CHAPTER 1: INTRODUCTION 1.

1 Voice over Internet Protocol (VoIP) Voice over Internet Protocol (Voice over IP, VoIP) is a family of technologies, methodologies, communication protocols, and transmission techniques for the delivery of voice communications and multimedia sessions over Internet Protocol (IP) networks, such as the Internet. Other terms frequently encountered and often used synonymously with VoIP are IP telephony, Internet telephony, voice over broadband (VoBB), broadband telephony, and broadband phone. VoIP systems employ session control protocols to control the set-up and tear-down of calls as well as audio codecs which encode speech allowing transmission over an IP network as digital audio via an audio stream. The codec used is varied between different implementations of VoIP (and often a range of codecs are used). Some implementations rely on narrowband and compressed speech, while others support high fidelity stereo codecs. 1.2 Softphone ( eyeBeam ) For our project, we had chooseeyeBeam softphone. Eye Beam is a free VoIP phone that available for various type of operating system such as Windows, iPhone or any Android based smart phone such as Google Nexus, Sony Xperia, Motorola Droid or Samsung Galaxy. Connect EyeBeam softphone to a VoIP provider or to a VoIP PBX to make calls to any VoIP, mobile or landline number.EyeBeam also known as a full-fledged SIP-based softphone VoIP app that allows user to have a sophisticated voice and video communication experience whether the user are an individual or part of a business. It is more of a multimedia communicator than a simple VoIP app. Being a third-party SIP-based app, EyeBeam is not attached to any VoIP service, meaning that you need to have a service to tie to it. This also means that you are free to use any VoIP service that supports the SIP protocol. It is a great tool for advanced users and VoIP administrators. EyeBeam is one product in the line of SIP-basedsoftphone VoIP apps proposed by CounterPath, the other products being entry-level X-Lite and more sophisticated Bria. EyeBeam has more features and is richer than X-Lite in many ways, since X-Lite is free and EyeBeam has a price. X-Lite is intended to entice users to upgrade to paid versions of the CounterPath products.
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EyeBeam is different from Bria in the sense that Bria is more contact-centered and therefore better poised for business and corporate environments and collaboration, while EyeBeam offers more of what is required and expected of a properly said full-fledged VoIP softphone app. Standard Telephone Features The eyebeam softphone has all standard telephone features,including: Six lines Speakerphone Mute Redial Hold Do not disturb Call ignore Call history Call forward or called transfer Call record Six-party audio conferencing Four-party video conferencing Four video codec that going to use in this project H.263, H.264, H.263 + 1998,and H.263 Four audio codec that going to use in this projectbroadvoice32,speex, G729 and G.711

Enhance features and function The eyebeam softphone also supports the following voip features and function: Instant messanging and presence using the SIMPLE protocol. Managed contact list Log in with up to ten different Voip service providers. Zero-touch configuration of audio and video device Acoustic echo cancellation, automatic gain control voice activity detection

Automatic selection of the best codec based on the other partys capability, the available bandwidth, and network condition. eyebeam switches the codec within a call in response to change network condition

1.3 Callcentric For our project, we had used Callcentric phone system for our SIP server. Callcentric Phone System is a SIP server that works with popular VOIP Gateways and SIP phones that allow connecting with each other easily. Callcentric SIP server is a web-based, easily and friendly use that can runs on all popular versions of Operating system. Configuration is performed via an easy to use Web interface. The Session Initiation Protocol (SIP) is a protocol for establishing real time communication sessions with one or more participants. Its most frequently used for Voice communications but it can handle video as well, as well as future applications. SIP was designed to be independent of the transport layer, i.e it can work on UDP, TCP or STCP. All voice/video communications take place via another protocol, usually RTP. There are many RFCs surrounding SIP, but the most important one is RFC 3261. SIP is a text based protocol that looks and acts very much like the HTTP protocol. The original designers (Henning Schulzrinne& Mark Handley) wanted to make a protocol that had its roots in the IP world, rather than in the telecoms world. Sip has been an amazing success, being the major driver in the adoption of VOIP and Computer Telephony in recent years. All major manufacturers have adopted the standard and availability of SIP software, SIP hardware and Sip service providers is widespread. Sip servers are responsible for setting up the calls between Sip devices. SIP servers usually combine several of the SIP server functions such as SIP proxy and SIP register into one piece of software. 1.4 Network Monitoring System (PRTG) To monitor the network, we had used PRTGsoftware as our network monitoring system. PRTG is a web-based, 24x7 real-time QoS monitoring tool for networks. The software itself canmonitors network availability and network usage using SNMP, Packet Sniffing, WMI, IP SLAs and Netflow and various other protocols. This tool can help to minimize downtime for
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critical systems by providing administrators with early warnings of outages, rather than having to wait for user complaints or trouble tickets.PRTG also help to diagnose VoIP quality issues or failures quickly and take immediate corrective measures before the end user feels the impact. It provides a constant watch on the VoIP "QoS" metrics and traffic patterns over time enables optimal capacity planning thus preventing any chances of VoIP call quality degradation.The PRTG over VoIP monitoring software gives you the possibility to act quickly and proactively, maintaining a high quality of service and avoiding the economic costs of downtimes. The Advantages of PRTG in Monitoring Quality-of-Service (QoS) No add-ons, plug-ins or specific infrastructure is necessary. Everything you need is included in PRTG. No special hardware necessary with PRTG. Other monitoring products that provide QoS monitoring only offer IP-SLA, which requires expensive Cisco hardware.

PRTG can monitor IP-SLA, plus it measures the quality of a connection itself. In fact, our QoS sensors basically perform the same measurements as the IP-SLA feature does.

Using PRTG, you can monitor QoS even in low profile networks that run in part or fully on cheap, unmanaged switches, and across VPNs and WAN connections that the user may not have control over. When measuring QoS, with PRTG you are not limited to measuring connections between expensive Cisco boxes (which are mostly located "close to the core" anyway), so you may not even see the connection problems which are more likely on network edges, where often cheaper hardware is used.

With PRTG, you can also measure the QoS in the "corners" of the network or remote locations. All you need is a PC running Windows, install a probe and then you can measure the data line quality.

Even the Freeware Edition includes both the IP-SLA and QoS sensor!
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PRTG is likely the cheapest commercial enterprise grade software available for IP-SLA and QoS monitoring.

CHAPTER 2 : LITERATURE REVIEW 2.1 Introduction To VOIP Voice over IP (Voice over Internet Protocol or "VoIP") technology converts voice calls from analog to digital to be sent over digital data networks. This allows the university to use its existing data network (which includes fiber and copper wiring between and within buildings, and Cisco routers and switches) to also transport telephone service throughout the campus. By offering Voice over IP (or VoIP), UofL is moving toward a "converged" network, where voice, data, and video all travel along the UofL gigabit network. 2.2 Benefits of VoIP The VoIP system offers UofL many benefits as a replacement for its current telephone technology:

VoIP uses a single communications network for both telephones and computers instead of the separate phone and computer networks that are prevalent today

VoIP uses programmable sets that provide new features, applications and capabilities such as allowing the university to quickly relay alerts or messages to all locations.

VoIP makes it easy to administer the system and individual features can be configured through a simple web interface.

The IP phones can access the university phone directory, allowing users to find the most up-to-date telephone numbers right on their phones.

Using VoIP positions the University Future Technologies and future needs of students, faculty, and staff.

Users will be able to store a Personal Address Book and Fast Dial list, either using a Web interface or through manually entering information on the IP phone set. The speakerphones can be used in emergencies as a Public Address system. 2.3 Disadvantage of Monitoring Equipment One way to determine the effects of individual network performance factors (e.g., latency) on overall judged quality is to install monitoring equipment in a network and simultaneously collect data from users on judged quality of network performance. This approach has the advantage of face validity, i.e., applying in an obvious way to real-world conditions. The network monitoring approach does have disadvantages such as:

The first is expense and technical difficulty. If monitoring equipment for, say, latency, packet loss, and bandwidth were not already in place, then installing it could be expensive. If monitoring equipment were already in place, there is still the technical difficulty of getting users of the network to make judgments about its performance at specific times that could later be correlated with actual network performance.

Second is correlated variables, i.e., the fact that, for many systems, performance with respect to one metric or variable is correlated with performance with respect to other metrics simultaneously. An example is that latency, packet loss and packet jitter might all be simultaneously good or simultaneously bad, depending on the level of traffic. While this might reflect the true state of the world, it makes it difficult to isolate the effects of specific network performance factors.

The third one is repeatability. Suppose that, over some period of monitoring, the biggest effect on judged network performance quality was attributable to latency. If the study were re-done, there would be no guarantee that the network conditions observed in
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the first study would recur in the second study. The relationship between latency and judged performance quality might then also be different.

2.4 Quality of Service (QoS) Parameter Ranges. In an experiment, the range of values each parameter (or experimental variable) assumes can have an effect on the relative importance of the variable to the overall outcome of the experiment. In general, variables that take on a restricted range of values, compared to other variables, have a smaller effect on the experimental outcome. Bandwidth was varied using a control on the videoconferencing units. (Bandwidth could have been manipulated using the emulator as well.) The values used in both experiments were 128 kbits/s, 384 kbits/s, 768 kbits/s. Because the network subnet used in the labwas isolated from other parts of the local Ethernet system and free from competing traffic, the network could deliver any of the bandwidths specified. Latency. A baseline latency of more than 100 ms one way was estimated for the videoconferencing system due to signal-processing, irrespective of network latency.1 In addition to this baseline latency, the following one-way latencies were added by the emulator; these latencies are relatively long by telecommunications standards [ITU, 1996a; TIA, 2005]: 0 ms, 150 ms, 300 ms. Packet Loss. The range of packet loss for besteffort managed networks is agreed to be 02% [TIA, 2005]. However, this range is for random non-bursty packet loss (in which the loss probability is constant over time). However, bursty packet loss apparently is more characteristic of actual network performance [Clark, 2003]. Jitter is defined as a statistical variance of the RTP data packet inter-arrival time. In the Real Time Protocol, jitter is measured in timestamp units. The first step to dealing with jitter successfully is to know how large it is. However, we do not need to compute the precise value. In RTP, the receiving endpoint computes an estimate using a simplified formula (a first-order estimator). The jitter estimate is sent to the other party using RTCP (the Real Time Control Protocol).
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CHAPTER 3: METHODOLOGY AND TESTING Methodology used in this VOIP mini project is Iterative Methodology. This methodology is the process life cycle. The main reason of choosing the Iterative Methodology is because of it allows turning into the step before if any error occurred. In this project the methodology consists of:

Research

Design Implementati on Testing Evalute

3.1 Iterative Methodology: Research Research about the softphone, SIP server and the NMS to use in this mini project. Design

 SIP server, softphone and NMS (network monitoring system) software: o SIP Server Call Centric. o Softphone Eye Beam softphone. o NMS PRTG. Conducted on voice and video conferencing QoS involving SIP server and taking into account factors like delay, jitter, loss, MOS and R Factor. Testing and monitoring will be done in WAN environment.

3.2 Implementation

The first step is to register at www.callcentric.com to activate the SIP server. This

registration is free. Call Centric will act as SIP server to the softphone.

 After the registration complete, the page as above will appear. Call Centric will give the caller id to the user. It is the phone number.

At the eye beam softphone, configure as above figure. PRTG also install at the server to monitor the network as the connection established

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 PRTG is used to capture delay, jitter, loss, MOS and R Factor.

3.3 Testing

The testing is done to accomplish the objective to monitor the difference of Voice and Video Conference on SIP Server in VOIP implementation. Factors that are considered are jitters, loss, MOS, delay and R factor. Eye Beam Phone System as SIP Server has to be run first. Then, PRTG as NMS software is run to monitor the network. Calls are made between four users as video and voice based on different codec. Several call being made and the data have been captured.

4.4 Evaluate

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 After the data have been captured, there will be analysis phase to determine which codec are better using while voice and video conference call. CHAPTER 4: RESULT AND ANALYSIS In this part, the result of the QOS(Quality of Service) for each codec will be show. As mention before, author using four combination of codec which are broadvoice32 audio codec and H.263 video codec, speex audio codec and H.264 high end video codec, G729 audio codec and H.263 + 1998 video codec and G.711 audio codec and H.263 low end video codec. Comparison will be made in this part where author will observe the entire graph and differentiate the entire table. The result is tested within 20 minutes. 4.1 Result 1: Broadvoice32 Audio Codec and H.263 Video Codec.

Figure 1: PRTG sensor graph using broadvoice32 audio codec and H.263 video codec.

Jitter Minimum: 0ms Maximum: 4ms Average: 2ms

Delay Minimum: -18ms Maximum: 33ms Average: 0ms

Packet Lost 0%

MOS 4.4

Table 1: QOS result using broadvoice32 audio codec and H.263 video codec.

Figure 1 and table shows the result of broadvoice32 audio codec and H.263 video codec. By observing the graph, author fined that this combination of codec is not so stable where the graph
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is drop between 1.20pm to 1.25pm.Result from table 1 shows that, even though the graph is not stable but there no packet loss and the minimum delay is -18ms and the maximum delay is 33ms. Other than that, the MOS is 4.4. This finding shows that, although this codec is not so stable but there is no packet loss but just little bit of delay. 4.2 Result 2: Speex Audio Codec and H.264 High End Video Codec.

Figure 2: PRTG sensor graph using speexaudio codec and H.264 high end video codec.

Jitter Minimum: 0ms Maximum: 5ms Average: 4ms

Delay Minimum: -20ms Maximum: 37ms Average: 0ms

Packet Lost 0%

MOS 4.4

Table 2: QOS result using speex audio codec and H.264 high end video codec.

Figure 2 and table shows the result of speex audio codec and H.264 high end video codec. By observing the graph, author fined that this combination of codec is more stable than result 1. This is because the graph is not drop below than 0.4% compared to result where the graph is drop between 1.20pm to 1.25pm. Result from table 2 shows that, even though the graph stable and there no packet loss but it has more delay than result 1 where the minimum delay is -20ms and the maximum delay is 37ms. The MOS is 4.4 same like result 1. This finding shows that although this codec is stable but there is more delay than result 1, but still there is no lost packet.

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4.3 Result 3: G729 Audio Codec and H.263 + 1998 Video Codec.

Figure 3: PRTG sensor graph using G729 audio codec and H.263 + 1998 video codec.

Jitter Minimum: 0ms Maximum: 5ms Average: 3ms

Delay Minimum: -20ms Maximum: 42ms Average: 0ms

Packet Lost 0%

MOS 4.4

Table 3: QOS result using G729 audio codec and H.263 + 1998 video codec.

Figure 3 and table shows the result of G729 audio codec and H.263 + 1998 video codec. By observing the graph, author fined that this combination of codec is more stable than result 1 and 2. This is because the graph is rising. Result from table 3 shows that, even though the graph stable and there no packet loss but its maximum delay is more delay than result 2 and 1 where the maximum delay is 42ms, beside the minimum delay is more than result 1 which is -20ms. The MOS is 4.4 same like result 1 and 2. This finding shows that although this codec graph is rising and looks stable but there is more delay than result 1 and 2, but still there is no lost packet.

4.5 Result 4: G.711 Audio Codec and H.263 Low End Video Codec.
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Figure 4: PRTG sensor graph using G.711 audio codec and H.263 low end video codec.

Jitter Minimum: 0ms Maximum: 5ms Average: 4ms

Delay Minimum: -24ms Maximum: 40ms Average: 0ms

Packet Lost 0%

MOS 4.4

Table 4: QOS result using G.711 audio codec and H.263 low end video codec.

Figure 4 and table shows the result of G.711 audio codec and H.263 low end video codec. By observing the graph, author fined that this combination of codec is not stable than result 1, 2 and 3. This is because the graph dropping and at 2.55am to 3.00am the graph is drop to 0.2% which is not good. Result from table 4 shows that, the graph is not stable but still there no packet loss but its maximum is more than result 1 and 2 and the minimum delay is more than result 1, 2 and 3 which is far from good. Although the graph and the result are worst than result 1, 2 and 3 but still the MOS is 4.4 same like result 1, 2 and 3. This finding shows that codec is not good but still there is no packet lost and the MOS is 4.4.

CHAPTER 5: CONCLUSION

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From the finding, author can conclude that all of the codec has its own ability. Some of the codec may not stable but the delay is less than the other but some of the codec is stable but the delay is more. Other than that, there is a codec where it is not stable and the delay is worst from all the codec that has been tested. Although the codec have different result for graph, jitter and delay but still the there is no packet loss found and the MOS remain the same which is 4.4. Result 1, 2 and 3 is tested at peak hour where the connection is so good but result 4 is tested not in peak hour. This finding shows that even though result 1, 2 and 3 is tested at peak hour but it has better result than result 4 which author can conclude that codec for result 4 is not suitable for peak hour or not. But for result 4, the result may have been influenced from other such as the device itself, internet connection, operating system and more. More research needs to be done. Although there is different of graph, jitter and delay for each codec but there is no packet lost and the MOS remain the same which author can conclude that all the codec can be used but the delay is not same. From author point of view and experienced from the testing, codec for result 2 which is G729 audio codec and H.263 + 1998 video codec is better than all other 3 codec where the connection stable and just a little bit of delay.

References for Literature Review

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1. TelTel, http://www.teltel.com/ 2. Dahmouni, H., Vaton, S., Rosse, D.: A markovian signature-based approach to ip traffic classification. In: Proceedings of the 3rd annual ACM workshop on Mining network data, San Diego, California, USA, pp. 2934 (2007). 3. Clark, A. (2003). Packet loss distributions and packet loss models.ITU-T Contribution Com 12D97-E. 4. ITU-T. (1996a). One-way transmission time. Recommendation G.114. 5. TIA Telecommunications Industry Association (2005). Network model for evaluating multimedia transmission performance over Internet Protocol. TIA Standard Draft TIA921.

Project Scope Chapter 1: Introduction Mohammad Azri B. Mohd Negara 52208110245 Chapter 2: Literature Review Mohamad Syafiq Fikri B. Nordin 52208209356 Chapter 3: Methodology and Testing Mohd Zulfadli B. Izazu 52208110095 Chapter 4: Result And Analysis Khairil Qazrin B. Mohd Latif 52208110154 Chapter 5: Conclusion Khairil Qazrin B. Mohd Latif 52208110154

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