Analysis of Quality Services in VoIP
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Published: Mon, 19 Feb 2018
Background to Research
Due to the Innovative changes in telephony devices and related technologies world wide, the time has come to analysis the quality in telephone devices and provide improved versions of communication channels. Locally the implementation of telephony services is getting increased; many new organizations are setting up their resources to make this system and its facilities available to the users. The research in the telephone industries is in progress since last many years shown a great improvement in all over the world. Previously this telephony service used PSTN  which uses 54 kbps channel now after the improvement and change in the technology this telephonic service shifted to internet protocol. As Internet is a widely used medium for data receiving and transfer. Now this new technology becomes Voice over IP.
The concept of VoIP (Voice over Internet Protocol)  originated in about 1994, when hobbyists began to recognize the potential of sending voice data packets over the Internet rather than communicating through standard telephone service. This allows PC users to avoid long distance charges, and it was in 1994 that the first Internet Phone Software appeared. While contemporary VoIP uses a standard telephone hooked up to an Internet connection. Previous efforts in the history of VoIP required both callers to have a computer equipped with the same software, as well as a sound card and microphone. These early applications of VoIP were marked by poor sound quality and connectivity, but it was a sign that VoIP technology was useful and promising. The evolution of VoIP occurred in next few years, gradually reaching the point where some small companies were able to offer PC to phone service in about 1998. Phone to phone service soon followed, although it was often necessary to use a computer to establish the connection. Like many Internet applications in the late 1990’s, early VoIP service relied on advertising sponsorship to subsidize costs, rather than by charging customers for calls. The gradual introduction of broadband Ethernet service allowed for greater call clarity and reduced latency, although calls were still often marred by static or difficulty making connections between the Internet and PSTN (public telephone networks). However, startup VoIP companies were able to offer free calling service to customers from special locations.
The breakthrough in VoIP history  came when hardware manufacturers such as Cisco Systems and Nortel started producing VoIP equipment that was capable of switching which means that functions that previously had been handled by a telephony service now implement in computer’s CPU and will work as “switching” a voice data packet into something that could be read by the PSTN (and vice versa) could now be done by another device, thus making VoIP hard ware less computer dependent. Once hardware started becoming more affordable, larger companies were able to implement VoIP on their internal IP networks, and long distance providers even began routing some of the calls on their networks over the Internet. Usage of VoIP has expanded from the year 2000, dramatically. Different technical standards for VoIP data packet transfer and switching and each is supported by at least one major manufacturer – no clear “winner” has yet emerged to adopt the role of a universal standard. Whereas companies often switch to VoIP to save on both long distance and infrastructure costs, VoIP service has also been extended to residential users. In the Span of few years, VoIP has gone from being a fringe development to a mainstream alternative to standard telephone service.
At present there are two standards that are in use for VoIP switching and gateways: SIP and H.323. SIP  mainly relates to end-user IP Telephony applications, while H.323 is a new ITU standard for routing between the circuit-switched and packet-switched worlds used for termination of an IP originated call on the PSTN, but the converse is also becoming common at a very fast rate. As the technology getting advanced and many improvements have been implemented in making sure to maintain the quality of voice and data over the internet should be maintained. The main purpose of this thesis is to discuss the techniques to maintain the quality of VoIP and the role of protocols in VoIP which are H.323 and SIP
Area of Research
The area of research focuses on Study and Analysis of Quality Services in VoIP and the discussion of Role of H.323 and SIP  Protocols. Many techniques and mathematical models have been developed and implemented. As a matter of fact this thesis is not intended to provide any new model or strategy for improving Quality services in VoIP but to get the picture based on the standard matrix of measurement of QoS of VoIP like MOS .
Analysis of Quality Services of VoIP
Due to the emerging and advancements in the telecommunication making All-IP integrated communicating infrastructure capable to support applications and services with diverse needs and requirements. During the last few years a lot of attention is given to delivering voice traffic over both the public internet and corporate Intranets. IP Telephony, or VoIP, does not only provide more advanced services (example personalized call forwarding, instant messaging etc) than PSTN, but it also aims to achieve the same level of QoS and reliability ,. As opposed to PSTN, VoIP utilizes one common network for signaling and voice transport and thus enjoys several advantages with respect to the telephony services that are through All-IP networks infrastructures. The most important factors that influence the adoption of VoIP include improved network utilization by using advanced voice CODECS that compress the voice samples below 54 Kbps, possibilities to offer value added services(i.e. instant message, personalized call forwarding etc.) just to mention a few. In VoIP world many Quality impairments  introduced today by the Internet, it is important to provide mechanism in order to measure the level of quality that is actually provided today in the internet to interactive multimedia applications. That is, to measure how extensive are the loss, the delay and delay jitter impairments and how bad their impact on the perceived QoS,  is. There are a large number of methods proposed and some of them standardized which monitor the distorted signal and provide a rating that correlates well with voice quality. The most important parameters that affect the VoIP Quality are the following:
- Network Packet Loss
Demonstration Methodology; Simulation
The OPNET Simulation is used during aforesaid research work  and is a very powerful network simulator. Main purposes are to optimize cost, performance and availability. The following tasks are considered:
- Build and analyze models.
- Configure the object palette with the needed models.
- Set up application and profile configurations.
- Model a LAN as a single node.
- Specify background service utilization that changes over a time on a link.
- Simulate multiple scenarios simultaneously.
- Apply filter to graphs of results and analyze the results.
Role and Analysis of H.323 & SIP Protocols
Based on the research works that has been done so far, this part of the thesis will discuss and elaborate the H.323 and SIP  protocols and a comparative analysis of these two protocols based on their specification will discuss in detail in the next chapters
Results and Conclusions
The final conclusion of the simulation results will be shown and a comparative analysis of different CODECS with their performances from the simulated results and Role of H.323 and SIP protocols will be discussed.
VoIP and Quality of Service
In past traditional technology, telephone calls are carried through Public Switched Telephone Networks (PSTN), which provides high-quality voice transmission between two or more parties. Whereas the type of data such as email, web browsing etc. are carried over packet-based data networks like IP, ATM and Frame Relay. In the last few years, there has been a rapid shift towards using data networks to carry both the telephone calls and the data together. This so called convergence of voice and data networks is very appealing due to many considerations. VoIP systems digitize and transmit analog voice signals as a stream of packets over a digital data network.
VoIP technology insures proper reconstruction of voice signals, compensating for echoes due to the end-to-end delay, for jitter and for dropped packets and for signaling required for making telephone calls. The IP network used to support IP telephony can be a standard LAN, a network of leased facilities or the Internet. VoIP calls can be made or received using standard analog, digital and IP phones. VoIP gateways serve as a bridge between the PSTN and the IP network . A call can be placed over the local PSTN network to the nearest gateway server, which moves it onto the Internet for transport to a gateway at the receiving end. With the use of VoIP gateways, computer-to-telephone calls, telephone-to-computer calls and telephone-to-telephone calls can be made with ease.
Access to a local VoIP gateway for originating calls can also be supported in a variety of ways. For example, a corporate PBX (Private Branch Exchange) can be configured so that all international direct dialed calls are transparently routed to the nearest gateway. High-cost calls are automatically supported by VoIP to obtain the lowest cost. To ensure interoperability between different VoIP manufacturers, VoIP equipment must follow agreed upon procedures for setting up and controlling the telephone calls. H.323 is one such family of standards that define various options for voice (and video) compression and call control for VoIP. Other calls setup and control protocols being utilized, and or being standardized include SIP, MGCP , and Megaco. IP Telephony goes beyond VoIP transport and defines several value added business and consumer applications for converged voice and data networks. Examples include Unified Messaging, Internet Call Center, Presence Management, Location Based Services etc.
During the last few years, the voice over data network services have gained increased popularity. Quick growth of the Internet Protocol (IP) based networks, especially the Internet, has directed a lot of interest towards Voice over IP (VoIP). The VoIP technology has been used in some cases, to replace traditional long-distance telephone technology, for reduced costs for the end-user. Naturally to make VoIP infrastructure and services commercially viable, the Quality of Service (QoS) needs to be at least close to the one provided by the Public Switched Telephone Network (PSTN). On the other side, VoIP associated technology will bring to the end user value added services that are currently not available in PSTN.
VoIP and QoS
In the networks of packet switching, the traffic engineering term is abbreviated as (QoS) or Quality of Service , , which refers to resource reservation control mechanisms instead of it, is to be understood as achieved service quality. Quality of Service (QoS). This Quality of services guarantees are important for the limited capacity network, for example in cellular data communication, especially for real-time streaming multimedia applications, for example voice over IP and IP-TV . Quality of Service may or may not be agreed by Network or protocols and software and reserve capacity in the network nodes, for example during a session establishment phase. But in the entire the achieved level of performance, for example the data rate and delay, and priorities in the network nodes. The reserved capacity might be released during a tear down phase. Quality of Service does not supported by the Best Effort network Service. The ITU standard X.902 as defined the QoS quality requirements on the collective behavior.
The Quality of Service on all the aspects of a connection, such as guaranteed time to provide service, voice quality , echo, loss, reliability and so on. Grade of Service term, with many alternative definitions, rather than referring to the ability to reserve resources.
The convergence of communications and computer networks has led to a rapid growth in real-time applications, such as Internet Telephony or Voice over IP (VoIP). However, IP networks are not designed to support real-time applications and factors such as network delay, jitter and packet loss lead to deterioration in the perceived voice quality. In this chapter, brief background information about VoIP networks which is relevant to the thesis is summarized. The VoIP network, protocol and system structure along with the brief over view of the QoS of VoIP  are described in this chapter. Voice coding technology and main Codecs also discussed in the thesis (i.e. G.729, G.723.1) are discussed. Network performance characteristics (e.g. packet loss and delay/delay variation) are also presented in next sections.
In past years when the Internet was first deployed, it lacked the ability to provide Quality of Service guarantees due to limits in router computing power. It is therefore run at default QoS level, or “best effort. The Technical Factors includes reliability, scalability, effectiveness, maintainability, Grade of Service, etc.
- Dropped packets
- Out-of-order delivery
Quality of Service (QoS)  can be provided by generously over-provisioning a network so that interior links are considerably faster than access links. This approach is relatively simple, and may be economically feasible for broadband networks with predictable and light traffic loads. The performance is reasonable for many applications, particularly those capable of tolerating high jitter, such as deeply-buffered video downloads.
Commercially involved VoIP services are often competitive with traditional telephone service in terms of call quality even though QoS mechanisms are usually not in use on the user’s connection to his ISP and the VoIP provider’s connection to a different ISP. In high load conditions, however, VoIP quality degrades to cell-phone quality or worse. The mathematics of packet traffic indicates that a network with QoS can handle four times as many calls with tight jitter requirements as one without QoS. The amount of over-provisioning in interior links required to replace QoS depends on the number of users and their traffic demands. As the Internet now services close to a billion users, there is little possibility that over-provisioning can eliminate the need for QoS when VoIP  becomes more commonplace. For narrowband networks more typical of enterprises and local governments, however, the costs of bandwidth can be substantial and over provisioning is hard to justify. In these situations, two distinctly different philosophies were developed to engineer preferential treatment for packets which require it.
Early work used the “IntServ” philosophy of reserving network resources. In this model, applications used the Resource reservation protocol (RSVP) to request and reserve resources through a network. While IntServ mechanisms do work, it was realized that in a broadband network typical of a larger service provider, Core routers would be required to accept, maintain, and tear down thousands or possibly tens of thousands of reservations. It was believed that this approach would not scale with the growth of the Internet, and in any event was antithetical to the notion of designing networks so that Core routers do little more than simply switch packets at the highest possible rates.
The second and currently accepted approach is “DiffServ” or differentiated services. In the DiffServ model, packets are marked according to the type of service they need. In response to these markings, routers and switches use various queuing strategies to tailor performance to requirements. (At the IP layer, differentiated services code point (DSCP) markings use the 5 bits in the IP packet header. At the MAC layer, VLAN IEEE 802.1Q and IEEE 802.1D can be used to carry essentially the same information). Routers supporting DiffServ use multiple queues for packets awaiting transmission from bandwidth constrained (e.g., wide area) interfaces. Router vendors provide different capabilities for configuring this behavior, to include the number of queues supported, the relative priorities of queues, and bandwidth reserved for each queue.
VoIP Networks Connections
Common VoIP network connections normally include the connection from phone to phone, phone to PC (IP Terminal or H.323/SIP Terminal ) or PC to PC, as shown in Figure 2.1. The Switched Communication Network (SCN) can be a wired or wireless network, such as PSTN, ISDN or GSM.
Perceived QoS or User-perceived QoS is defined as end-to-end or mouth to ear, as the Quality perceived by the end user. It depends on the quality of the gateway (G/W) or H.323/SIP terminal and IP network performance. The latter is normally referred to as Network QoS, as illustrated in Figure 2.1. As IP network is based on the “best effort” principle which means that the network makes no guarantees about packet loss rates, delays and jitter, the perceived voice quality will suffer from these impairments (e.g. loss, jitter and delay).
There are currently two approaches to enhance QoS for VoIP applications. The first approach relies on application-level QoS mechanisms as discussed previously to improve perceived QoS without making changes to the network infrastructure. For example, different compensation strategies for packet loss (e.g. Forward Error Correction (FEC)) and jitter have been proposed to improve speech quality even under poor network conditions. The second approach relies on the network-level QoS mechanism and the emphasis is on how to guarantee IP Network performance in order to achieve the required Network QoS. For example, IETF is working on two QoS frameworks, namely DiffServ (the Differentiated Services) and IntServ (the Integrated Services) to support QoS in the Internet. IntServ uses the per-flow approach to provide guarantees to individual streams and is classified as a flow-based resource reservation mechanism where packets are classified and scheduled according to their flow affiliation. DiffServ provides aggregate assurances for a group of applications and is classified as a packet-oriented classification mechanism for different QoS classes. Each packet is classified individually based on its priority.
VoIP Protocol Architecture
Voice over IP (VoIP) is the transmission of voice over network using the Internet Protocol. Here, we introduce briefly the VoIP protocol architecture, which is illustrated in Figure 2.2. The Protocols that provide basic transport (RTP ), call-setup signaling (H.323 , SIP ) and QoS feedback (RTCP ) are shown.
VoIP System Architecture
Figure 2.3 shows a basic VoIP system (signaling part is not included), which consists of three parts – the sender, the IP networks and the receiver . At the sender, the voice stream from the voice source is first digitized and compressed by the encoder. Then, several coded speech frames are packetized to form the payload part of a packet (e.g. RTP packet). The headers (e.g. IP/UDP/RTP) are added to the payload and form a packet which is sent to IP networks. The packet may suffer different network impairments (e.g. packet loss, delay and jitter) in IP networks. At the receiver, the packet headers are stripped off and speech frames are extracted from the payload by depacketizer. Play out buffer is used to compensate for network jitter at the cost of further delay (buffer delay) and loss (late arrival loss). The de-jittered speech frames are decoded to recover speech with lost frames concealed (e.g. using interpolation) from previous received speech frames.
Analysis of QoS Parameters
A Number of QoS  of parameters can be measured and monitored to determine whether a service level offered or received is being achieved. These parameters consist of the following
- Network availability
Network availability can have a significant effect on QoS. Simply put, if the network is unavailable, even during brief periods of time, the user or application may achieve unpredictable or undesirable performance (QoS) . Network availability is the summation of the availability of many items that are used to create a network. These include network device redundancy, e.g. redundant interfaces, processor cards or power supplies in routers and switches, resilient networking protocols, multiple physical connections, e.g. fiber or copper, backup power sources etc. Network operators can increase their networks availability by implementing varying degrees of each item.
Bandwidth is probably the second most significant parameters that affect QoS. Its allocation can be subdivided in two types
- Available bandwidth
- Guaranteed bandwidth
Many Networks operators oversubscribe the bandwidth on their network to maximize the return on investment of their network infrastructure or leased bandwidth. Oversubscribing bandwidth means the BW a user is subscribed to be no always available to them. This allows users to compete for available BW. They get more or less BW depending upon the amount of traffic form other users on the network at any given time. Available bandwidth is a technique commonly used over consumer ADSL networks, e.g., a customer signs up for a 384-kbps service that provides no QoS (BW) guarantee in the SLA. The SLA points out that the 384-kbps is typical but does not make any guarantees. Under lightly loaded conditions, the user may achieve 384-kbps but upon network loading, this BW will not be achieved consistently. This is most noticeable during certain times of the day when more users access the network.
Network operators offer a service that provides minimum BW and burst BW in the SLA. Because the BW is guaranteed the service is prices higher than the available BW service. The network operator must ensure that those who subscribe to this guaranteed BW service get preferential treatment (QoS BW guarantee)  over the available BW subscribers. In some cases, the network operator separates the subscribers by different physical or logical networks, e.g., VLANs, Virtual Circuits, etc. In some cases, the guaranteed BW service traffic may share the same network infrastructure with available BW service traffic. This is often the case at location where network connections are expensive or the bandwidth is leased from another service provider. When subscribers share the same network infrastructure, the network operators must prioritize the guaranteed the BW subscribers traffic over the available BW subscribers’ traffic so that in times of networks congestion the guaranteed BW subscribers SLAs are met. Burst BW can be specified in terms of amount and duration of excess BW (burst) above the guaranteed minimum. QoS mechanism may be activated to discard traffic that use consistently above the guaranteed minimum BW that the subscriber agreed to in the SLA.
Network delay is the transit time an application experiences from the ingress point to the egress point of the network. Delay can cause significant QoS issues with application such as SNA and fax transmission that simply time-out and final under excessive delay conditions. Some applications can compensate for small amounts of delay but once a certain amount is exceeded, the QoS becomes compromised.
For example some networking equipment can spoof an SNA session on a host by providing local acknowledgements when the network delay would cause the SNA session to time out. Similarly, VoIP gateways and phones provide some local buffering to compensate for network delay. Finally delay can be both fixed and variables. Examples of fixed delay are:
Application based delay, e.g., voice codec processing time and IP packet creation time by the TCP/IP software stack  .
Data transmission (queuing delay) over the physical network media at each network hop. Propagation delay across the network based on transmission distance
Examples of variable delays are:
- Ingress queuing delay for traffic entering a network node
- Contention with other traffic at each network node
- Egress queuing delay for traffic exiting a network node
Jitter is the measure of delay variation between consecutive packets for a given traffic flow. Jitter has a pronounced effect on real time delay sensitive applications such as voice and video. These real time applications expect to receive packets at a fairly constant rate with fixed delay between consecutive packets. As the arrival rates increases, the jitter impacts the applications performance  . A minimal amount of jitter may be acceptable, but as jitter increases the application may become unusable. Some applications, such as voice gateways and IP phones,  can compensate for small amounts of jitter. Since a voice application requires the audio to play out at constant rate, in the next packet time, the application will replay the previous voice packets until the next voice packet arrives. However if the next packet is delayed too long it is simply discarded when it arrives resulting in a small amount of distorted audio. All networks introduce some jitter because of variability in delay introduced by each network node as packets are queues. However as long as the jitter is bounded, QoS can be maintained.
Loss can occur due to errors introduced by the physical transmission medium. For example, most landline connections have very low loss as measured in the Bit Error Rate. However, wireless connections such as satellite, mobiles or fixed wireless networks have a high BER that varies due to environment or geographical conditions such as fog, rain, and RF interference, cell handoff during roaming and physical obstacles such as trees, building and mountain . Wireless technologies often transmit redundant information since packets will inherently get dropped some of the time due to the nature of the transmission medium.
Loss can also occur when congested network nodes drop packets. Some networking protocols such as TCP (Transmission Control Protocol) offer packets loss protection by retransmitting packets that may have been dropped or corrupted by the network. When a network becomes increasingly congested, more packets are dropped and hence more TCP transmission. If congestion continues the network performance will significantly decrease because much of the BW is being used to retransmit dropped packets. TCP will eventually reduce its transmission window size, resulting in smaller packets being transmitted; this eventually will reduce congestion, resulting in fewer packets being dropped. Because congestion has a direct impact on packet loss, congestion avoidance mechanism is often deployed. One such mechanism is called Random EARLY Discard (RED). RED algorithms randomly and intentionally drop packets once the traffic reaches one or more configured threshold. RED takes advantage of the TCP protocol’s window size throttle feature and provides more efficient congestion management for TCP-based flows. Note that RED only provides effective congestion control for application or protocols with TCP like throttling mechanism
Determine the order in which traffic is forwarded as it exits a network node. Traffic with higher emission priority is forwarded a head of traffic with a lower emission priority. Emission priorities also determine the amount of latency introduced to the traffic by the network node’s queuing mechanism. For example, delay-tolerant application such as email would be configured to have a lower emission priority than delay sensitive real time applications such as voice or video. These delay tolerant applications may be buffered while the delay sensitive applications are being transmitted.
In its simplest of forms, emission priorities use a simple transmit priority scheme whereby higher emission priority traffic is always forwarded ahead of lower emission priority traffic. This is typically accomplished using strict priority scheduling (queuing) the downside of this approach is that low emission priority queues may never get services (starved) it there is always higher emission priority traffic with no BW rate limiting.
A more elaborate scheme provides a weighted scheduling approach to the transmission of the traffic to improve fairness, i.e., the lower emission priority traffic is transmitted. Finally, some emission priority schemes provide a mixture of both priority and weighted schedulers.
Are used to determine the order in which traffic gets discarded. The traffic may get dropped due to network node congestion or when the traffic is out of profile, i.e., the traffic exceeds its prescribed amount of BW for some period of time. Under congestion, traffic with a higher discard priority gets dropped before traffic with a lower discard priority. Traffic with similar QoS performance can be sub divided using discard priorities. This allows the traffic to receive the same performance when the network node is not congested. However, when the network node is congested, the discard priority is used to drop the more eligible traffic first. Discard priorities also allow traffic with the same emission priority to be discarded when the traffic is out of profile. With out discard priorities traffic would need to be separated into different queues in a network node to provide service differentiation. This can be expensive since only a limited number of hardware queues (typically eight or less) are available on networking devices. Some devices may have software based queues but as these are increasingly used, network node performance is typically reduced.
With discard priorities, traffic can be placed in the same queue but in effect the queue is sub divided into virtual queues, each with a different discard priority. For example if a product supports three discard priorities, then one hardware queues in effect provides three QoS Levels.
Table 3.1 illustrates the QoS performance dimensions required by some common applications. Applications can have very different QoS requirements. As these are mixed over a common IP transport network, without applying QoS the network traffic will experience unpredictable behavior .
Networked applications can be categorized based on end user expectations or application requirements. Some applications are between people while other applications are a person and a networked device application, e.g., a PC and web server. Finally, some networking devices, e.g., router-to-router. Table 3.2 categorizes applications into four different traffic categories:
- Network Control
Some applications are interactive whereby two or more people actively participate. The participants expect the networked applications to respond in real time. In this context real time means that there is minimal delay (latency) and delay variations (jitter) between the sender an
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