A Survey of Bandwidth Optimization Techniques and Patterns in VoIP Services and Applications <s> INTRODUCTION <s> Interet telephony is possibly the fastest-growing part of communications today. This article discusses what exactly it is, who needs it, and how it works. Internet telephony, or voice over Internet protocol (VoIP), is the provision of phone service over the Internet. But in sharp contrast with conventional telephony, it carries voice traffic as data packets over a packet-switched data network instead of as a synchronous stream of binary data over a circuit-switched, time-division multiplexed (TDM) voice network. There are some substantial benefits (as well as some sticky problems) to the scheme, which is why companies and individuals are finding it increasingly attractive. <s> BIB001 </s> A Survey of Bandwidth Optimization Techniques and Patterns in VoIP Services and Applications <s> INTRODUCTION <s> In the recent past, there has been a tremendous increase in the popularity of VoIP services as a result of huge growth in broadband access. The same voice-over-Internet protocol (VoIP) service poses new challenges when deployed over a wireless mesh network, while enabling users to make voice calls using WiFi phones. Packet losses and delay due to interference in a multiple-hop mesh network with limited capacity can significantly degrade the end-to-end VoIP call quality. In this work, we discuss the basic requirements for efficient deployment of VoIP services over a mesh network. We present and evaluate practical optimizing techniques that can enhance the network capacity, maintain the VoIP quality and handle user mobility efficiently. Extensive experiments conducted on a real testbed and ns-2 provide insights into the performance issues and demonstrate the level of improvement that can be obtained by the proposed techniques. Specifically, we find that packet aggregation along with header compression can increase the number of supported VoIP calls in a multihop network by 2-3 times. The proposed fast path switching is highly effective in maintaining the VoIP quality. Our fast handoff scheme achieves almost negligible disruption during calls to roaming clients <s> BIB002 </s> A Survey of Bandwidth Optimization Techniques and Patterns in VoIP Services and Applications <s> INTRODUCTION <s> Voice over IP (VoIP) service providers implementing different protocols in their core networks are required to coordinate in order to provision, support and deliver voice services to end clients. This paper presents a performance measurement experience for the transport of voice services across multiple heterogeneous wired networks. Our test results show that Multiple Protocol Label Switching (MPLS) protocol slightly outperforms IP in delivering voice services. Moreover, the type of call signaling protocol used, tunneling category adopted and the type of Virtual Private Network (VPN) implemented affect the overall quality and performance of voice services in terms of jitter and delay. <s> BIB003 </s> A Survey of Bandwidth Optimization Techniques and Patterns in VoIP Services and Applications <s> INTRODUCTION <s> Voice over Internet Protocol (VoIP) is a transmission technology that allows voice to be delivered over the Internet with support of Internet Protocol. Recently, the integration of VoIP and Wireless Local Area Network (WLAN) has become popular, driven by mobility requirements of the users. However, WLAN architecture was primarily designed to carry data and not for voice. Hence, there is a need to enhance the quality of voice transmission over the WLAN. Several metrics such as delay, jitter, and packet lost have impact on the quality of service (QoS) of the VoIP system. Minimizing total transmission delay, jitter and packet lost would mean improving the QoS of VoIP. In this paper, a cross-layer interaction approach is proposed to improve the quality of voice delivery, hence the QoS of VoIP. The approach in achieving this is by introducing a communication agent between Data Link layer and Transport layer, by which congestion misinterpretation by the Transport layer could now be eliminated. In particular, the IP protocol of Transport layer will be utilized to communicate with the agent to improve the QoS of the VoIP system. <s> BIB004 </s> A Survey of Bandwidth Optimization Techniques and Patterns in VoIP Services and Applications <s> INTRODUCTION <s> Voice over Internet Protocol (VoIP) is a voice communication system that proffers the transmission and reception of audio (voice) signal and (or) data over the internet. One very crucial problem in a VoIP network is inefficient bandwidth utilization which in most case is caused by header overhead resulting from the attachment of a big (40 bytes) to small payloads (10 to 30 bytes). Given the need to reduce the amount of bandwidth being wasted or perhaps improving the utilization of bandwidth which ultimately could enhance network performance and VoIP quality of service, several schemes and techniques have been proposed. The improvement and (or) optimization of bandwidth can be achieved through; packet header reduction and defines a process of multiplexing and de-multiplexing packet headers to curb overflows. Silence suppression offer a technique of repressing the silent portions (packets) in a voice conversation using Voice Activity Detection algorithm, as a result, the transmission rate during the inactive periods of speech is reduced, and thus, the average transmission rate can be reduced. Voice/ Packet Header compression is considered the most productive of all the technique, and it presents an event whereby VoIP packets are compressed from the conventional 40 bytes of size to a smaller byte size (2 bytes in the best case). Bandwidth save is reached using compression and decompression codecs of varying data and bit rates. It is envisaged that an improvement in the performance of codecs would yield a better result in terms of optimizing bandwidth in a VoIP network. <s> BIB005 </s> A Survey of Bandwidth Optimization Techniques and Patterns in VoIP Services and Applications <s> INTRODUCTION <s> Deploying wireless campus network becomes popular in many world universities for the services that are provided. However, it suffers from different issues such as low VoIP network capacity, network congestion effect on VoIP QoS and WLAN multi rate issue due to linkadaptation technique. In this paper a cross layer call admission control (CCAC) scheme is proposed to reduce the effects of these problems on VoWLAN based on monitoring RTCPRR(Real Time Control Protocol ReceiverReport) that provides the QoS level for VoIP and monitoring the MAC layer for any change in the data rate. If the QoS level degrades due to one of the aforementioned reasons, a considerable change in the packet size or the codec type will be the solution. A wireless campus network issimulated using OPNET 14.5 modeler and many scenarios are modeled to improve this proposed scheme. <s> BIB006
The volume of voice traffic on IP networks continues to increase at speedy strides, with substantial growth in the use Skype, Tango and other voice applications. Voice over Internet Protocol (VoIP) also called also called, IP Telephony, Internet telephony, Broadband telephony, Broadband Phone and Voice over Broadband  a technology that unveils a communication system that allows for the conduction and reception of voice signals and data over the internet network. While using the Internet Protocol as the most basic transport mode over which both TCP and UDP are utilized BIB005 , VoIP system uses designated codecs that convert voice signals into digital data forms (bits)  yielding an output that is transmitted through networked infrastructure over the internet. These bits are usually reconstructed at the destination using attribute data and timestamps accordingly. Equitable cost savings, ease of deployment while leveraging on existing infrastructure BIB002 , scalability, improved productivity, flexibility and mobility  are key drivers for the proliferation of this technology. FCC predictions were that by 2008, 44% of corporate organizations were going to run VoIP lines BIB002 . It is so envisioned that in time, VoIP technology (applications and schemes) will substitute traditional Public Switched Telephone Network (PSTN) Technology BIB001 in BIB005 . This ascendency over PSTN is and will continue to be driven by numerous factors; notably higher reliability due to systematic and automated side-stepping of inherent challenges of network over congestions; with the ability to execute calls, regardless of geographical location using computers or other portable mobile devices  . However, voice traffic operates in real-time and is centered on small UDP packets, thus, incredible network loads are exerted attendant devices due to the emergence of huge packets per second per voice call  . Additively, voice applications that are VoIP-enabled are greatly inefficient in bandwidth given that packet header sizes are often same as the payload size with overheads of nearly 40%  . More so, acceptable use standards advocate for efficient Quality of Service, a paradigm that plainly describes the gratification level experienced by users of any VoIP application. Technically, a user will recognize fragmentations in voice transmission and poor quality when high delay, loss or distortions occur. Therefore, choices for packet sizes are made based on network delays, through accessible bandwidth and packet losses inside the network. One very crucial problem in a VoIP network is the potential wastage of bandwidth caused by the earlier noted inefficiency also caused by attachment of huge (typically 40 bytes) unwanted bandwidth to smaller payloads ( typically 10 to 30 bytes) desirable bandwidth (payloads). Consequently, the need to reduce bandwidth wastage; or better still improve the utilization of bandwidth which could eventually boost VoIP quality and network performance cannot be overstated. Maintaining excellent quality voice services to users is a major concern for VoIP service providers  in BIB003 ; a VoIP service that is fast (no or minimal delay) and efficient. However, -accelerating‖ VoIP traffic is quite vague, and often misleading. Why? Because VoIP traffic like all other IP traffics, moves at the speed of light. It can thus be delayed by network congestion and other relative issues that could amount to despoiled voice quality. Nonetheless -accelerating‖ VoIP traffic makes no contextual sense since VoIP packets cannot be made to travel faster than the current speed of light, and (or) participants cannot be made to talk any faster than they are able on telephony applications  . From a technical viewpoint and following tolerable standards, VoIP datagrams must be conveyed by the network with a negligible amount of jitter in order to preserve its quality. Latency must be retained below 150ms  if possible, and any techniques adopted to expedite traffic delivery must eliminate or reduce latency and jitter to the nearest minimum. Nonconformities to these standards will explicitly affect voice quality. This paper survey the various schemes ad techniques adoptable for the optimization of bandwidth in Voice-Over Internet Protocol Networks and (or) Services Several schemes and(or) techniques have been suggested as viable solutions for enhancing bandwidth usage, and this discourse seeks to review the varying techniques with a view to ascertaining respective weaknesses. Figure 1 illustrates a sample VoIP Setup while figure 2 shows the VoIP protocol stack with respect to its TCP/IP protocol standard BIB004 in BIB006 Figure 2: VoIP implementation in TCP/IP protocol standard BIB004 in BIB006 II. VOIP BANDWIDTH OPTIMIZATION: SPURS AND TECHNIQUES.