VoIP Capacity Research Articles

Performance Evaluation Voice over WLAN Capacity in Three Standards IEEE 802.11

In the modern landscape of telecommunication, wireless mobile connectivity and migrating voice telephone services to IP technology are two of the most prominent developments that are transforming the landscape. These two concepts can be utilized together using networks which carry voice services over wireless LAN (VoWLAN). However, due to the unique characteristics of each, certain issues inevitably arise and must be handled to allow for suc-cessful deployment. In this paper OPNET 14.5 Modeler was used to carry out extensive simulation scenarios to examine the single cell WLAN VoIP ca-pacity in regards to three IEEE 802.11 WLAN standards which are (a, b, and g) as well as applying different packet sizes, data rates and codec types for analysis.

VoIP Performance Evaluation and Capacity Estimation Using different QoS Mechanisms

Data networks and new mobile networks (4G and 5G) deploy packet switching in their core networks. Due to this architecture VoIP protocols has a wide deployment in these environments. VoIP capacity and quality must be considered during VoIP protocols implementations. The aim of QoS mechanisms is to satisfy voice traffic requirements; this is deployed by using many tools like congestion management utilities, congestion avoidance utilities, and link fragmentation tools. Each tool has an impact on voice performance. Queuing is very important mechanisms in the traffic management system. Certain routers in data networks must deploy some QoS tools that control how different packets are temporally buffered until transmission on the interface. This paper studies the effect of different QoS tools on VoIP application performance and capacity via OPNET simulation. Also, the maximum VoIP capacity which gives accepted quality will be investigated.

Analyzing Traffic Models Using IP Multimedia Server–Client Systems for Consumer Wireless Multimedia System Devices

This paper analyzes the traffic models of Markov-modulated Poisson-process (MMPP) and Markov-Channel (MC) using IP multimedia server–client systems over Campus WiFi networks. The purpose of MMPP is to discover the optimal-delay divergence path to detect and recover the link-failure to mitigate the packet dropping ratio. The purpose of MC is to create a cross-model layer to probe the lossy and VoIP capacity to satisfy the constraints of radio resources to achieve better results in terms of throughput rate and packet jitter for some extent. To show the major significances of MMPP and MC, this paper designs and develops the existing traffic models, like 1xEVDO, DTMC and DOrA in multimedia application systems. The result proves that the proposed model can achieve better communication performance in compared with the existing models.

Secondary VoIP Capacity in Opportunistic Spectrum Access Networks with Friendly Scheduling

In conventional cognitive radio, the primary network usually remains unchanged. In some cases, however, the primary network operator may wish to accommodate secondary user access. In this paper, we assess the secondary user VoIP capacity when primary basestation scheduling is designed to be secondary network friendly. Friendliness is measured by the number of connections that can be supported subject to typical quality of service constraints in the presence of delay tolerant primary traffic. An offline scheduler is first derived that maximizes friendliness using an integer linear program formulation. We show that this schedule can be found using a minimum cost flow graph construction in time complexity that is polynomial in the number of time slots. Two online scheduling algorithms are then compared that achieve various levels of friendliness. The first algorithm operates by having the primary network temporally shape its residual capacity subject to satisfying its own packet deadline constraints. The second algorithm assumes virtual secondary calls and applies scheduling to both primary traffic and virtual secondary traffic. Results are presented for a variety of parameters that show the degree to which friendly scheduling can improve secondary user VoIP capacity compared to non-friendly primary scheduling.

Boosting VoIP Capacity via Service Differentiation in IEEE 802.11e EDCA Networks

This paper considers the performance problem of VoIP over 802.11e WLANs caused by the unfairness between uplink and downlink as well as the inefficient EDCA. A novel medium access control scheme named BEDCA (Balanced EDCA) is presented, which provides service differentiation between the access point (AP) and the mobile stations (STAs) to enhance VoIP capacity. In BEDCA, the expression of AP's contention window is obtained which is a relative constant value independent of the participating STAs. The minimum contention window of the STAs is traffic-aware based on the proposed algorithm. The performance improvement of BEDCA is verified through intensive simulations and the results show the capacity improvement of 82.1% compared to EDCA.

On the performance of secure user-centric VoIP communication

Motivated by the increased Wi-Fi coverage in metropolitan areas and the emergence of user-centric wireless access schemes, we focus on the provision of secure, user-centric voice services and explore their potential performance-wise, by designing a VoIP communications scheme tailored to open-access wireless environments, but also with wider applicability, and experimenting with it to estimate its upper bounds on VoIP capacity, under constraints posed by user-centrism; operation at low-cost and on user-controlled equipment, minimal dependence on centralized entities, and tackling specific security challenges. We identify quality degradation factors and quantify their importance by simple analysis and experimentation, showing that typical user Wi-Fi equipment can sustain a satisfactory number of concurrent secure VoIP sessions with acceptable Quality of Experience and, at the same time, protection from malicious user activity can be offered to access providers, while a level of roaming privacy can be guaranteed.

LTE 무선 네트워크에서 Voice over IP 용량 분석

4세대 이동통신 시스템인 LTE에서는 음성 서비스를 제공하기 위한 코어 망이 존재하지 않으며, IP 기반의 패킷 코어망으로 통합되었다. 따라서, 음성 서비스를 제공하기 위해서는 VoIP 방식이 도입되어야 하지만, LTE 초기에는 서비스 커버리지의 불연속성 등으로 3세대 네트워크의 음성 서비스 코어망을 통하여 일시적으로 음성 서비스를 제공하게 된다. 그러나, 향후 LTE 네트워크가 본격적으로 확산될 경우 고품질 음성 서비스를 위하여 필연적으로 VoIP 방식으로의 전환이 불가피하다. 본 논문에서는, 향후 본격적으로 상용화가 예상되는 VoLTE 서비스의 용량을 분석한다. 분석 결과에 따르면, LTE에서는 채널 대역폭 증가 비율 대비 VoLTE 용량 증가 비율이 높음을 확인할 수 있었다. 또한, VoLTE 용량 감소 없이 제어 채널용 자원을 늘릴 수 있는 방안이 존재함을 확인하였다. The 4th generation mobile communication system, LTE, does not support an additional core network to provide voice service, and it is merged into a packet network based on all IP. Although Voice service over LTE can be supported by VoIP, it will be provided by the existing 3G networks because of the discontinuity of LTE coverage. However, it is inevitable to adopt VoIP over LTE to provide high quality voice service. In this paper, we investigate the capacity of VoIP over LTE. Our results indicate that spectral efficiency can be significantly improved as channel bandwidth increases in terms of VoLTE capacity. In addition, we can achieve higher VoLTE capacity without decreasing control channel capacity.

Evaluation of VoIP Capacity for IEEE802.11b WiFi Environment under Voice Coding Methods

In this paper we simulate the capacity of VOIP calls through WiFi network by computer simulations using OPNET modeler. The results show that sudden quality degradations occur on all VoIP calls when the number of call of an AP(Access Point) increases beyond a specific value. The reason of the quality degradation was turned out to be the queueing delay at the down link of AP. Under the IEEE 802.11b environments, the maximum number of VoIP calls of an AP maintaining the required voice quality (MOS > 2.5), was evaluated as 5, 12, and 27 when we use G.711, G.729a, and G.729a VAD codec, respectively.

Improvement in Performance of the VoIP over WLAN

Wireless VoIP, typically over 802.11 WLAN, is becoming increasingly popular, but even further elevates the challenge of delay and loss reduction. Degradation of speech quality caused by packet delay and loss of voice traffic is still one of the critical technical barriers of the VoIP system. Furthermore, apart from these limitations WLAN will need to support a large number of concurrent VoIP communication since VoIP is spreading rapidly. These motivations led me to do the analysis of VoIP capacity in IEEE 802.11e WLAN and to investigate the methodology to enhance this capacity while maintaining an overall good quality of service.

On Maximizing VoIP Capacity and Energy Conservation in Multi-Rate WLANs

This letter highlights a key problem that arises when frame aggregation, a popular method for boosting VoIP capacity, is used in Wireless Local Area Networks (WLANs) that employ asynchronous power save mode (PSM). Specifically, it shows how the PSM proposed by the Wi-Fi Alliance renders frame aggregation ineffective. It then proposes a novel opportunistic scheduler that restores the benefits of frame aggregation whilst ensuring stations have minimal energy expenditure.

Achieving maximal VoIP calls in 802.11 wireless networks

IEEE 802.11b WLAN deployments have mushroomed in recent years. However, the original 802.11 MAC layer can not meet the requirements of modern wireless multimedia applications. Accordingly, the IEEE 802.15.3 standard has been proposed to increase the data rate in wireless multimedia applications whilst simultaneously reducing the cost and power consumption of the portable devices within the network. In this study, the 802.15.3 MAC protocol is implemented over an 802.11 WLAN. In implementing the 802.15.3 superframes, the timing functions are accomplished using high-resolution timers such that the traditional CSMA/CA protocol can be retained. It is shown that the emulated 802.15.3 MAC layer is capable of supporting 12 full-duplex voice calls or 24 half-duplex voice calls, which is consistent with the theoretical VoIP capacity of 802.11b WLANs.

Performance Prediction Model for Interference-limited Dynamic OFDMA Systems

This paper presents a performance model for dynamic resource allocation in interference-limited OFDMA-based cellular networks. Specifically, we derive an analytical framework by which it is possible to estimate the VoIP capacity of the cell per downlink frame, purely based on the average SINR per terminal. Hence, this framework can be used for admission control. The major difficulty to overcome is to predict the impact of dynamic resource allocations on the system performance in the presence of multiple interfering cells. We show that dynamic resource allocations transform the distribution of subcarrier SINR and derive an approximate, closed-form representation of these transformed distributions. Based on these derivations, we obtain rate probability mass functions per terminal which predict system performance up to a gap of about 26 % compared to optimal system performance.

Measurement and Analysis of the VoIP Capacity in IEEE 802.11 WLAN

We measured the capacity for VoIP traffic in an IEEE 802.11b wireless testbed and compared it with the theoretical capacity and our simulation results. We identified factors that have been commonly overlooked in past studies but affect experiments and simulations. We found that in many papers, the capacity for VoIP traffic has been measured via simulations or experiments without considering these factors, showing different capacity in each paper. After these corrections, simulations and experiments yielded a capacity estimate of 15 calls for 64 kb/s CBR VoIP traffic with 20 ms packetization interval and 38 calls for VBR VoIP traffic with a 0.39 activity ratio. Furthermore, we measured the capacity for VoIP traffic using each access category introduced in the 802.11e standard and the effect of the TCP traffic on VoIP traffic. We found that while the 802.11e standard can protect the QoS of VoIP against TCP traffic, it does not improve the capacity due to the significant retransmissions during TXOP.

Performance of VoIP over Multiple Co-Located IEEE 802.11 Wireless LANs

IEEE 802.11 WLAN has high data rates (e.g., 11 Mbps for 802.11b and 54 Mbps for 802.11g), while voice streams of VoIP typically have low-data-rate requirements (e.g., 29.2 Kbps). One may, therefore, expect WLAN to be able to support a large number of VoIP sessions (e.g., 200 and 900 sessions in 802.11b and 802.11g, respectively). Prior work by one of the authors, however, indicated that 802.11 is extremely inefficient for VoIP transport. Only 12 and 60 VoIP sessions can be supported in an 802.11b and an 802.11g WLAN, respectively. This paper shows that the bad news does not stop there. When there are multiple WLANs in the vicinity of each other-a common situation these days-the already low VoIP capacity can be further eroded in a significant manner. For example, in a 5 times 5, 25-cell multi-WLAN network, the VoIP capacities for 802.11b and 802.11g are only 1.63 and 10.34 sessions per AP, respectively. This paper investigates several solutions to improve the VoIP capacity. Based on a conflict graph model, we propose a clique-analytical call admission scheme, which increases the VoIP capacity by 52 percent from 1.63 to 2.48 sessions per AP in 802.11b. For 11g, the call admission scheme can also increase the capacity by 37 percent from 10.34 to 14.14 sessions per AP. If all the three orthogonal frequency channels available in 11b and 11g are used to reduce interferences among adjacent WLANs, clique-analytical call admission scheme can boost the capacity to 7.39 VoIP sessions per AP in 11b and 44.91 sessions per AP in 11g. Last but not least, this paper expounds for the first time the use of coarse-grained time-division multiple access (CoTDMA) in conjunction with the basic 802.11 CSMA to eliminate the performance-degrading exposed-node and hidden-node problems in 802.11. A two-layer coloring problem (which is distinct from the classical graph coloring problem) is formulated to assign coarse time slots and frequency channels to VoIP sessions, taking into account the intricacies of the carrier-sensing operation of 802.11. We find that CoTDMA can further increase the VoIP capacity in the multi-WLAN scenario by an additional 35 percent, so that 10 and 58 sessions per AP can be supported in 802.11b and 802.11g, respectively.

Improved VoIP capacity in mobile WiMAX systems using persistent resource allocation

Efficient support of voice traffic has always been one of the key metrics for evaluating and selecting radio access technologies. Even for next-generation broadband wireless technologies primarily focused on the mobile Internet, special handling of VoIP traffic in the physical and MAC layers is required to maximize voice capacity. While Mobile WiMAX Release 1.0 and 802.16e have all the key features necessary to support mobile VoIP traffic, special attention is given in Mobile WiMAX Release 1.5 and 802.16REV2 to further optimizing VoIP capacity through reduction in the MAC layer overhead associated with signaling messages. This article focuses on features and solutions used in Mobile WiMAX and the 802.16 standard to support voice traffic and the expected performance in Release 1.0/802.16e-2005, as well as gains from optimization concepts such as persistent allocation added in Release 1.5/802.16REV2. In each case, MAP overhead reduction and the projected improvements in VoIP capacity are presented using typical industry accepted models and assumptions. The results show about 15 percent increase in bidirectional VoIP capacity.

An Analysis of Joint Aggregation, Bursting, Routing, and Rate Adaptation for Increasing VoIP Capacity in Multi-Hop 802.11 Networks

Due to the low cost and ease of deployment, singlehop and multi-hop 802.11 networks have become attractive solutions for providing last-mile broadband (wireless) access in urban environments. However, a critical issue in using such networks to support applications such as Voice over IP is the widely known overheads in the Medium Access Control (MAC) layer and Physical (PHY) layer for each transmission. The effect of these overheads can be more severe in multi-hop deployments, and can limit the number of VoIP calls per Gateway Access Point (GAP) in a multi-hop system to be no greater than that of a single-hop single-GAP system. In this paper, we build upon our previous work that showed that a multi-hop network, with one GAP using a single channel, can in fact support more users than a single-hop network. To do so functions such as aggregation have to be intelligently used at relays in the network. We build on this prior work giving a theoretical framework that considers the joint tradeoff of aggregation, bursting, and PHY rate adaptation given a set of admission and routing decisions. We compare call-capacity estimates from the theory to those derived from simulations, and illustrate through examples the advantages of considering routing/ admission decisions jointly with bursting and aggregation.

VoIP cross-layer load control for hybrid satellite-WiMAX networks

Hybrid architectures including satellites are of undoubted interest for wireless networks deployment in rural and remote areas. We focus on VoIP capacity service optimization over a hybrid network and we compare performance under temperate and tropical propagation conditions. We have chosen a (GEO) satellite and wireless network combination, based on DVB-S2/RCS and WiMAX standards. This combination has already proven to be a relatively simple hybrid combination with enormous potential at both the service and architecture levels. In order to tackle the design of the hybrid network on a difficult propagation/topographical environment, we propose a cross-layer design across both layers and networks. Our proposal is based on commercially available VoIP codecs, and the cross-layer design does not require breaking the protocol stacks of both networks. We propose a centralized scenario where VoIP load control is performed through a physical-transport-application cross- layer information flow (inherently cross-layer across both networks). In particular, VoIP rate is centrally controlled in order to match the satellite link capacity drops due to physical layer adaptation to channel conditions. Realistic modeling is considered for both the hybrid architecture and the channels. We present numerical results of the proposed design showing that our cross-layer load control technique is able to maintain the active WiMAX number of connections for a wide range of rain attenuation in both temperate and tropical areas.

VoIP support using group resource allocation based on the UMB system

It is now clear that the latest evolution of the wireless communication system, for example, the ultra mobile broadband (UMBtrade) system developed in 3GPP2, has significantly increased the packet data efficiency of the system. However, as voice is still a major part of the revenue stream for many operators, voice efficiency is still of prime concern in the industry. A detailed end-to-end scheme for supporting VoIP in the air interface, designated for the UMB system, is described. Full duplex simulations were performed to evaluate the proposed scheme. The results show that the UMB system has significantly improved upon the VoIP capacity. A 5-MHz UMB system can support 320 VoIP users while simultaneously supporting a forward link data throughput of over 2.26 Mbps. Although more of the delay budget was expanded on the reverse link, the system is still reverse link limited.

WLAN VoIP capacity allocation using an adaptive voice packetization server

WLAN VoIP capacity is known to be very low due to the effects of overheads at various protocol layers. An IEEE 802.11b access point (AP) for example, can support only about 10 G.711 voice connections using a 20 ms packetization interval without advanced header compression. This poor performance can be substantially improved if the connection latency margin and packet loss performance are known in advance and are used in the selection of RTP packetization parameters. In this paper we take this process one step further and investigate the use of an adaptive voice packetization server (AVP-RTS) which splits the RTP VoIP connection into two separate call legs. In this way each call leg can use different packetization parameters, thus assigning the capacity gain asymmetrically across the connection. Algorithms are proposed and compared for performing this assignment. We show that by using the AVP-RTS server we can significantly improve the multi-AP VoIP capacity for certain typical WLAN situations.

Boosting VoIP Capacity of Wireless Mesh Networks through Lazy Frame Aggregation

A novel method of voice frame aggregation for wireless mesh networks is presented. In the method, the degree of aggregation is automatically regulated by the congestion level on the wireless link. On the IEEE 802.11-based mesh network, it is shown to yield approximately twice the call capacity, while incurring no additional delay for frame aggregation.