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A New Method to Improve Voice over IP (VoIP) Bandwidth Utilization over Internet Telephony Transport Protocol (ITTP)
Abstract
The world witnessed a revolution of new technologies that serve humankind and make their life easier. Voice over Internet Protocol (VoIP) is one of such technologies. VoIP is a technology of making voice calls over an IP network. One of the problems that slow the spreading of VoIP is the inefficient bandwidth utilization that resulting from the big packet header. In this paper, we proposed a new method to enhance VoIP bandwidth employment over the Internet Telephony Transport Protocol (ITTP) protocol. The proposed method improves VoIP bandwidth utilization from two dimensions. The first dimension is by multiplexing several VoIP packets to the same receiving end in one header, instead of a separate header for each VoIP packet. The second dimension is by compressing the VoIP packet payload. The evaluation result of the proposed method showed a noticeable reduction of the consumed bandwidth, by up to 48.9%, in comparison to the traditional method of ITTP without VoIP packets multiplexing or VoIP packet payload compression.
... RTP collaborates with the UDP to have the option to transfer the voice information throughout computer networks. Through this, the VoIP packet preamble contains 12 bytes RTP, 8 bytes UDP and 20 bytes IP (40 bytes RTP/UDP/IP [RUI]) Abualhaj et al., 2019). Contrastingly, the VoIP codecs make a tiny speech frame (packet data) ranging from 10-30bytes. ...
... For instance, the part of the BW spent by the 40 bytes RUI preamble with each of the previous codecs is 74%, 80%, 57.1% and 66.6% separately. Subsequently, the large preamble length produced because of the RUI preamble is the primary driver to BW's wasteful utilisation Abualhaj et al., 2019;Roay et al., 2013). ...
... The vast majority of this data is nonessential to move the VoIP systems traffic, especially the client-to-client calls (Hartpence, 2013;Perkins, 2003;Gao, 2019). These irrelevant RUI preamble data are the primary drivers of the wasteful utilisation of the computer network BW Abualhaj et al., 2019;Roay et al., 2013). The packet aggregation and preamble compression are the two primary techniques intended to address the wasteful utilisation of the computer network BW. ...
The adoption of the Voice over Internet Protocol (VoIP) system is growing due to several factors, including its meagre rate and the numerous contours that can be joined with VoIP systems. However, the wasteful utilisation of the computer network is an inevitable problem that limits the rapid growth of VoIP systems. The essential explanation behind this wasteful utilisation of the computer network bandwidth (BW) is the considerable preamble length of the VoIP packet. In this study, we invent a technique that addresses the considerable preamble length of the VoIP packet. The designed technique is known as the manikin voice frame (MVF). The primary idea of the MVF technique is to utilise the VoIP packet preamble tuples that are not essential to the voice calls, particularly client-to-client calls (voice calls between only two users). Specifically, these tuples will be utilised for reserving the data of the VoIP packet. In certain instances, this will make the VoIP packet data manikin or even make it empty. The performance assessment of the introduced MVF technique demonstrated that the utilisation of the computer network BW has enhanced by 33%. Along these lines, the MVF technique indicates potential progress in resolving the inefficient usage of the computer network BW.
... Table 1 lists some of the most widely used VoIP codecs (Gupta et al., 2018;Ortega, 2019). The standard VoIP packet preamble, on the other hand, consists of 12 octets of RTP, 8 octets of UDP, and 20 octets of IP (40 octets of RUI) (Hussein et al., 2020;Abualhaj et al., 2019b). The addition of 40 octets of RTP/UDP/IP (RUI) preamble to a little VoIP packet data (10 to 30 octets) causes a significant increase in packet preamble overhead ranging from 57% to 80%. ...
... The addition of 40 octets of RTP/UDP/IP (RUI) preamble to a little VoIP packet data (10 to 30 octets) causes a significant increase in packet preamble overhead ranging from 57% to 80%. As a result, the BW of the network is squandered (Hussein et As previously stated, the primary cause of VoIP servers' inefficient BW is preamble overhead, caused by connecting a 40-octet RUI to a short VoIP packet data (Hussein et al., 2020;Abualhaj et al., 2019b;Vulkan et al., 2014). There are now two basic protocols in use: UDP and IP. ...
... Thus, the preamble compression methodology profoundly decreases preamble overhead, also enhancing BW utilization (Shambour et al., 2020;Hussein et al., 2020;Seytnazarov et al., 2018;Maqhat et al., 2014). Aside from these two methodologies, a novel protocol, named Internet Telephony Transport Protocol (ITTP), has been created to convey IP voice calls (Abualhaj et al., 2019b;Abu-Alhaj et al., 2012). In the next part, the ITTP protocol, as well as packet coalescence and preamble compression algorithms, will be thoroughly detailed. ...
Voice over IP (VoIP) is widely utilized by organizations, schools, colleges, and so on. Nevertheless, VoIP numerous challenges that hinder its spread. One of the significant challenges is the poor exploit of the VoIP technology network bandwidth (BW), caused by the huge preamble of the VoIP packet. This paper suggests a novel methodology to manage this huge preamble overhead challenge. The proposed methodology is named runt payload VoIP packet (RPV). The core principle of the RPV methodology is to reemploy and exploit the VoIP packet preamble’s data (fields) that are superfluous by VoIP technology, especially for unicast IP voice calls. Generally, those fields will be used to convey the VoIP packet payload. Consequently, diminish or zero the length of the payload and, therefore, spare the BW. The results of the investigation into the suggested RPV methodology indicated significant enhancement in the BW exploitation of VoIP technology. For instance, the saved BW in the examined environment with the LPC codec came to up to 25.9%.
... Tab. 1 shows some of the common VoIP codecs [8,9]. On one the other hand, the typical VoIP packet header consists of 12 bytes of realtime transfer protocol (RTP), 8 bytes of user diagram protocol (UDP), and 20 bytes of IP (a total of 40 bytes RTP/UDP/IP) [10,11]. Attaching 40 bytes RTP/UDP/IP header to a small VoIP packet payload (10-30 bytes) induces high packet header overhead, thereby resulting in wasted network bandwidth [10][11][12]. ...
... On one the other hand, the typical VoIP packet header consists of 12 bytes of realtime transfer protocol (RTP), 8 bytes of user diagram protocol (UDP), and 20 bytes of IP (a total of 40 bytes RTP/UDP/IP) [10,11]. Attaching 40 bytes RTP/UDP/IP header to a small VoIP packet payload (10-30 bytes) induces high packet header overhead, thereby resulting in wasted network bandwidth [10][11][12]. A packet header overhead is calculated by dividing the packet header size by the total packet size (header size plus payload size). ...
... As mentioned, the main reason for the inefficient bandwidth utilization of VoIP service is header overhead, which results from attaching a 40-byte RTP/UDP/IP header to a small VoIP packet payload [10][11][12]. The existing UDP and IP are general protocols used to carry all types of data [13]. ...
... Therefore, the Codec produces small voice frames size, typically, 10 to 30 bytes based on the used codec. Table I shows some of the common VoIP Codecs [9,10,11]. As for VoIP protocols, there are two types: signaling protocols and media transfer protocols [12,13]. ...
... H.323 and Session Initiation Protocol (SIP) are the two common signaling protocols [13,14]. On the other hand, the 12-bytes Real-time Transport Protocol (RTP), 6-bytes Internet Telephony Transport Protocol (ITTP), and 4-bytes Inter-Asterisk eXchange (IAX) are the main examples of media transfer protocols [9,13,15]. Both RTP and IAX take the help of the 8-bytes User Datagram Protocol (UDP) to be able to convey the voice data, while ITTP is able to carry the voice data by itself [9,13,15]. ...
... On the other hand, the 12-bytes Real-time Transport Protocol (RTP), 6-bytes Internet Telephony Transport Protocol (ITTP), and 4-bytes Inter-Asterisk eXchange (IAX) are the main examples of media transfer protocols [9,13,15]. Both RTP and IAX take the help of the 8-bytes User Datagram Protocol (UDP) to be able to convey the voice data, while ITTP is able to carry the voice data by itself [9,13,15]. As mentioned earlier, adding these protocols along with the 20-bytes IP protocol to the small VoIP packet payload leads to a considerable amount of the wasted bandwidth [7,8]. ...
... This preamble is made up of a 12-byte real-time transfer protocol (RTP), an 8-byte user datagram protocol (UDP), and a 20byte Internet Protocol (IP) protocol. Abualhaj et al., 2019b). This totals a large 40 byte RTP/UDP/IP preamble added to small digital voice chunks between 10 and 30 bytes. ...
... For simplicity, RTP/UDP/IP preamble will be abbreviated as RUI. Therefore, the used preamble consumes a considerable share of the IP network bandwidth that is dedicated to VoIP data Abualhaj et al., 2019b;Vulkan et al., 2014). For example, with each of the above codecs, the 40-byte RUI consumes 57.1%, 66.6%, 74%, and 80% of the bandwidth, respectively. ...
The use of Voice over Internet Protocol (VoIP) innovation is rising due to its various merits. Nevertheless, the ineffective use of bandwidth is a key dilemma that restricts the fast-rising use of VoIP innovation. The main factor behind this ineffective use of the bandwidth is the sizable VoIP packet preamble. This research creates a technique to address this dilemma of packet preamble. The created technique is known as payload elimination (PldE). The fundamental concept of the PldE technique is to exploit the information (elements) of the VoIP packet preamble that is superfluous for point-to-point calls. In general, these elements are utilized to transport the payload of VoIP packets. Consequently, the payload size of VoIP packet will be lowered or removed, preserving the available bandwidth. The performance test of the PldE technique indicated an improvement of up to 41.6% in the exploitation of IP network bandwidth. So, the PldE technique is showing signs that it could help solve the problem of the IP network's inefficient use of bandwidth.
... The ubiquitous of VoIP resulting from i) the big number of VoIP applications that provide a low rate or free VoIP calls, ii) VoIP applications can be used by any handheld device such as mobile phone, iPad, and laptop, and iii) the useful services that can be accompanied by VoIP calls including interactive voice recognition and transfer the voicemail to email [3] [4]. Nevertheless, bandwidth exploitation and quality of service are the two main issues that slow the VoIP propagation [4] [5]. This article focuses on VoIP application bandwidth exploitation. ...
... Fig. 1 shows the VoIP packet format when using ITTP protocol. Several approaches have been proposed by the researchers to improve VoIP bandwidth exploitation such as VoIP packet multiplexing, VoIP packet header compression, and VoIP packet payload compression [5][12] [13]. This article proposes a new VoIP packet payload compression method to improve VoIP bandwidth exploitation over ITTP/IP protocols. ...
... The PA-CH method preserves the allocated bandwidth through packet aggregation and uses redundant fields. For packet aggregation, the larger the size of the P-agg packet is, the more the allocated bandwidth is preserved (Abualhaj et al., 2016;Roay, 2013;Vulkan, 2014;Abualhaj et al., 2019). On one hand, sometimes the number of calls that running concurrently is inadequate to aggregate P-agg with a large size. ...
Old telecommunication systems are gradually being replaced by a new system that works over IP networks, which is known as voice over internet protocol (VoIP). VoIP has several merits (e.g., very cheap call rate), which make it increasingly popular in the telecommunication world. However, VoIP faces numerous impediments that decelerate its promotion. One of the top impediments is the wasted bandwidth caused by VoIP systems. Numerous methods have been proposed to handle this impediment, including packet aggregation methods. This paper proposes a novel aggregation method, called packet aggregation and carrier header (PA-CH), to reduce the amount of the large bandwidth caused by VoIP. As the name suggests, PA-CH saves in bandwidth by aggregating the packets in a header and using the redundant fields in the packet header to carry a portion of the packet voice data. The performance of the introduced PA-CH method was investigated based on three main metrics, namely, link capacity, allocated bandwidth reduction, and voice data shortening. Simulation results indicate that the proposed PA-CH method outperforms the comparison methods in three factors. For instance, the proposed method’s allocated bandwidth reduction ratio reaches 51% when the number of calls running concurrently reaches 100. Therefore, the proposed PA-CH method achieves its goal of reducing the wasted bandwidth caused by VoIP.
... The PM-CH technique protects the network bandwidth by packet multiplexing and the use of the Timestamp field. Clearly, the larger the M-pkt packet is, the better the bandwidth is protected [2,18,19]. However, the PM-CH technique ought to compromise between the length of the M-pkt packet and the forced delay to maintain a good call quality or protect the bandwidth. ...
Timeworn telecommunication are progressively being substituted by a new one that run over IP networks, which is recognized as voice over internet protocol (VoIP). VoIP has a number of qualities (e.g., inexpensive call rate), which make it progressively widespread in the telecommunication domain. However, VoIP faces plentiful obstacles that slow its growth. One of the major obstacles is poorly utilizing the network bandwidth. A number of techniques have been offered to handle this obstacle, including packet multiplexing techniques. This paper designs an original multiplexing techniques, called packet multiplexing and carrier header (PM-CH), to decrease the quantity of the bandwidth consumed by VoIP. PM-CH protect the bandwidth by multiplexing the packets in a header and using the Timestamp field in the RTP header. The achievement of the PM-CH technique was examined depends on connection capacity and payload shortening. Simulation outcomes show that the PM-CH technique outperforms the contrast technique in the two factors. For instance, the PM-CH technique’s connection capacity outperforms the comparable technique by 58.9% when the connection bandwidth is 1000 kbps. Consequently, the PM-CH technique attains its objective of reducing the unexploited bandwidth caused by VoIP.
... The major concern of IMS is involved in fixed-networks, WiMAX, or Wireless LAN to provide a guaranteed end-to-end service with acceptable quality due to several problems such as mobile handover and wireless signal fading. [8] [9]. ...
The IP Multimedia Subsystem (IMS) is recently expected to have the major architecture framework to be involved in the Next Generation Network. IMS works to bridge the multimedia communication among variety of applications over Internet. IMS bears its multimedia signals and stream through different means of transport protocols; TCP, UDP, and SCTP. TCP is a connection-oriented protocol that provides reliable data delivery and congestion control. To setup connection with TCP, IMS entities requires extra operations to complete, that operation process called (worker process) costs the multimedia server extra overload and delay. This paper investigates the indirect impacts of TCP connection that resulted from the Call Session Control Function (CSCF) servers when it deal with video communication. Two parameters are evaluated in experiment which are CPU usage and response time with two different scenarios. The experimental shows that outbound scenario performs better than the inbound scenario due to the extra operations required to setup new TCP connection for inbound
... The 12-byte Real-time Transport Protocol (RTP), 6byte Internet Telephony Transport Protocol (ITTP), and 4-byte Inter-Asterisk eXchange (IAX) are the main examples of media transfer protocols. Both RTP and IAX take the help of the 8-byte User Datagram Protocol (UDP) to be able to convey the voice data, while ITTP is able to carry the voice data by itself [10][11] [12]. As mentioned earlier, adding these protocols along with the 20-byte IP protocol to the small VoIP packet payload leads to a considerable amount of BW waste. ...
... Therefore, VoIP packet might be delayed or lost which degrades the call quality. Apart from call quality, VoIP encounter a bandwidth efficiency utilization dilemma that needs to be dealt with and overcome [5,6]. ...
The inefficient use of the IP network bandwidth is a fundamental issue that restricts the exponential spreading of Voice over IP (VoIP). The primary reason for this is the big header size of the VoIP packet. In this paper, we propose a method, called Short Voice Frame (SVF), that addresses the big header size of the VoIP packet. The main idea of the SVF method is to make effective use of the VoIP packet header fields that are unneeded to the VoIP technology. In particular, these fields will be used for temporarily buffering the voice frame (VoIP packet payload) data. This will make the VoIP packet payload short or even zero in some cases. The performance evaluation of the proposed SVF method showed that the use of the IP network bandwidth has improved by up to 28.3% when using the G.723.1 codec.
Voice over IP (VoIP) wastes a valuable amount of bandwidth because of its large packet header size compared to its small packet payload. The main objective of this paper is to reduce the amount of this wasted bandwidth, by proposing a new packets coalescence method, called Payload Shrinking and Packets Coalesce (PS-PC). The proposed PS-PC method reduces the amount of the wasted bandwidth by i) coalesces a group of VoIP packets in one header instead of a separate header to each packet and ii) shrinks the VoIP packet payload to a smaller one based on a certain algorithm. The proposed PS-PC method is deployed at the sender side VoIP gateway that represents an exit point to a myriad number of simultaneous VoIP calls. The performance evaluation showed better bandwidth usage when deploying the proposed PS-PC method with ITTP protocol in comparison to the traditional ITTP protocol without the PS-PC method.
The Junction-less tunnel field-effect transistor (JLTFET) is a captivating device due to its excellent electrical properties and less variability in comparison to MOSFET at the nanometer regime. In this regard, we investigate a silicon-based pocket doped JLTFET in which an InAs pocket is inserted across the source-channel junction to enhance tunneling probability. In this respect, we have considered analog/RF and DC Figure of merit analysis for the conventional and pocket doped JLTFET (PD- JLTFET) in terms of an electric field, transfer characteristics, transconductance, parasitic capacitances, cut-off frequency, gain-bandwidth product, and maximum oscillation frequency. Additionally, we have examined the effect of spacer length variation across the junction between source and channel. The ATLAS device simulator is used for the simulations of the conventional JLTFET and PD-JLTFET. The proposed PD-JLTFET has shown a higher ION/IOFF ratio (~1013) and improved sub-threshold swing (~9.08 mV/decade). The notable characteristics demonstrated by PD-JLTFET make it an optimum device for low power and high switching application.
This article presents a low power consumption and fast voltage level shifter, which can increase the input voltage level lower or comparable to a threshold voltage to nominal voltage level and high voltage level to low voltage level. The proposed level shifter is based on a pull-up network which has regulated cross-coupled structure. This helps to enhance the switching speed and reduction of dynamic power. Furthermore, split input inverter and pMOS diode used to reduce the static power consumption. For proposed level shifter simulation results have performed in a 180 nm CMOS technology using cadence virtuoso tool. While converting supply voltage from low to the high voltage level of 0.4 V and 1.8 V, respectively at applied frequency 1 MHz, power consumption and propagation delays are 139.6 nW and 19.92 ns, respectively. And during conversion high to the low supply voltage are 1.8 V and 0.4 V, 120.2 nW and 15.31 ns, respectively.
In a wireless communication system, the Quality of Service (QoS) becomes an active area of research as the demand for real-time applications like voice or video communication increases day by day. In this paper, a new hybrid INTER-SR & INTRA-SR based reliable wireless network is designed for Voice over Internet Protocol (VoIP) applications. It is mainly consisting of one optimization algorithm named as Genetic Algorithm (GA) in conjunction with two classification approaches such as Artificial Neural Network (ANN) and Support Vector Machine (SVM). The proposed algorithm selects a reliable and more reliable route based on the nodes properties such as energy consumption and delay. In the end, three distinct QoS parameters such as routing overhead, delay and jitter are analyzed for two communication scenarios such as with attack and without attack in MANET simulator. The results demonstrate that the proposed algorithm performs with satisfying results compared to individual routing and optimization approaches.
The number of applications running over computer networks has been increasing tremendously, which increased the number of packets running over the network as well leading to resource contention, which ultimately results in congestion. Congestion increases both delay and packet loss while reducing bandwidth utilization and degrading network performance. Network congestion can be controlled by several methods, such as random early detection (RED), which is the most well-known and widely used method to alleviate problems caused by congestion. However, RED and its variants suffer from linearity and parametrization problems. In this paper, we proposed a new method called fuzzy logic RED (FLRED), which extends RED by integrating fuzzy logic to overcome these problems. The proposed FLRED method relies on the average queue length (aql) and the speculated delay (DSpec) to predict and avoid congestion at an early stage. A discrete-time queue model is used to simulate and evaluate FLRED. The results showed that FLRED outperformed both RED and effective RED (ERED) by decreasing both delay and packet loss under heavy congestion. Compared with ERED and RED, FLRED decreased the delay by up to 1.5 and 4.5% and reduced packet loss by up to 6 and 30%, respectively, under heavy congestion. These findings suggest that FLRED is a promising congestion method that can save network resources and improve overall performance.
Wireless mesh networks (WMNs) have been increasingly applied in private and public networks during the last decade. In a different context, voice over IP (VoIP) has emerged as a new technology for making voice calls around the world over IP networks and is replacing traditional telecommunication systems. The popularity of the two technologies motivated the deployment of VoIP over WMNs. However, VoIP over WMNs suffers from inefficient bandwidth utilization because of two reasons: i) attaching 40-byte RTP/UDP/IP header to a small VoIP payload (e.g., 10 bytes) and ii) 841 µs delay overhead of each packet in WMNs. Among several solutions, VoIP packet multiplexing is the most prominent one. This technique combines several VoIP packets in one header. In this study, we will survey all the VoIP multiplexing methods over WMNs. This study provides a clear understanding of the VoIP bandwidth utilization problem over WMNs, discusses the general approaches in which packet multiplexing methods could be performed, provides a detailed study of present multiplexing techniques, shows the aspects that hinder the VoIP multiplexing methods, discusses the factors affected by VoIP multiplexing schemes, shows the merits and demerits of different multiplexing approaches, provides guidelines for designing a new improved multiplexing technique, and provides directions for future research. This study contributes by providing guidance for designing a suitable and robust method to multiplex VoIP packets over WMNs.
Over the past few years, the telecommunications sector has started moving toward the use of Voice over Internet Protocol (VoIP) technology. VoIP technology employs Internet infrastructure and protocols to transfer VoIP data between call parties. Unfortunately, none of the existing Internet transport layer protocols address VoIP application requirements. Typically, the Real-time Transport Protocol (RTP) application layer protocol and the User Datagram Protocol (UDP) transport layer protocol are bound together to address VoIP application requirements. However, a combination of RTP and UDP reduces the quality of VoIP applications and causes inefficient bandwidth utilization. In the present work, a dedicated transport protocol named Internet Telephony Transport Protocol (ITTP) was designed to carry VoIP application data. ITTP is designed to address key VoIP requirements and handle the problems resulting from RTP/UDP. A simple mathematical model was used to evaluate ITTP bandwidth efficiency and compare it with RTP/UDP. ITTP was found to improve bandwidth usage substantially. In addition, ITTP was simulated using Network Simulation 2. The results showed that ITTP had better performance compared with RTP/UDP in terms of packet loss, delay and bandwidth usage.
VoIP technology is becoming increasingly important as they gain market share in PSTN technology. In the short run, it is likely that VoIP technology will dominate the telecommunications market. This evolution towards VoIP technology makes sense for improving the VoIP technology efficiency under the auspices of the internet. However, the PSTN still provide voice quality better than the VoIP technology voice quality. This would be due to many contributing factors such as delay and packet loss. Apart from internet hardware infrastructure, the VoIP transfer protocols cause the delay and packet loss problems as well. Typically, the Real-time Media Transfer Protocols (RTMTPs), such as the Real-time Transport Protocol (RTP), are working in conjunction with the transport layer protocols (TLPs) to carry the VoIP applications data. However, some of the TLPs protocols are transfer the VoIP applications data by themselves. The aim of this paper is to present the capability of the TLPs protocols to transfer the VoIP applications data, and the obstacles face these protocols to do so. For easier comparison and discussion, we have divided TLPs protocols into three groups, the Reliable Transport Layer Protocols (RTLPs), Unreliable Transport Layer Protocols (UTLPs), and Reliable and Unreliable Transport Layer Protocols (RUTLPs). We have studied the three groups from VoIP point of view. In addition, we have showed that the UTLPs group needs to work with the RTMTPs to be able to transfer the VoIP applications data. Finally, we have showed the problems caused by each group to the VoIP applications. As result, a comparison among the transport layer protocols is presented according to the problems resulting from theses protocols when transferring the VoIP applications data. The comparison can help to choose the suitable protocols to carry the VoIP data. However, after studied all the TLPs protocols, the paper recommend to use the UTLPs group in conjunction with the RTMTPs, in order to transfer the VoIP applications data.
This document describes a method for compressing the headers of IP/UDP/RTP datagrams to reduce overhead on low-speed serial links. In many cases, all three headers can be compressed to 2-4 bytes.
Voice over IP (VoIP) is expected to be the key of the communication industry in near future. Inefficient bandwidth usage by VoIP consumed the network bandwidth. The problem of inefficient bandwidth usage is handled by several techniques such as silence suppression, packets multiplexing, and voice compression. This work proposed new architecture to save network bandwidth for VoIP. We called the new architecture MuxComp. MuxComp, combines packets multiplexing and voice compression. Whereas, multiplexing the voice packets reduce the overheads, furthermore give the opportunity to compress the packets again. The MuxComp architecture consists of two entities. The MuxCmp entity is VoIP gateway in the sender side, which performs the multiplexing and compression processes to the packets and sends them to their destinations. The DCmpDMux entity is VoIP gateway in the receiver side, which performs the reverse processes of MuxCmp through de-compression and de-multiplexing the packets. Furthermore, this work is considered the size of the multiplexed packets, whereas the size must not exceed the network Maximum Transfer Unit (MTU), therefore avoiding the fragmentation delay through the network.
In recent, years, the field of telecommunications began to move toward Voice over Internet Protocol (VoIP) technology. VoIP technology applications generate packets with small payload sizes to minimize packetization delay. That, is, increasing the header overhead wastes the network link bandwidth. Packet multiplexing is a mechanism to improve the exploitation of network link bandwidth. Various multiplexing mechanisms are proposed to improve link bandwidth exploitation when using RTP/UDP protocols. This paper proposes an efficient, multiplexing mechanism, called ITTP Multiplexing (ITTP-Mux). Unlike previous mechanisms, ITTP-Mux mechanism, multiplexes VoIP packets when using ITTP protocol, not, RTP/UDP. In addition, the ITTP-Mux mechanism assembles VoIP packets, which exist, in the same path, in a single IP header instead of an IP header to each packet. Therefore, header overhead is lessened and network link bandwidth is saved. ITTP-Mux also adds 1-byte mini-header to each packet to distinguish the assembled packets. The proposed. ITTP-Mux mechanism is simulated and compared with the traditional ITTP protocol (without, multiplexing) using five factors, namely, number of calls, goodput, header overhead, bandwidth usage, and saved bandwidth. Based on all these factors, ITTP-Mux mechanism, outperforms the traditional ITTP protocol. For example, the result shows that, bandwidth usage is improved by up to 29.1% in the tested cases.
In a wireless multi-hop network environment, energy consumption of mobile nodes is an important factor for the performance evaluation of network life-time. In Voice over IP (VoIP) service, the redundant data size of a VoIP packet such as TCP/IP headers is much larger than the voice data size of a VoIP packet. Such an inefficient structure of VoIP packet causes heavy energy waste in mobile nodes. In order to alleviate the effect of VoIP packet transmission on energy consumption, a packet aggregation algorithm that transmits one large VoIP packet by combining multiple small VoIP packets has been studied. However, when excessively many VoIP packets are combined, it may cause deterioration of the QoS of VoIP service, especially for end-to-end delay. In this paper, we analyze the effect of the packet aggregation algorithm on both VoIP service quality and the energy consumption of mobile nodes in a wireless multi-hop environment. We build the cost function that describes the degree of trade-off between the QoS of VoIP services and the energy consumption of a mobile node. By using this cost function, we get the optimum number of VoIP packets to be combined in the packet aggregation scheme under various wireless channel conditions. We expect this study to contribute to providing guidance on balancing the QoS of VoIP service and energy consumption of a mobile node when the packet aggregation algorithm is applied to VoIP service in a wireless multi-hop networks.
Gradually, Voice over Internet Protocol (VoIP) has been dominating the telecommunications world. Unfortunately, its applications are injecting a huge number of small packets in the network, which produces high overhead and therefore wastes network bandwidth. This paper proposed the use of a novel multiplexing technique, Delta-Multiplexing, to save the wasted bandwidth. In the Delta-Multiplexing technique, the VoIP packets destined to the same destination gateway are aggregated in a single UDP/IP header, therefore reducing the header overhead and saving network bandwidth. Moreover, the Delta-Multiplexing technique reduces the size of the packets payload by transmitting the difference between the consecutive packets payloads. Accordingly, the Delta-Multiplexing technique greatly saves bandwidth. We have simulated the Delta-Multiplexing technique using a 14-byte LPC codec. The result showed that Delta-Multiplexing is capable of saving between 68% and 72% as compared to conventional techniques (without multiplexing). Moreover, the Delta-Multiplexing technique reduces the number of VoIP packets running over the network, therefore reducing network traffic, overload, and congestion, thus improving the overall network performance.
IP telephony systems are expected to be deployed worldwide in the
near future because of their potential for integrating the multimedia
communication infrastructure over IP networks. Phone-to-phone connection
over an IP network via IP telephony gateways (IP-GWs) is a key feature
of the system. In an IP telephony system, a low-bit-rate voice codec is
used to improve bandwidth efficiency. However, due to the packet
transfer method over the IP network, it is necessary to add packet
headers, including IP, UDP, and RTP headers, which increases the header
overhead and thus decreases transfer efficiency. Moreover, because there
will be large numbers of short voice packets flowing into the IP
network, the load on the Internet will increase. We propose voice stream
multiplexing between IP-GWs to solve these problems. In this scheme,
multiple voice streams are connected between a pair of IPCWs, enabling
multiplexed voice stream transfer. The voice stream multiplexing
mechanism can reduce the header overhead as well as decrease the number
of voice packets. The voice stream multiplexing we propose is to
concatenate RTP packets destined for the same IP-GW at a multiplexing
interval period into a single UDP packet. The advantage of this method
is that no new additional header is required and the current
well-defined H.323 and RTP standards can be applied with minimum
changes. We implemented and tested the system
Voice communications such as telephony are delay sensitive. Existing voice-over-IP (VoIP) applications transmit voice data in packets of very small size to minimize packetization delay, causing very inefficient use of network bandwidth. This paper proposes a multiplexing scheme for improving the bandwidth efficiency of existing VoIP applications. By installing a multiplexer in an H.323 proxy, voice packets from multiple sources are combined into one IP packet for transmission. A demultiplexer at the receiver-end proxy restores the original voice packets before delivering them to the end-user applications. Results show that the multiplexing scheme can increase bandwidth efficiency by as much as 300%. The multiplexing scheme is fully compatible with existing H.323-compliant VoIP applications and can be readily deployed.
Comparative Evaluation and Analysis of IAX and RSW", international journal of computer science and information security (ijcsis)
- M S Kolhar
- M M Abu-Alhaj
- O Abouabdalla
- T C Wan
- A M Manasrah
M. S. Kolhar, M. M. Abu-Alhaj, O. Abouabdalla, T.C. Wan,
and A. M. Manasrah, "Comparative Evaluation and Analysis
of IAX and RSW", international journal of computer
science and information security (ijcsis), 13, January, 2010.
Multiplexing SIP Applications Voice Packets Between SWVG Gateways
- M Abualhaj
- M S Kolhar
- M Halaiyqah
- O Abouabdalla
- R Sureswaran
M. AbuAlhaj, M. S. Kolhar, M. Halaiyqah, O. Abouabdalla,
and R. Sureswaran, "Multiplexing SIP Applications Voice
Packets Between SWVG Gateways", In Proceedings of
International Conference on Computer Engineering and
Applications (ICCEA 2009), Manila, 2009, 1-5