Voice Over Internet Protocol Packet Research Articles

MVF: A Novel Technique to Reduce the Voip Packet Payload Length

Abstract 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.

Improving VoIP Bandwidth Utilization Using the PldE Technique

Abstract 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.

Zeroize: A New Method to Improve the Utilization of 5G Networks When Running VoIP over IPv6

5G technology is spreading extremely quickly. Many services, including Voice Over Internet Protocol (VoIP), have utilized the features of 5G technology to improve their performance. VoIP service is gradually ruling the telecommunication sector due to its various advantages (e.g., free calls). However, VoIP service wastes a substantial share of the VoIP 5G network’s bandwidth due to its lengthy packet header. For instance, the share of the packet header from bandwidth and channel time reaches 85.7% of VoIP 5G networks when using the IPv6 protocol. VoIP designers are exerting considerable efforts to solve this issue. This paper contributes to these efforts by designing a new technique named Zeroize (zero sizes). The core of the Zeroize technique is based on utilizing the unnecessary fields of the IPv6 protocol header to keep the packet payload (voice data), thereby reducing or “zeroizing” the payload of the VoIP packet. The Zeroize technique substantially reduces the expanded bandwidth of VoIP 5G networks, which is reflected in the wasted channel time. The results show that the Zeroize technique reduces the wasted bandwidth by 20% with the G.723.1 codec. Therefore, this technique successfully reduces the bandwidth and channel time of VoIP 5G networks when using the IPv6 protocol.

Down to Zero Size of VoIP Packet Payload

Voice over Internet Protocol (VoIP) is widely used by companies, schools, universities, and other institutions. However, VoIP faces many issues that slow down its propagation. An important issue is poor utilization of the VoIP service network bandwidth, which results from the large header of the VoIP packet. The objective of this study is to handle this poor utilization of the network bandwidth. Therefore, this study proposes a novel method to address this large header overhead problem. The proposed method is called zero size payload (ZSP), which aims to reemploy and use the header information (fields) of the VoIP packet that is dispensable to the VoIP service, particularly the unicast IP voice calls. In general, these fields are used to carry the VoIP packet payload. Therefore, the size of the payload is reduced to save bandwidth. The performance estimation results of the proposed ZSP method showed a considerable improvement in the bandwidth utilization of the VoIP service. For example, the saved bandwidth in the tested scenario with the G.723.1, G.729, and LPC codecs reached 32%, 28%, and 26% respectively.

ENHANCEMENT OF VoIP PERFROMANCE IN MANET USING FUZZY LOGIC

Voiceover Internet Protocol (VoIP) application is a vital technology that is quickly growing in the Mobile Ad hoc NETwork (MANET). Packet loss is a factor that can significantly affect the Quality of Service (QoS) for VoIP performance. Due to the dynamic nature of MANET, it is a challenging task to maintain the desired packet loss rate. This paper aims to enhance the performance of VoIP in the MANET using a fuzzy logic model. The input for the model is VoIP packet loss and the outputs are the optimal parameters of MANET (node number, pause time, maximum speed, and maximum connection).Network Simulator (NS2) was used to perform all simulations. MATLAB was used to implement the proposed fuzzy model. Moreover, the performance of the model was evaluated using NS2, and the results show that our proposed fuzzy model offers a significant enhancement in terms of the VoIP packet loss rate (P.LR).

Effective Voice Frame Shrinking Method to Enhance VoIP Bandwidth Exploitation

The traditional telecommunication system (e.g. landline telephone system) are increasingly being replaced by Voice over Internet Protocol (VoIP) systems because of the very low or free rate. However, one of the main handicaps of VoIP adoption is the inefficient bandwidth exploitation issue. A key approach to handle this issue is packet multiplexing. This article proposes a new VoIP packet payload compression method that enhances bandwidth exploitation over Internet Telephony Transport Protocol (ITTP) protocol. The proposed method is called payload shrinking over ITTP (ITTP-PS). As the name implies, the proposed ITTP-PS method shrinks the VoIP packet payload based on a certain mechanism. The ITTP-PS method has two entities, namely, sender ITTP-PS (S-ITTP-PS) and receiver ITTP-PS (R-ITTP-PS). The main function of the S-ITTP-PS entity is to shrink the VoIP packet payload, while the main function of the R-ITTP-PS entity is restoring the VoIP packet payload to its normal size. To perform the R-ITTP-PS entity function, the ITTP-PS method will reemploy the flag bits in the IP protocol header. The ITTP-PS method has been implemented and compared with traditional ITTP protocol without shrinking the VoIP packet payload. The comparison based on the VoIP packet payload shrinking ratio and isochronous calls capacity improvement ratio. The result showed that the VoIP packet payload shrinking ratio has enhanced by up to around 20%, while the isochronous calls capacity improvement ratio has enhanced by up to around 9.5%. Therefore, enhancing the VoIP bandwidth exploitation over ITTP protocol.

ITTP-PG: A Novel Grouping Technique to Enhance VoIP Service Bandwidth Utilization

Recently, the field of telecommunications started to migrate to Voice over Internet Protocol (VoIP) service. VoIP service applications produce packets with short payload sizes to reduce packetization delay. That is, increasing the preamble size and expends the network link bandwidth. Packet grouping is a technique to enhance the employment of network link bandwidth. Numerous grouping techniques are suggested to enhance link bandwidth employment when using RTP/UDP protocols. Unlike previous research, this article suggests a packet grouping technique that works over the Internet Telephony Transport Protocol (ITTP), not RTP/UDP. This technique is called ITTP Packet Grouping (ITTP-PG). The ITTP-PG technique groups VoIP packets, which exist in the same route, in a single ITTP/IP preamble instead of an ITTP/IP preamble to each packet. Consequently, preamble size is diminished and network link bandwidth is saved. ITTP-PG also adds 3-byte runt-preamble to each packet to distinguish the grouped packets. The suggested ITTP-PG technique is simulated and compared with the conventional ITTP protocol (without grouping) using three elements, namely, the number of concurrent VoIP calls, preamble overhead, and bandwidth usage. Based on all these elements, the ITTP-PG technique outperforms the conventional ITTP protocol. For example, the result shows that bandwidth usage improved by up to 45.9% in the tested cases.

Enhancing VoIP BW Utilization over ITTP Protocol

The revolution of Voice over Internet Protocol (VoIP) technology has propagated everywhere and replaced the conventional telecommunication technology (e.g. landline). Nevertheless, several enhancements need to be done on VoIP technology to improve its performance. One of the main issues is to improve the VoIP network bandwidth (BW) utilization. VoIP packet payload compression is one of the key approaches to do that. This paper proposes a new method to compress VoIP packet payload. The suggested method works over internet telephony transport protocol (ITTP) and named Delta-ITTP method. The core idea of the Delta-ITTP method is to find and transmit the delta between the successive VoIP packet payloads, which is typically smaller than the original VoIP packet payload. The suggested Delta-ITTP method implements VoIP packet payload compression at the sender side and decompression at the receiver side. During the compression process, the Delta-ITTP method needs to keep some values to restore the original VoIP packet payload at the receiver side. For this, the Delta-ITTP method utilizes some of the IP protocol fields and no additional header is needed. The Delta-ITTP method has been deployed and compared with the traditional ITTP protocol without compression. The result showed that up to 19% BW saving was achieved in the tested cases leading to the desired enhancement in the VoIP network BW utilization.

Probabilistic classification of quality of service in wireless computer networks

There is an increasing reliance on wireless computer networks for communicating various types of time sensitive applications such as voice over internet protocol (VoIP). Quality of service (QoS) can play an important role in wireless computer networks as it can facilitate evaluation of their performance and can provide mechanisms to improve their operation. In this study probabilistic neural network (PNN) and Bayesian classification were developed to process delay, jitter and percentage packet loss ratio for VoIP traffic. Both methods successfully categorized the transmission of VoIP packets into low, medium and high QoS categories but overall the Bayesian approach performed more accurately than PNN. By accurately determining the network's QoS, an improved understanding of its performance is obtained.

Covert Voice over Internet Protocol Communications with Packet Loss Based on Fractal Interpolation

The last few years have witnessed an explosive growth in the research of information hiding in multimedia objects, but few studies have taken into account packet loss in multimedia networks. As one of the most popular real-time services in the Internet, Voice over Internet Protocol (VoIP) contributes to a large part of network traffic for its advantages of real time, high flow, and low cost. So packet loss is inevitable in multimedia networks and affects the performance of VoIP communications. In this study, a fractal-based VoIP steganographic approach was proposed to realize covert VoIP communications in the presence of packet loss. In the proposed scheme, secret data to be hidden were divided into blocks after being encrypted with the block cipher, and each block of the secret data was then embedded into VoIP streaming packets. The VoIP packets went through a packet-loss system based on Gilbert model which simulates a real network situation. And a prediction model based on fractal interpolation was built to decide whether a VoIP packet was suitable for data hiding. The experimental results indicated that the speech quality degradation increased with the escalating packet-loss level. The average variance of speech quality metrics (PESQ score) between the “no-embedding” speech samples and the “with-embedding” stego-speech samples was about 0.717, and the variances narrowed with the increasing packet-loss level. Both the average PESQ scores and the SNR values of stego-speech samples and the data-retrieving rates had almost the same varying trends when the packet-loss level increased, indicating that the success rate of the fractal prediction model played an important role in the performance of covert VoIP communications.

A survey on VoIP packet loss techniques

Voice over internet protocol (VoIP) is a group of hardware and software that facilitates people to utilise the internet as the transmission medium for telephone calls by transmitting voice data in packets using IP instead of using conventional circuit transmissions of the public switched telephone network (PSTN). VoIP is also considered as internet telephony, IP telephony or voice over the internet (VOI). VoIP communication practically needs a high-speed internet connection for consistent functionality. There are several internet telephony applications existing at present. VoIP has established necessities for data packets to arrive at their target in a more controlled time. The techniques are used to quantifying the survey of VoIP. The packet loss techniques are proposed to service the VoIP.

Burst packet loss and enhanced packet loss‐based quality model for mobile voice‐over Internet protocol applications

Voice over Internet protocol (VoIP) quality monitoring platforms need to be as simple as possible, especially at low capacity end-points, such as smart phones. The existing simple packet loss-based quality model, with only the input metric of the packet loss parameter, needs to be enhanced to increase the accuracy of quality prediction while ensuring that implementation is simple. Several burstiness parameters for describing burst loss characteristics in VoIP packet streams has been presented. The authors investigate how closely each one is correlated to perceptual quality. Then, from the correlation results, two dominant parameters are chosen that will be used as additional input metrics to increase the accuracy of quality prediction. A packet loss-based and enhanced quality prediction model is also proposed for VoIP services over the Internet. This can replace the speech quality estimation model in ITU G.1070, which aims to implement VoIP quality monitoring platforms at low capacity end-points. Thus, mobile end-points, such as smart phones, can embed the proposed quality prediction model to measure VoIP quality on a real-time basis.

Cross-Layer Scheduling and Resource Allocation for Heterogeneous Traffic in 3G LTE

3G long term evolution (LTE) introduces stringent needs in order to provide different kinds of traffic with Quality of Service (QoS) characteristics. The major problem with this nature of LTE is that it does not have any paradigm scheduling algorithm that will ideally control the assignment of resources which in turn will improve the user satisfaction. This has become an open subject and different scheduling algorithms have been proposed which are quite challenging and complex. To address this issue, in this paper, we investigate how our proposed algorithm improves the user satisfaction for heterogeneous traffic, that is, best-effort traffic such as file transfer protocol (FTP) and real-time traffic such as voice over internet protocol (VoIP). Our proposed algorithm is formulated using the cross-layer technique. The goal of our proposed algorithm is to maximize the expected total user satisfaction (total-utility) under different constraints. We compared our proposed algorithm with proportional fair (PF), exponential proportional fair (EXP-PF), and U-delay. Using simulations, our proposed algorithm improved the performance of real-time traffic based on throughput, VoIP delay, and VoIP packet loss ratio metrics while PF improved the performance of best-effort traffic based on FTP traffic received, FTP packet loss ratio, and FTP throughput metrics.

VoIP receiver-based adaptive playout scheduling and packet loss concealment technique

This paper proposes a high performance playout scheduling and packet loss concealment algorithm at the receiver for enhancing Voice over Internet Protocol (VoIP) speech quality. In the proposed method, arriving packets are classified by an adaptive thresholding approach based on the analysis of multiple features of short signal segments. The excellent classification results are used in the playout scheduling and packet loss concealment. In adaptive playout, the buffering time is minimized by way of playing out normally or compressing each packet according to accurate network jitter estimation. Additionally, linear prediction-based packet loss concealment delivers high voice quality by alleviating the metallic artifacts due to concealing consecutive packet loss or recovering lost packets1.

Fourth Stage of Voice Priority Queue for VoIP over WLANs

Voice over Internet Protocol (VoIP) is growing rapidly during this decade. VoIP is seen as a short-term and long-term transmission for voice and audio traffic and is moving on Wireless Local Area Networks (WLANs) based on IEEE 802.11 standards. Currently, packet scheduling algorithms like Weighted Fair Queuing (WFQ), was mainly designed to provide the bandwidth reservation. The Strict Priority (SP) is low-cost to maintain the delay sensitive voice traffic. Also, a number of research scheduling solutions have been proposed like General processor sharing (GPS), Deficit Round Robin (DRR), Contention-Aware Temporally fair Scheduling (CATS). Unfortunately, the current scheduling won’t be able to handle the VoIP packets properly and they have drawbacks over real-time applications. The objective of this research is to propose a Fourth Stage of Voice Priority Queue (VPQ) packet scheduling and algorithm to ensure more throughput, fairness and efficient packet scheduling for VoIP performance of queues and traffics. A new scheduler flexible which is capable of satisfying the VoIP traffic flows. Experimental topologies on NS-2 network simulator were analyzed for voice traffic.

The VPQ Scheduler In Access Point For VoIP Over WLAN

The Voice over Internet Protocol (VoIP) application has observed the fastest growth in the world of telecommunication. VoIP is seen as a short-term and long-trem transmission for voice and audio traffic. Meanwhile, VoIP is moving on Wireless Local Area Networks (WLANs) based on IEEE 802.11 standards. Currently, there are many packet scheduling algorithms for real-time transmission over network. Unfortunately, the current scheduling will not be able to handle the VoIP packets with the proper manner and they have some drawbacks over real-time applications. The objective of this research is to propose a new Voice Priority Queue (VPQ) packet scheduling and algorithm to ensure more throughput, fairness and efficient packet scheduling for VoIP performance of queues and traffics. A new scheduler flexible which is capable of satisfying the VoIP traffic flows. Experimental topologies on NS-2 network simulator wereanalyzed for voice traffic. Preliminary results show that this can achieve maximum and more accurate VoIP quality throughput and fairness index in access point for VoIP over WLANs. We verified and validated VPQ an extensive experimental simulation study under various traffic flows over WLANs.

Mobility of Mobile Station in Voice Priority Queue Scheduler for VoIPWLANs

Voice over Internet Protocol (VoIP) is growing rapidly during this decade. VoIP is seen as a short-term and long-trem transmission for voice and audio traffic. Meanwhile, VoIP is moving on Wireless Local Area Networks (WLANs) based on IEEE 802.11 standards. Currently, many packet scheduling algorithms like Weighted Fair Queuing (WFQ), was mainly designed to provide the bandwidth reservation. The Strict Priority (SP) is low-cost to maintain the delay sensitive voice traffic. Also, a number of research scheduling solutions have been proposed like General processor sharing (GPS), Deficit Round Robin (DRR), Contention-Aware Temporally fair Scheduling (CATS). Unfortunately, the current scheduling will not be able to handle the VoIP packets with the proper manner and they have some drawbacks over real-time applications. The objective of this research is to propose a new Voice Priority Queue (VPQ) packet scheduling and algorithm to ensure more throughput, fairness and efficient packet scheduling for VoIP performance of queues and traffics. A new scheduler flexible which is capable of satisfying the VoIP traffic flows. Experimental topologies on NS-2 network simulator were analyzed for voice traffic. Preliminary results show that this can achieve maximum and more accurate VoIP quality throughput and fairness index in mobility of mobile station. We verified and validated VPQ an extensive experimental simulation study under various traffic flows over WLANs.

Accurate modeling of VoIP traffic QoS parameters in current and future networks with multifractal and Markov models

In this paper, we analyze the jitter and packet loss behavior of Voice over Internet Protocol (VoIP) traffic by means of network measurements and simulations results. As result of these analyses, we provide a detailed characterization and accurate modeling of these Quality of Service (QoS) parameters. Our studies have revealed that VoIP jitter can be modeled by self-similar processes with short range dependence (SRD) or long range dependence (LRD). The discovery of LRD (a kind of asymptotic fractal scaling) in VoIP jitter was followed by a further work that shows the evidence of multifractal behavior. The implication of these behaviors for VoIP and other interactive multimedia services is that receiver de-jitter buffer may not be large enough to mask the jitter with LRD and multifractal characteristics. On the other hand, we use a description of VoIP packet loss based on microscopic and macroscopic packet loss behaviors, where these behaviors can be modeled by 2-state and 4-state Markov chains, respectively. Based on the above mentioned points, we present a methodology for simulating packet loss. Besides, we found relationships between Hurst parameter (H) with the packet loss rate (PLR); these relationships are based on voice traffic measurements and can be modeled by means of a power-law function, characterized by three fitted parameters. The proposed models can be used to: (i) design a de-jitter buffer, (ii) implement a synthetic generator of VoIP jitter data traces, where the synthetic jitter data traces can be used as test vectors to carry out the performance evaluation of a de-jitter buffer of VoIP system, and (iii) design effective schemes for packet loss recovery.

QoS and Channel-Aware Packet Bundling for Capacity Improvement in Cellular Networks

We study the problem of multiple-packet bundling to improve spectral efficiency in cellular networks. The packet size of real-time data, such as voice over Internet Protocol (VoIP), is often very small. However, the common use of time-division multiplexing (TDM) limits the number of VoIP users supported because a packet has to wait until it receives a time slot, and if only one small VoIP packet is placed in a time slot, capacity is wasted. Packet bundling can alleviate such a problem by sharing a time slot among multiple users. A recent revision of cdma2000 1xEV-DO has introduced the concept of the multiuser packet (MUP) in the downlink to overcome limitations on the number of time slots. However, the efficacy of packet bundling is not well understood, particularly in the presence of time-varying channels. We propose a novel quality-of-service (QoS) and channel-aware packet bundling algorithm that takes advantage of adaptive modulation and coding. We show that optimal algorithms are nondeterministic polynomial time (NP)-complete, recommend heuristic approaches, and use analytical performance modeling to show the gains in capacity that can be achieved from our packet bundling algorithms. We show that channel utilization can be significantly increased by slightly delaying some real-time packets within their QoS requirements while bundling those packets with like channel conditions. We validate this paper through extensive OPNET simulations with a complete evolution-data optimized (EV-DO) implementation.

Performance Evaluation of Opportunistic Vertical Handover Considering On–Off Characteristics of VoIP Traffic

Vertical handover is one of the key technologies to support seamless connectivity across multiple wireless communication systems and guarantee quality of service (QoS) for the applications therein. This paper proposes an opportunistic vertical handover scheme for a voice over Internet Protocol (VoIP) connection that makes use of the on-off characteristics of voice traffic by aligning the mutual silence period of the two parties engaged in conversation with the service disruption time that occurs during the vertical handover procedure. From the six-state Brady model, we derive a simplified two-state Markov model in which the characteristics of both the talk-spurt period and the mutual silence period are obtained. We then analyze the performance of the proposed scheme with respect to the handover execution time and the packet loss time. The analysis and simulation results show that the proposed scheme significantly decreases the VoIP packet loss time, whereas the total time required for the vertical handover procedure increases but remains within a tolerable time limit.