is to get the feedback mainly based on the quality of the data as well as the transmission of information. (Irshad et al., 2015). The features are set for the end-to-end delivery of the data services that is set for the real-time characteristics. It includes the support that comes from the lowering of the layers as well as having control over the larger resources for the switches and the routers. The usual transmission does not offer any reliability or the control of the flow or congestion.

This is only working on the wireless VoIP clients, where the downstream of the call quality information where the forwarding mode has been the tunnel mode. The current feature is mainly based on decrypting the tunnel mode on the remote APs. It is only for the clients with codec: G711, G729, and G723.(Rahangdale et al., 2014).

VoIP has been working on the data network where the security weakness and the types of the attacks are based on the availability of the network. The data is exposed to the different points of attack which could be for the intruders. The risks have been with the IP, like the computer virus, Denial of Service and man in the middle attack which are dangerous to the system. The system also includes the service vulnerability as well as worms, virus and the other problems. The voice communication controls are the session control protocols or the IP addresses which are enclosed in the packet. VoIP has been set to spamming over the internet telephone where the marketing of the voice messages is mainly on targeting the IP phones. The attackers have a hijack identity and stealing of the money through exploiting the VoIP technology. (Arshad et al., 2016). The victim also receives the email or contracted through the call to a customer service number, where there is a prompt for the voice menus, in the attempt to steal the number of the account of the people, along with the PIN as well as other information.

1. The encryption using the IPSEC, TLS and S/MIME.

This is mainly to handle the confidentiality of the signals that is transmitted with the application layer protocol which could easily be used for the lowering of the protocol stack, particularly the network as well as the transport layers. (Farash et al., 2014). The encryption and the decryption are set for the CPU intensity to take time for processing with the impact on performance. The encryption has been used for the physical access as well as to the VoIP servers and gateways that allow the attacker to perform the traffic analysis.

1. The authentication of the user and the device

The processing of the servers is mainly through the phone like bootstraps for the temporary configuration which is important for the setup of the network. The standards are based on the threats where the authentication is also based on using the MAC with the IP phone which could be the best solution to the problem. The authentication also includes the automatic registration for the calling of the processed server that can easily be disabled. The authentication of the user with the password and the username and the PIN number is important for the avoidance of any issues. (Layman et al., 2015).

1. Controlling all the interactions with the segments of data

The IP-based telephony is mainly for the calls as well as to maintain the quality of service, scalability as well as security. This is important for managing the different logical networks over the segmentation of the IP voice that comes from the traditional network of the IP address. The data segments can easily be set with the virtualised LAN, to control the access as well as handling the stateful firewalls.

1. The inspection of the packet

Through this, the Network Intrusion Prevention System can easily parse and handle the analysis for the VoIP protocols. This is set for handling the attacks that target the services which can easily be detected or prevented.

1. Anti-virus software as well as security patch

The computer using this function works for the VoIP connection as well as the gateways which could be used for the access to the firewall, with the anti-virus as well as the other malicious code software. This is set to handle the latest virus signature or the malicious codes that could be set over the voice segments. The security patches are mainly to handle the gateways which are easy for keeping it up-to-date. (Lau et al., 2015). The prevalence is based on the virus threats for the clients where the VoIP is dedicated to the IP hardware that is set for the software based system. It also enables the disallowing of the data packets with the bypass of the firewall. The computers are found to be vulnerable to the virus or the worms.

1. Best practice for the VoIP phone

The users have to be aware of the plan with the contingency to make all the voice calls, even if the system fails. (De Rango et al., 2014). The vishing attacks could be avoided when the users do not release any sensitive information like the data of the credit card, bank account or the login credentials. This is mainly for the users which are set over the software based IP phones where the computer also need to be protected with the personalized firewalls as well as the anti-virus malicious code repair software that is set with the signatures as well as malicious code.

References

Chen, Y., Wang, B., Liu, D., Li, J.J., Xue, Y., Sakata, K., Zhu, L.Q., Heldt, S.A., Xu, H. and Liao, F.F., 2014. Hsp90 chaperone inhibitor 17-AAG attenuates Aβ-induced synaptic toxicity and memory impairment. Journal of Neuroscience, 34(7), pp.2464-2470.  Available through: <https://www.ncbi.nlm.nih.gov/pubmed/24523537> [Accessed 8 April 2017] 

Kou, X., Li, J., Liu, X., Chang, J., Zhao, Q., Jia, S., Fan, J. and Chen, N., 2017. Swimming attenuates D-galactose-induced brain aging via suppressing miR-34a-mediated autophagy impairment and abnormal mitochondrial dynamics. Journal of Applied Physiology, pp.jap-00018. Available through: < https://www.ncbi.nlm.nih.gov/pubmed/28302704> [Accessed 8 April 2017]

Temprana, E., Myslivets, E., Kuo, B.P., Liu, L., Ataie, V., Alic, N. and Radic, S., 2015. Overcoming Kerr-induced capacity limit in optical fiber transmission. Science, 348(6242), pp.1445-1448. Available through: < https://science.sciencemag.org/content/348/6242/1445 > [Accessed 8 April 2017]

Pandya, R.J., Chandra, V. and Chadha, D., 2014. Impairment aware routing and wavelength assignment algorithms for optical WDM networks and experimental validation of impairment aware automatic light-path switching. Optical Switching and Networking, 11, pp.16-28. Available through: < https://www.sciencedirect.com/science/article/pii/S1573427713000623 > [Accessed 8 April 2017]

Porter, G., Strong, R., Farrington, N., Forencich, A., Chen-Sun, P., Rosing, T., Fainman, Y., Papen, G. and Vahdat, A., 2013. Integrating microsecond circuit switching into the data center (Vol. 43, No. 4, pp. 447-458). ACM. Available through: < https://dl.acm.org/citation.cfm?id=2486007 > [Accessed 8 April 2017]

Liu, H., Lu, F., Forencich, A., Kapoor, R., Tewari, M., Voelker, G.M., Papen, G., Snoeren, A.C. and Porter, G., 2014, April. Circuit Switching Under the Radar with REACToR. In NSDI (Vol. 14, pp. 1-15). Available through: < https://www.usenix.org/node/179750 > [Accessed 8 April 2017]

Perelló, J., Spadaro, S., Ricciardi, S., Careglio, D., Peng, S., Nejabati, R., Zervas, G., Simeonidou, D., Predieri, A., Biancani, M. and Dorren, H.J., 2013. All-optical packet/circuit switching-based data center network for enhanced scalability, latency, and throughput. IEEE Network, 27(6), pp.14-22. Available through: < https://ieeexplore.ieee.org/document/6678922/ > [Accessed 8 April 2017]

Tanaka, T., Takahara, T., Nishihara, M., Yan, W., Li, L., Tao, Z. and Rasmussen, J.C., 2014, July. Optical Discrete Multi-tone for Software-defined Optical Packet Switching over Optical Circuit Switching Network. In Photonics in Switching (pp. PW3B-4). Optical Society of America. Available through: < https://www.osapublishing.org/abstract.cfm?uri=ps-2014-PW3B.4 > [Accessed 8 April 2017]

Zhang, L., Tang, S. and Cai, Z., 2014. Efficient and flexible password authenticated key agreement for voice over internet protocol session initiation protocol using smart card. International Journal of communication systems, 27(11), pp.2691-2702. Available through: < https://onlinelibrary.wiley.com/doi/10.1002/dac.2499/abstract> [Accessed 8 April 2017]

Farash, M.S. and Attari, M.A., 2014. An anonymous and untraceable password?based authentication scheme for session initiation protocol using smart cards. International Journal of Communication Systems. Available through: < https://link.springer.com/article/10.1007/s12083-015-0409-0 > [Accessed 8 April 2017]

Farash, M.S., 2016. Security analysis and enhancements of an improved authentication for session initiation protocol with provable security. Peer-to-Peer Networking and Applications, 9(1), pp.82-91. Available through: < https://link.springer.com/article/10.1007/s12083-014-0315-x > [Accessed 8 April 2017]

Irshad, A., Sher, M., Rehman, E., Ch, S.A., Hassan, M.U. and Ghani, A., 2015. A single round-trip sip authentication scheme for voice over internet protocol using smart card. Multimedia Tools and Applications, 74(11), pp.3967-3984. Available through: < https://www.ncbi.nlm.nih.gov/pubmed/28302704> [Accessed 8 April 2017]

Rahangdale, T.G., Tijare, P.A., Sawalkar, S.N., Balhare, Z.J. and Gulhane, V.S., 2014. An overview on security analysis of Session Initiation Protocol in VoIP network. IJRAT, 2(4), pp.190-5. Available through: < https://www.ncbi.nlm.nih.gov/pubmed/28302704> [Accessed 8 April 2017]

Arshad, H. and Nikooghadam, M., 2016. An efficient and secure authentication and key agreement scheme for session initiation protocol using ECC. Multimedia Tools and Applications, 75(1), pp.181-197.

Farash, M.S. and Attari, M.A., 2014. An efficient and provably secure three-party password-based authenticated key exchange protocol based on Chebyshev chaotic maps. Nonlinear Dynamics, 77(1-2), pp.399-411. Available through: < https://www.ncbi.nlm.nih.gov/pubmed/28302704> [Accessed 8 April 2017]

Layman, R., Smith, R.M. and Alexander, J., Vonage Business Solutions, Inc., 2015. METHOD FOR PROVIDING TIERED LOAD BALANCING FOR A HOSTED VOICE-OVER INTERNET PROTOCOL (VoIP) PRIVATE BRANCH EXCHANGE (PBX). U.S. Patent Application 14/752,086. Available through: < https://www.ncbi.nlm.nih.gov/pubmed/28302704> [Accessed 8 April 2017]

De Rango, F., Fazio, P., Scarcello, F. and Conte, F., 2014. A new distributed application and network layer protocol for voip in mobile ad hoc networks. IEEE Transactions on Mobile Computing, 13(10), pp.2185-2198. Available through: < https://www.ncbi.nlm.nih.gov/pubmed/28302704> [Accessed 8 April 2017]

Lau, P. and Vo, L.B., Verizon Patent and Licensing Inc., 2015. Session initiation protocol (SIP) signaling to keep a voice over internet protocol (VoIP) session active during a call hold. U.S. Patent 8,984,067. Available through: < https://www.cisco.com/c/en/us/td/docs/interfaces_modules/cable/line_cards/voice_adapter/cva122_122e/feature/guide/SIP_fm.html> [Accessed 8 April 2017]

Morin, D., Telecommunication Systems, Inc., 2016. Reverse 911 using multicast session internet protocol (SIP) conferencing of voice over internet protocol (VoIP) users. U.S. Patent 9,369,294. Available through: < https://www.cse.wustl.edu/~jain/cis788-99/ftp/voip_protocols/ > [Accessed 8 April 2017]

Arenburg, R.T., Barillaud, F., Dutta, S. and Mendoza, A.V., International Business Machines Corporation, 2016. Remedying distortions in speech audios received by participants in conference calls using voice over internet protocol (voip). U.S. Patent Application 15/057,789.

Zhang, L., Tang, S. and Cai, Z., 2014. Robust and efficient password authenticated key agreement with user anonymity for session initiation protocol-based communications. IET Communications, 8(1), pp.83-91. Available through: < https://link.springer.com/article/10.1007/s12083-014-0315-x > [Accessed 8 April 2017]

Bangor, A., Brandt, J.L. and Reynolds, D.F., At&T Intellectual Property I, LP, 2014. Methods and apparatus for alternative billing of voice over internet protocol (VoIP) services. U.S. Patent 8,750,826. Available through: < https://www.cisco.com/c/en/us/td/docs/interfaces_modules/cable/line_cards/voice_adapter/cva122_122e/feature/guide/SIP_fm.html> [Accessed 8 April 2017]

Khasnabish, B., Zte Corporation (China) and Zte (Usa) Inc., 2015. Method and system for implementing integrated voice over internet protocol in a cloud-based network. U.S. Patent Application 14/797,938. Available through: < https://www.cse.wustl.edu/~jain/cis788-99/ftp/voip_protocols/> [Accessed 8 April 2017]

Zhang, L., Tang, S. and Zhu, S., 2016. An energy efficient authenticated key agreement protocol for SIP-based green VoIP networks. Journal of Network and Computer Applications, 59, pp.126-133. Available through: < https://www.sciencedirect.com/science/article/pii/S1084804515001666 > [Accessed 8 April 2017]

Morsy, A., Kern, D. and Dennis, G., Heartv Llc, 2014. System for providing secondary content relating to a VoIp audio session. U.S. Patent 8,774,172. Available through: < https://www.google.si/patents/US8774172 > [Accessed 8 April 2017]

Ramanathan, S. and Schaffa, F.A., International Business Machines Corporation, 2015. Accessing really simple syndication (rss) enabled content using session initiation protocol (sip) signaling. U.S. Patent Application 14/588,485. Available through: < https://link.springer.com/article/10.1007/s12083-014-0315-x> [Accessed 8 April 2017]

Punreddy, A., Regan, M., Snider, M.A., Shea, R.W. and Silver, B., Aspect Software, Inc., 2014. Testing and monitoring voice over internet protocol (VoIP) service using instrumented test streams to determine the quality, capacity and utilization of the VoIP network. U.S. Patent 8,660,016. Available through: < https://www.cse.wustl.edu/~jain/cis788-99/ftp/voip_protocols/ > [Accessed 8 April 2017]

Curpen, R., Zamfir, S., Iliescu, I. and Balan, T., 2016. PERFORMANCE ANALYSIS OF VoIP SERVICES IN LTE NETWORKS. Bulletin of the Transilvania University of Brasov. Engineering Sciences. Series I, 9(2), p.29. Available through: < https://www.ncbi.nlm.nih.gov/pubmed/28302704> [Accessed 8 April 2017]