VoIP Performance Research Articles

Enhanced Quality of Service in Worldwide Interoperability for Microwave Access Networks

The WiMAX system effectively supports wide variety of broadband wireless access technologies, which including high speed internet and multimedia access with high Quality of service (QoS) requirements. Real time services such as VoIP are becoming popular and are major revenue earners for network service providers. However, there are still many challenges that need to be addressed to provide a steady and good quality voice connection over the best effort WiMAX network .To support flexibility, efficiency and various requirements of QoS over a range of different applications and environments several provisioning and mechanisms are provided .This research paper investigates the performance of VoIP traffic over WiMAX networks, the impact of various voice codec schemes and statistical distribution for VoIP over WiMAX has been investigated in detail. Through various simulation experiments under realistic networking scenarios, this study provides an insight into the VoIP performance in the WIMAX networks. The simulations results indicate that better choice of voice codec's and statistical distribution have significant impact on VoIP performance in the WiMAX networks and Performance of selected parameters will be done using the network simulator OPNET

Guest Editorial

Guest Editorial

NCTU‐VT: a freeware for wireless VoIP performance measurement

AbstractVoice over IP (VoIP) is a promising low‐cost voice communication over the wireless IP network. To provide satisfactory VoIP services, the Quality of Service (QoS) of the wireless network should be guaranteed. This paper proposes a VoIP performance measurement freeware called NCTU VoIP Testing Tool (NCTU‐VT). We compare NCTU‐VT with two commercial tools SmartVoIPQoS and IxChariot in terms of packet loss, latency, and Mean Opinion Score (MOS) of the VoIP sessions in Wi‐Fi network. Our study indicates that these three tools can accurately measure VoIP performance in Wi‐Fi environment. Copyright © 2010 John Wiley & Sons, Ltd.

Enhanced VoIP Support in OFDMA-based Packet Radio Networks

As the first OFDMA-based cellular packet radio network, Evolved UTRAN has already been well studied in 3GPP and semi-persistent scheduling was selected as the most relevant packet scheduling scheme for VoIP traffic. With further evolution of Evolved UTRAN to meet the ambitious goal set by IMT-A, the performance of VoIP needs to be enhanced. In this article, we establish a semi-persistent scheduling centered framework for efficient VoIP support in OFDMA-based packet radio network and evaluate different enhancement methods, such as adaptive transmission bandwidth in persistent resource allocation, dynamic packet bundling, and frequency diversity transmission that can be added into this framework. In addition, we investigate practical constraints, e.g. wideband CQI reporting on real network performance as well as possible corrective measures. The whole system design is verified through large-scale network level simulations in which all the details of various VoIP-specific radio resource management algorithms and enhancement mechanisms as well as interactions between them are explicitly modeled. Simulation results indicate that VoIP in Evolved UTRAN can be well supported using this framework and further improved through different enhancement methods. Compared with fixed transmission bandwidth approach, adaptive transmission bandwidth approach can provide additional capacity gain of 11 and 8% respectively in 3GPP Macro Case 1 and Case 3 deployment scenario. In a highly-loaded network, switching from adaptive transmission bandwidth to dynamic packet bundling can accommodate 18 and 15% more users in Case 1 and 3 respectively. The performance degradation due to wideband CQI reporting can be partly compensated with frequency diversity transmission. It is finally concluded that the enhanced performance of VoIP is sufficient to meet the ambitious target set by IMT-A in all four ITU deployment scenarios.

Adaptive Localized Active Route Maintenance Mechanism to Improve Performance of VoIP over Ad Hoc Networks

An important characteristic of a mobile ad hoc network routing protocol is connectivity maintenance. The protocol’s ability to maintain routes and react to topology changes due to link failures affects the delay, packet loss, and throughput of real-time applications. However, the time required to recover from a link failure in the current AODV protocol is too long for real-time applications. The objective of this paper is to develop a mechanism to allow fast response to link breaks to enable real-time communications, such as Voice over IP. We propose AODVM-ALARM, an adaptive localized route maintenance mechanism, over an existing ad hoc multipath routing protocol. The mechanism dynamically monitors and maintains on-going communications by utilizing special control messages and packet analysis techniques to enable fast response to link breaks. The mechanism can recognize, respond and recover quickly from link breaks due to topology changes. AODV-ALARM has been tested in an ad hoc experimental testbed, and compared with AODV-UU and AODVM protocols. Results show that AODVM-ALARM significantly improves the response time to link failures and achieves an overall improvement in the network performance in terms of the failover delay, failover loss and throughput when compared with the previous ad hoc routing protocols. AODVM-ALARM which is easily deployable, fault tolerant and have a fast response to topology changes can be used to rapidly establish VoIP communications, playing a significant role in emergency response for disaster recovery when the network infrastructure might be broken.

VOIP Performance Over Broadband Wireless Networks Under Static and Mobile Environments

VOIP Performance Over Broadband Wireless Networks Under Static and Mobile Environments

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.

VoIP Performance over Different Interior Gateway Protocols

Voice over IP technology integrates data and voice networks and offers flexibility by supplying device interoperability using standards-based protocols. Routing is an essential data networking function that provides an efficient real-time data delivery VoIP requires. Best-effort networks leverage Interior Gateway Protocol technologies to determine paths for routing packets between hosts. Architects for best-effort networks provide services by over-provisioning the links and routers so that network congestion does not introduce unwanted levels of latency, jitter, and packet loss. Route re-convergence can be detrimental to VoIP users in the middle of a conversation, as it impacts both latency and jitter. This paper studies how VoIP performance can be affected by different routing behaviors which include Routing Information Protocol, Open Shortest Path First and Enhanced Interior Gateway Routing Protocol. Network modeling and simulation have been carried out with OPNET Modeler to evaluate and compare performances.

IEEE 802.11n MAC Enhancement and Performance Evaluation

The IEEE 802.11-based WiFi wireless technology is one of the most promising technologies to provide ubiquitous networking access. The IEEE 802.11 working group has always strived to improve this wireless technology through creating new amendments to the base 802.11 standard. Recently, IEEE 802.11n amendment was created to enhance 802.11 for higher throughput operation. Not only new Physical Layer enhancements are standardized, but new Medium Access Control Layer mechanism are also defined. In this paper, we examine the network performance enhancement by the proposed 802.11n MAC layer features: aggregation, block acknowledgement, and reverse direction mechanism. We implemented a new 802.11n module in the NS-2 simulation platform. The simulation results demonstrated the effectiveness of 802.11n MAC layer enhancement. VoIP performance is effectively improved with 802.11n MAC enhancement.

IPsec-Based VoIP Performance in WLAN Environments

The third-generation partnership project (3GPP) specifies 3G-WLAN interworking that lets mobile stations (MSs) access 3G networks via WLAN-based packet data gateways (PDGs). The specification mandates that packets delivered between an MS and a PDG should be protected by IPsec. This article studies IPsec-based VoIP performance (in terms of throughput, packet loss rate, latency, and jitter) in the 3G-WLAN integration environment. The study is designed to provide guidelines for selecting appropriate system parameter values for VoIP over WLAN.

AP Selection Algorithm for Real-Time Communications through Mixed WLAN Environments

Recent rapid development of high-speed wireless access technologies has created mixed WLAN (Wireless LAN) environments where QoS capable APs coexist with legacy APs. To provide QoS guarantee in this mixed WLAN environment, this paper proposes a new AP selection algorithm. The proposed algorithm assigns an STA (Station) to an AP in the overall WLAN service area. Simulation results show improvement in the VoIP performance in terms of an eMOS (estimated Mean Opinion Score) value and the FTP throughput compared to conventional algorithms.

Assessment of Internet voice transport with TCP

AbstractWe investigate whether the current best‐effort technology of Internet stands up to the high quality standards of real‐time voice communication. More precisely, we assess VoIP quality within the context of transport protocol support and efficiency. Initially, we focus on TCP and UDP supportive role from the perspective of VoIP performance. Applying our metric for real‐time performance, we reach the outcome that UDP is always not the protocol of choice for VoIP, since it occasionally exhibits inadequate performance. Beyond that, we evaluate a solution‐framework based on TCP protocols, which favour real‐time traffic. Based on our measurements, we assess the efficiency of the associated congestion control and congestion avoidance mechanisms in terms of VoIP performance. We also study the effect of packet size on protocol behaviour and VoIP quality. Furthermore, we investigate VoIP traffic friendliness, as well as potential tradeoffs between protocol performance and fairness. Copyright © 2006 John Wiley & Sons, Ltd.

VoIP Protection and Performance Improvement

A lot of ongoing work is geared towards improving the reliability, performance and QoS characteristics of service provider IP networks. In contrast, we propose novel enterprise-based techniques that exploit the fact that many enterprises are multi/dual-homed. In a form of "service-aware routing", certain (e.g., VoIP) packets are duplicated (e.g., at one edge router) and sent over multiple service providers. After traversing the service provider networks, only the first-to-arrive packets are kept and the later-arriving copies are discarded. In so doing, the result is not only better protection against node and link failures, and packet losses and errors, but also better QoS performance under normal (fault-free) operation. The packet-duplication process can be policy-based and take into account costs, bandwidth, and priority issues, permitting the system to behave like a simple "smart router" that automatically and continually makes use of the best (lowest-delay) service provider. In this paper, we present the main ideas behind the proposal, along with some initial analytical and experimental/simulation results and insights from a Linux-based implementation.