Proprietary Vs Open Source Software

Published Date: 02 Nov 2017

This section looks at modern day electronic learning (e-learning) and how these environments can be used to enhance the learning cycle of VoIP. This review looks at implementing a virtual environment, specifically the deployment methods deliverable content. Whilst performing this literature review it became apparent that there are considerable gaps in the research and development of virtual tutorial environments for VoIP. Research of literature written for virtual tutorials in other areas provided insight into the different research areas mentioned above.

Software Considerations

VoIP applications have certain requirements for deployment and implementation. Software architecture, memory access are example of these. Theses application need continual access to a computer with network access and a power source. With a virtualised environment an extra layer of software is required for the network to be built upon. Virtualisation software, operating systems, PBX server applications, Softphone clients and analysis tools need to be utilised to fulfil the requirement of this project. This leads to requirements for investigation of available implementations.

2.1.1 Operating Systems

There are a huge amount of operating systems (OS) in existence, all built to suit a wide variety of requirements. Investigation of these is out with the scope of this project, however there are some OS considerations when selecting PBX software. There are a host of VoIP server software which all come with unique capabilities and distributions. Integrics Enswitch, MediaCore Softswitch, Elastix, AskoziaPBX and GNU are examples of open source software which use Linux as their base distributions (Wikipedia, 2013). Other PBX Software can be run on Windows Environments such as; eyebeam, Cisco IP communicator, Avaua Server and AOL instant Messenger. Linux and Windows are the most popular operating systems bases for PBX server software. Investigation has proven that indeed it is noticeable that the vast majority of PBX server software is Linux based.

Because Linux is such flexible operating systems many PBX application have chosen it for distribution purposes. Linux has far better security that windows (K. Noyles, 2010) and is available cost free. User privileges play a major part in virus propagation, where 'root' user privileges can cause damage through viruses spreading easily. Linux operating systems boot faster than their Windows counterparts, Pupppy Linux can boot under 5 seconds (TechRepublic, 2011). Linux provides built in software packages (LINUX.COM, 2009) at zero cost whilst providing exceptional reliability. This is the reason so many PBX Server applications are deployed on Linux. Because of these benefits and the proven track record of Linux VoIP PBX distributions it is the obvious choice for this project.

2.1.2 PBX Software

PBX (Private Branch Exchange) is essentially a switch which creates connections between telephone devices within a private organisation. The advantages of a PBX is in that the organisation does not need as many telephone lines, reducing costs. A virtual PBX provides the same functionality by emulating the switch operations. There are numerous virtual PBX providers (Wikipedia, 2013), selecting one of these can be a challenge. Some free or open source projects lack enhance PBX services while some proprietary applications are not cost effective. Before selecting a PBX it is important to be aware of which services are obtainable and their functionality. This project is focusing on Virtual or "Hosted" PBX applications. These systems do not need to be located within a organisations premises, allowing distributed administration of phone connectivity. Asterisk PBX is widely considered the best virtual PBX solution (B. Channabasavaiah, 2011), (voip-manager.net, 2011), (P. Gralla, 2006).

2.1.2.1 Asterisk

A popular Linux based open source PBX server software package is Asterisk. Asterisk has become the de facto standard in modernVoIP PBX systems (voipmanager.net, 2013) Asterisk is an open source converged telephony platform. It is designed to run on Linux primarily. Applications such as voicemail, hosted conferencing call queuing and agents, hold music and call parking are all standard. (L. Madsen., 2011). As an open source application Asterisk provides versatility for personnel user as well as business and user by developers. Asterisk also runs on a variety of different operating systems (Wikipedia, 2012). The advantages of this are that there are no licence costs to take into consideration and its versatility makes it ideal for deploying onto various systems. With vast support and documentation available for Asterisk implementation and operation is simplified. With Asterisk installation is fast and simple, with a web browser front end provided. It supports standard PBX functions, sample configurations and crucially it supports multiple virtualsed and separated PBXs to be emulated on the same Asterisk machine (voipmanager.net, 2013)

Two popular platforms for implementing Asterisk are CentOS and Ubuntu Server. CentOS is a free operating system distribution based upon the Linux kernel. It is copied entirely from the Red Hat Enterprise Linux (RHEL) distribution Ubuntu Server is a computer operating system based on the Debian Linux distribution and distributed as cost free and open source software, (Wikipedia, 2012). Notable differences; on an Ubuntu system the root password is generated at random which is not known to the administrator. Another difference is the use of the ‘sudo command‘ to attain root-privileges. Repositories of Ubuntu contain newer packages than CentOS, it tends to be a little less conservative than CentOS, conversely CentOS packages focuses only on security upgrades. CentOS uses YUM package manager with RPM packages whereas Ubuntu uses apt and DEB packages. Ubuntu cleanly upgrades between the major versions, CentOS requires more reinstalls.

Asterisk supports a extensive range of video and Voice over IP protocols, including the Session Initiation Protocol (SIP) File Structure. Asterisk is a complicated system, which has many resources. These resources make use of the file system in many ways. Since Linux provides so much flexibility in this regard, it is useful to understand what data is being stored.

The Asterisk configuration files include extensions.conf, sip.conf, modules.conf, and numerous other files that identify parameters for the different channels, resources, modules, and functions that may well be in use. These files can be located in /etc/asterisk. Asterisk does not have an internal concept of trunks or stations. In Asterisk, everything that comes through or out of the system passes from end to end through a channel. There are numerous different types of channels; however, the Asterisk dial plan handles all channels in a comparable manner, which means that, for example, an internal user can be present on the end of an external trunk (e.g., a mobile phone) and be treated by the dial plan in exactly the same manner as that user would be if they were on an internal extension.

The conversation between two end devices when initiating a connection with use of the SIP protocol via an Asterisk gateway is explained below. The user agent in telephone 'A' does not know the IP address of 'B'. But it is aware the IP address of the SIP proxy (could be the address is 10.10.1.99). The user agent will create an INVITE request and send on to the proxy. The To: header of the request has the SIP URI <sip:[email protected]>. The body of the INVITE request has an SDP (Session Description Protocol) message providing the parameters (codec, IP address, ports) the called party is required to send its RTP stream to the caller. The SIP proxy without delay responds with "100 Trying" and then it forwards the INVITE request to the objective telephone 'B'. The proxy server adds one Via: header to the message. As mentioned before, the SIP proxy can access the position database and then knows the IP addresses of all the registered devices. Telephone 'B' starts calling and then sends the reply "180 Ringing" to the proxy server. The proxy will send the response to the telephone 'A'.

The called user answers the phone then their telephone sends the response "200 OK". The body of the response has an SDP message so that the caller is aware of where to send the RTP stream. The proxy server sends the response to the caller. The caller (telephone A) confirms the receiving of "200 OK" with the ACK message. The proxy server sends the ACK to the telephone B. Then, the call has been established and individual users start sending the RTP streams. When one of the users hangs up, their telephone sends the request BYE and also the SIP proxy sends the message to the other party. The other party replies to the BYE request with "200 OK" (also, the proxy server forwards the response to the other side). Both parties cease sending RTP data and the call is finished.

Asterisk is built on modules, a module is a loadable component that allows a specific functionality, such as channel drivers (for example, chan_sip.so), or a resource that allows the connection to an outside technology (such as func_odbc.so) Asterisk modules are booted based on the /etc/asterisk/modules.conf file. Dialplan applications are used in and by extensions.conf to define the variety of actions that can be applied to a VoIP call. The Dial() application, for example, is in charge of making departing connections to external resources and is questionably the most significant dialplan application. The sip.conf file contains the LocalSets context which is referring to the Locals sets in the Dialplan (extensisons.conf). The literal extensions can contain in their name: numbers (0-9), letters A,B,C,D (some hard phones have these letters) or all letters (a-z)

2.1.2.2 Communications Manager (CUCM)

Another very popular telephony system is Cisco Unified Communications Manager (CUCM). This is a software-based call-processing system developed by Cisco Systems with an IP communications processing system for up to 40,000 users (Cisco 2012). A major disadvantage to using CUCM is the proprietary nature of the application. Additional it can only be used with Cisco hardware and is not compatible with the Linux operating system. The scalability and resources available for CUCM ensure that this is a versatile and easily deployable call control platform (Wikipedia 2012). Whilst it is true that as a telephony system CUCM has some major advantages over its rivals (most notably the resources and training material) it is not as versatile as the Asterisk platform. (T. Keating, 2006) has highlighted a trend of migration from CUCM to Asterisk, citing cost as the main factor. Cisco licensing requires annual fees per phone and use the proprietary Skinny Call Control Protocol by default which means they have to be re configured to use them with a different protocol/PBX. The reasons that Asterisk has become so popular as an alternative to Cisco environments are the; rich functionalities, ease of integration, ease of migration and low costs (www.eyepea.eu, 2012). Additionally it is a very IT-orientated which can be deployed in a virtualised infrastructure.

2.1.2 Virtualisation Software

VMware is a virtual-machine platform that makes it achievable to run an unmodified operating system as a user level application. The OS running on the VMware can be rebooted, crashed, modified, and reinstalled all whilst not affecting the integrity of other applications running on the computer. It is important to look at VMWare workstation in more detail to optimise the configuration. VMware virtual machines become vastly portable between computers, because every host looks virtually identical to the guest. The following figure shows how three virtual machines will be used for deployment.

A virtual-machine monitor is another layer of software in between the hardware and the operating system that can virtualise all of the hardware resources of the machine (VMware, 2012). It essentially builds a virtual hardware execution environment called a "virtual machine" (VM). Numerous VMs can be used at the same time, and each VM provides segregation from the real hardware and other activities of the underlying system. Because, it provides the false impression of a standard PC (Personal Computer) hardware within a VM

matching virtual network. Virtualisation does supply an outstanding flexibility and portability, but can also bring in degradation in network performance, especially in elevated performance throughput and lower latency devices. A virtual appliance is defined as a minimum virtual machine image that just contains the software appliance built to run in a virtualised environment. But, it can also build custom appliances or packages for teaching, software experimentation and for performance and network analysis.

2.1.3 Proprietary vs. Open Source Software

When comparing proprietary vs. open source software the advantages and disadvantages have to be taken into consideration. Unlike proprietary software, open source software applications make their source code available for free, which can be customised to fit the unique needs of specific organisations (P. Nagy, 2007) . The arguments for open source are: lower cost, since development is done by a community of volunteers, customisation, you usually cannot make non-cosmetic changes to a proprietary system, nimble, open source projects adopt new trends faster than proprietary systems, openness, open source systems are usually designed with integration needs whereas commercial systems contain business motivations to lock organisations into a closed system, fast bug and security fixes. The arguments against open source are questionable quality, this is an invalid argument. All software, proprietary or open source, runs the scope from exceptional to poor. Open source projects presently tend to be more open than proprietary systems about discussing their bugs. No responsibility, the sticking point for open source is that it’s not a company, and there’s rarely any direct customer service.  Open source is a process and philosophy which produces software, but it is not a contract. Not aligned with corporate needs, open source software frequently starts small, and whilst they may be technically robust (scalable), they might not be designed with the requirements of corporate users. The proprietary system is to be expected to dominate the open source platform industry equally in terms of market share and profitability. This may clarify the dominance of Microsoft in the marketplace for PC operating systems (N. Economides, 2001).

Measuring Quality of Service

Common problems on VoIP networks can include; complete loss of service, interference, volume discrepancies, variable audio, lack of features and equipment malfunctions (HC. Liu, 2005)

There has been detailed analysis and research into network packet transfer considerations. Data loss, latency and consistency (jitter) are factors which can define a networks quality. In VoIP service is what communication facilities are available to consumers, QoS policies can be used to optimise the VoIP network depending on usage and deployment. (B. Cole, 2009) concludes that here are many quantitative factors that can affect the perceived quality of a VoIP call. B. Klepec, 2001) Defines three parameters as primary factors affecting voice quality within network offering VoIP technologies. These are: clarity, end-to-end delay and echo. These are all affected delay, jitter, packet loss and bandwidth. (V. Namboordiri, 2010) defines tolerable latency as 100-300ms, which can be measured using the ping command line tool. By pinging a host through a network which is used for VoIP applications the approximate round trip in milliseconds can be analysed. Jitter delay is one such consideration where packets arrive at their destination out of sequence. This can cause detrimental effects on the quality of a phone conversation. (Y. Amir , 2006) points towards the use of UDP within VoIP applications as a subject for quality degradation caused by network failure and packet loss. Jitter values captures the variation of data arrival time. To alleviate Jitter delay datagram's must be received and sent at the same rate, this is not always possible in networks. This is where a jitter buffer can be used between the VoIP application and the network layer. Jitter buffers come at a cost, by buffering datagram's alleviate the jitter there is extra delay in added. Mean Opinion Score (MoS) is a method of analysing QoS requirements., it measures average quality of speech patterns throught the network. Resource Reservation Protocol (RSVP) which is defined by IETFIP Differentiated Services allows applications to understand various network performance requirements (docwiki.cisco.com, 2013). RSVP is used in conjunction with routing protocols to build dynamic access lists for smart routing. Other mechanisms include Perpetual Speech Quality Measure (PSQM), Perceptual Analysis Measurement System (PAMS) and the E Model (ITU G.107) where packet loss is measures with over 10% loss being flagged.

2.2.1 Interference and Distortion

VoIP networks can suffer from static interference. This is generally the result of weather conditions. These include heavy rain, thunderstorms and wind. Which generates electricity on the phone lines. The static interference does not affect web browsing and general internet usage to a degree that is obvious . However this static interference does become more obvious when VoIP applications affecting voice quality. This can be alleviated by disconnecting the hardware form the main electrical supply to disburse the static build up. Distortion to voice signals has to be monitored and analysed to ensure QoS.

2.2.2 VoIP Bandwidth Dependency

VoIP requires a broadband connection simply because dial-up connections are not fast enough. Considerations need to be made based upon the user load, call volume and length of calls. During times of heavy use the service and quality of VoIP communication can be compromised. Some network devices attempt to overcome this problem by employing various queuing techniques (DiffServ, 802.1 p/q, IP ToS, etc.) (BlueCoat, 2009). These mechanisms are used to prioritise packet delivery where voice packets are given precedence over other traffic. Usually bandwidth is measured in MegaHertz (MHz) and bit rate is calculated by the bits per second (kbps). With VoIP communication bandwidth has higher requirements than other types of traffic. Where a connection may be able to support VoIP traffic there may be issues with multiple instances running on the same network. This proves the impetrative of providing sufficient bandwidth capabilities.

Network Segregation

An organisation could deploy VoIP using a network which shares data with other services. For example a private LAN could be used for web browsing, gaming and downloading as well as for VoIP. Because of this the network would need to have provisions in place which take into account the bespoke needs of VoIP traffic. A Virtual Local Area Network (VLAN) is a means for network segregation. Where as a LAN rely on physical switches or Hubs to propagate data. These devices are physically connected, this constricts deployment meaning that all devices have to be located in the same geographical area. VLANs also offer additional benefits over LANS's.

VLAN's offer a number of advantages over traditional LAN's. Performance is improved, by reducing traffic sent via broadcast and multicast messages. These messages increase latency and require more processing by routing devices. VLAN's use switches to make broadcast domains rather than using Routers. VLANs allow for simpler administration, making it easier to deal with user relocation in networks. VLAN's mean that routers do not need to be used which lowers the cost of implementation. There are security benefits also with VLANs by reducing the access to sensitive data from outsiders. VLANs allow for restriction of access, firewalls and central administration.

VoIP Codec's

VoIP sends voice data packets by first compressing the data. It is a similar principle to archiving a file to save memory, processing and bandwidth. Codec's are compressions software's which come in different incarnations. Codec's are each designed for specific purposes. It is important to select the correct codec for the communication use or quality of the voice call will be compromised. Numerous codecs are available to use with VoIP applications. It is important to consider the attributes of these to ensure that the correct decisions are made when implementing a VoIP network. Listed on the next are some of the most popular codec's which are used within a VoIP context.

Codec

Bandwidth & Kbps

Attributes

G.711

64

Provides accurate voice transmission. extremely low processor needs. requests at a minimum 128 kbps for shared communication

G.722

48/56/64

Adapts to altering compressions and bandwidth is preserved with network congestion.

G.723.1

5.3/6.3

High compression with high quality audio. Can use with dial-up. Lot of processor power.

G.726

16/24/32/40

An improved version of G.721 and G.723 (different from G.723.1)

G.729

8

Excellent bandwidth utilization. Error tolerant. License required.

GSM

13

High compression ratio. Free and available in many hardware and software platforms. Same encoding is used in GSM cellphones (improved versions are often used nowadays).

iLBC

15

Robust to packet loss. Free

Speex

2.15 / 44

Minimizes bandwidth usage by using variable bit rate.

Signalling Protocols

There are a number of protocols that may be employed in order to provide for VoIP communication services. In this section, we will focus on those which are most common to the majority of the devices deployed and being deployed today. Quite a few protocols are seen as industry leaders in terms of functionality and popularity. The protocols H.323 developed by the International Telecommunication Union (ITU), SIP which was developed as an alternative to H.323 and H.248 (MGCP) developed by Cisco as an alternative to H.323 are the three most popular in operation today (H. Ingo, 2007). Cisco Skinny Client Protocol (SCCP) was developed to communicate from H.323 Proxy and Cisco VoIP phone.

2.5.1 SIP Protocol

The SIP protocol is one of the most widely used communication sessions protocols for VoIP. Without the use of this protocol it would not be possible for end devices (soft phones) to communicate in the tutorial environment. The session setup, communication facilities, and header fields are just a few aspects of SIP that can all be studied to develop an understanding of how to best implement the tutorial environment.

Figure 1 shows a classic example of a SIP message communication between two users, "Alice" and "Bob". (Each message has a label with the letter "F" and a number for reference by this text.) In the example, Alice uses a SIP application on her Computer (referred to as a soft phone) to call Bob on his SIP soft phone over the Internet. This typical arrangement is usually referred to as the "SIP trapezoid" illustrated by the geometric shape of the dotted lines (rfc3261, 2002).

The example in Table 2 contains the full details of the session setup. The first line of the text-encoded message contains the method name (INVITE). The lines that follow are a list of header fields. This example contains a minimum required set.

The other header fields are briefly described below:

Via has the address (pc33.atlanta.com) at which Alice is expecting to getresponses to this request. To contains the display name (Bob) and a SIP or SIPS URI (sip:[email protected])

From also has a display name (Alice) and a SIP or SIPS URI (sip:[email protected]) that show the originator of the request.

Call-ID has a globally unique identifier for the call, made by the combination of a random string and the soft phones host name or IP address.

CSeq or Command Sequence contains an integer and a method name. The CSeq number is incremented for each new request.

Contact has a SIP or SIPS URI that represents a direct path to contact Alice, usually composed of the username at a fully qualified domain name (FQDN).

Content-Type has a description of the message body

Content-Length hass an octet (byte) count of the message body.

When a user agent client desires to initiate a session (for example, audio, video, or a game), it formulates an INVITE request (Wikipedia, 2012). The INVITE request asks a server to establish a session.

2.5.2 H.323 Protocol

H.323 is a protocol for communicating multimedia content. The protocol has been developed to incorporate real-time video and audio transfer over network similar to IP functionality. The majority of VoIP applications use the H.323 protocol to allow call setup and termination, transfer and forwarding. The different functions of H.323 utilise UDP and TCP and provides a standard which is comparable to the more up to date SIP protocol. QoS is an important part of the H.323 protocol, providing best effort packet delivery and prioritisation of traffic.

2.5.3 H.248 Protocol

H.248 or Media Gateway Control Protocol (MGCP) is a session management and signalling protocol required for multimedia communication. The protocol converts data from a format which circuit-switched networks require to that which a packet-switched network requires. Megaco is the name for an enhanced and developed version of H.248. The H.248 protocol is supported by the Internet Engineering Task Force (IETF) and standardised by the Telecommunications Union.

Network Analysis

Detailed analysis of a VoIP network allows for investigation into network traffic flow. Abnormal patterns can be detected which could be a sign of compromised security, availability or QoS problems. As mentioned some of the factors which can affect VoIP networks are; Jitter, Latency and QoS. The various VoIP protocols such as: UNISTEM, SCCP, SIP, H.323 and MGCP can be examined and evaluated using network analysis tools. The objective of VoIP network analysis is to understand and apply principles which affect how the network operates. It is important to recognise security risks and understand potential security threats. There are various applications which can be used for these can be used to understand traffic patterns, reconstruct VoIP conversations for quality assessment and to look at configuration and threats.

An excellent application which can be used for VoIP analysis is WireShark. Wireshark is a open source packet analyses product. Primarily its use it network troubleshooting, development and education. Wireshark can be configured to analyse protocols such as the SIP, H.248 and H.323. Capture options and filters can be used to capture packets from particular protocols. Wireshark has VoIP and Graph analysis functionality, information such as IP Address, direction of packets, message type, codes values, protocol and other useful information. Wireshark can also play RTP audio streams, this can reproduce the voice signals propagated during a VoIP call. Another useful feature is the jitter buffer which can be used to analyse QoS form both ends of a conversation.

The Environment

There are an estimated 1.4 billion computers in the world (Wiki.answers.com, 2012). In an age of environmental concern it is imperative that every step possible is taken to reduce our carbon footprint. The environmental impact of computer power consumption becomes clearer as researchers try to understand how much electricity computers consume as a collective (M. Talebi, 2008). Emulating hardware can be seen as a more sustainable method of computing, indirect energy use and green house gas emission attributable to activities that support ongoing PC use which advocates the use of virtual environments.

According to a report published by the Climate Group , a think-tank based in London England, computers, printers, mobile phones and the widgets that accompany them, account for the emission of 830m tones of carbon dioxide around the world in 2010 (B. Azerbajan, 2011). That is about 2.2% of the estimated total of emissions from human activity. According to the report, about a quarter of the emissions in question are generated by the manufacture of computers and so forth. The rest come from their use. It is obvious that the exponential demand for IT devices by the multifarious industries and the mushrooming growth of IT industry has led to various imbalances at the ecology and environmental level. A significant rise in electronic wastage only further highlights the need for responsible use of such technologies. A physical implementation of a VoIP infrastructure would at a bare minimum need the following devices; Voice Router, Analogue Telephone Adapter, IP Phone and a Computer Workstation Increasing the network size results in the need for additional hardware, which in turn leads to increased costs and emissions. Even for a small network used exclusively for training the CO2 emissions are far higher than that of an emulated environment.

Even in these fragile economic times, CIOs (Chief Information Officers) are investing in virtualisation. A 2008 survey from CIO Research indicates that 85% of CIOs surveyed have implemented virtualisation in the data centre, and 81% believe that their virtualisation efforts have resulted in significant savings (R. Freedman, 2009). Apart from the savings, and the corporate responsibility benefits from a Greener IT profile, CIOs cite other important benefits, which include simplified maintenance, improved disaster recovery procedures, and the capability to provision systems and new applications quicker.

While the green benefits of virtualisation are known, the survey also revealed some unexpected pitfalls amidst the virtualisation euphoria. 42% of CIOs surveyed said that politics and organisational challenges are as much a problem as technical points. Some CIOs also described that IT teams are still operating in separation, creating integration problems that can impede virtualisation success.

Amenities savings are also hugely significant, as smaller data centre footprints result in savings in real estate, power, and cooling expense. Lots of organisations have utilised virtualisation to stay away from building new data centres and to decrease the footprint of their existing data centres by up to 60%. Virtualised IT environments allow enhanced backup and data mending operations by facilitating automatic failover. These environments can additionally help increase development efforts by making it simpler to bring up development servers though the use of template driven provisioning.

Every silver lining has a cloud, virtual software is no exception. As previously mentioned, there are often many political and organisational problems related to virtualisation. For example, users frequently become accustomed to having their own dedicated servers that allows them to control and modify according to their own schedule and needs.

Teaching Perspective

A subject- matter expert or instructor often approaches the design of a course from a content perspective, that is, what to cover. In contrast, an instructional designer approaches the task by first defining the problems and then determining what knowledge and skills are needed to solve the instructional problem. Designing effective instruction by (D. Merrill, et al, 1996) states that the goal of instructional design is to make learning more efficient and effective and less difficult. This leads on to the question of what problems need to be solved in the VoIP tutorial. A brief description of these areas; Basic configuration, dial plans, voicemail, modules, protocols, security, and data handling. So for example a problem could be described as: User A needs to contact user B but user B is unavailable. How to create a voicemail service to leave user B a message.

The best tutorials are designed to organise information into small chunks that can be absorbed at a learner’s own pace and allow the user to interact by intermittent testing and feedback (G. Miller 1956). Rather than creating a large quiz or exam to be completed by the user at the end of the entire tutorial it may be more constructive to break this process into more manageable chunks. Not only would this act as a form of feedback to the user but it would also create logical goals in a more achievable manner.

Conclusions

It is apparent that a virtual tutorial environment dedicated to VoIP training and experimentation would be a valuable addition to the teaching syllabus of VoIP. Providing the user more control over their learning whilst alleviating limitations with traditional teaching techniques. Previous studies have highlighted the difficulty with implementing fully functioning learning environments. These studies have cited operational setbacks with hardware malfunctions being predominant. This can only be seen as an advantage of using emulated hardware as this alleviates the problem with hardware failure. However it highlights the need for extensive planning and testing before implementing the virtual environment.

Research concludes that e-learners need to be able to ask questions and share information and ideas. Ideally this tutorial would accommodate instructor interaction. Social support can be derived in the form of collaborative involvement with other users. The efficiency, effectiveness and success of this tutorial environment need to be measured. The effectiveness can only be derived by reviewing the user’s collective feedback. This feedback can be taken through an interactive quiz which will ask about the users experience and opinions on the tutorial as well as so questions to grade their understanding of the areas covered in the VoIP tutorial

Because open source software applications can be customised to fit the unique needs of specific organisations are usually designed with integration in mind it makes sense to choose this over a proprietary application. Commercial systems have business motivations to lock organisations into a closed system. Asterisk the open source converged telephony platform runs on a variety of different operating systems. There are no licence costs to take into consideration and its versatility makes it ideal. Asterisk will be the base telephony system for implementing the tutorial. The choice between CentOS and Ubuntu for installing the Asterisk system fairly arbitrary as the differences between the two have no real influence on implementation. Since these are both open source monetary costs are not applicable.

Environmental considerations of a VoIP virtual tutorial environment need to be assessed here; it's possible to measure the power consumption of the hardware needed for this environment. The power output averaged over a set time period then compared to the equivalent physical VoIP environment.

The goal of instructional design is to make learning more efficient and effective and less difficult. The best online tutorials are designed to organise information into small chunks that can be absorbed at a learner’s own pace and allow the user to interact with the program by intermittent testing and feedback Methodology. The tutorial needs to be engaging and not overly intensive. The next task is to write a tutorial which encompasses the points listed above.