Voice Over IP Sessions Research Articles

Energy/Quality-of-Experience TradeOff of Power Saving Modes for Voice Over IP Services

Deploying a power saving mode (PSM) is considered an effective solution for saving energy; however, a key challenge is how to simultaneously save energy and support a required level of quality of experience (QoE) using PSM for voice over IP (VoIP) services. This paper aims at developing a novel Markovian-based evaluation framework that assesses the energy consumption and QoE, achieved by a fixed sleeping time-based PSM. Based on this framework, a novel QoE-aware PSM is developed, which exploits the ON-OFF characteristics of a VoIP session to further improve the energy saving, while accounting for the effect of bursty losses on the QoE. The QoE and energy consumption achieved by the proposed PSM are evaluated via solving a system of linear equations. This facilitates the analytical formulation of the QoE/energy tradeoff that is needed for the successful integration of this QoE-aware PSM within the traffic offloading mechanisms. Numerical results validate the accuracy of the performance evaluation framework and verify the efficacy of the proposed QoE-aware PSM in saving energy while assuring a desired level of QoE. The Pareto-optimal operational regime show that the proposed QoE-aware PSM significantly reduces power consumption at a desired level of QoE compared with its counterparts.

Identification of Peer-to-Peer VoIP Sessions Using Entropy and Codec Properties

Voice over Internet Protocol (VoIP) applications based on peer-to-peer (P2P) communications have been experiencing considerable growth in terms of number of users. To overcome filtering policies or protect the privacy of their users, most of these applications implement mechanisms such as protocol obfuscation or payload encryption that avoid the inspection of their traffic, making it difficult to identify its nature. The incapacity to determine the application that is responsible for a certain flow raises challenges for the effective management of the network. In this paper, a new method for the identification of VoIP sessions is presented. The proposed mechanism classifies the flows, in real-time, based on the speech codec used in the session. To make the classification lightweight, the behavioral signatures for each analyzed codec were created using only the lengths of the packets. Unlike most previous approaches, the classifier does not use the lengths of the packets individually. Instead, it explores their level of heterogeneity in real time, using entropy to emphasize such feature. The results of the performance evaluation show that the proposed method is able to identify VoIP sessions accurately and simultaneously recognize the used speech codec.

Vertical Handover Management for VoIP Session over Broadband Wireless Networks

Today, wireless LAN (IEEE802.11g/n) has been the dominant wireless network that can provide high data rates in a limited coverage area. While emerging mobile WiMAX (IEEE802.16e) can provide a wireless broadband access to mobile users in a wide coverage area. These two different technolgies will co-exist while complementing each other in some regions, hence, a mobile node (MN) with dual interfaces traverses and executes many handovers (HOs) between 802.11g and 802.16e. Meanwhile, there is a huge demand for Voice over IP (VoIP) service over wireless networks. However, VoIP is a delay and loss sensitive application, hence, VoIP session is more likely to be deteriorated during HO between 802.11g and 802.16e. In order to maintain quality of VoIP session during HO, we proposed Vertical Handover Management (VHM) for VoIP session that focuses on HO initiation and decision strategy according to the wireless link condition and congestion state of wireless networks. The VHM exploits request to send (RTS) retries and round trip time (RTT) between an MN and an access point in an 802.11g interface as well as a carrier to interference noise ration (CINR) level and MN’s queue length of an 802.16e interface as HO triggers. We then conducted simulation experiments to evaluate the effectiveness of our proposed VHM using QualNet 4.5. Our simulation results show that our proposed VHM can preserve the quality of VoIP session during such HOs.

Proactive QoS Enhancement Technique for Efficient VoIP Performance over Wireless LAN and Cognitive Radio Network

VoIP over Wireless LANs is greatly affected bypath-loss, RF interference and other sources of signalattenuation in addition to network congestion. The primaryfactors involved in effective real-time communication,namely delay and loss, must be within certain controlledlimits in such a scenario. Wireless LAN Access Points must,therefore, act in tune with the codecs to ensure high qualityof the ongoing VoIP sessions. Further deployment of VoIPin opportunistic communication medium like cognitiveradio network requires thorough analysis of the factorsinvolved in design and implementation of such networksalong with modifications in VoIP parameters. In this paper,the various codec parameters are analyzed with respect towireless LAN access points. Thereafter, an optimizationalgorithm is developed based on proactive strategy to keeploss and latency within tolerable limits. The proposedalgorithm is hence implemented in the SIP enabled test-bedto provide optimum Voice over IP (VoIP) performance. Theproactive strategy is further applied to study VoIPperformance over cognitive radio networks. A simulationmodel is developed for VoIP transmission in cognitive radionetworks with two scenarios where the primary and thesecondary user involve in VoIP transmission respectively.For each scenario, extensive analysis of both VoIP and CRNparameters is performed and an algorithm is proposedbased on codecs and active queues to ensure highthroughput of VoIP traffic with enhanced QoS. Resultsverify significant performance improvement in VoIP callquality under both wireless LAN and cognitive radionetwork environment. Simulation and measured resultshave also been included.

Sync & Sense: VoIP Measurement Methodology for Assessing One-Way Delay Without Clock Synchronization

It is well known that measuring the one-way delay of voice-over-IP (VoIP) packets is a daunting task. The challenge lies in the fact that the Internet implicitly enforces the end-to-end principle. This means that the endpoints are expected to operate independently. Without a synchronized timing, it is difficult for an endpoint to measure the one-way delay. This paper presents the refined version of the novel VoIP measurement methodology called Sync & Sense of Periodic Stream that can overcome such a challenge. Sync & Sense is unique in that it can virtually synchronize the transmission and reception timing of the VoIP session without requiring a synchronized clock. Our performance study shows that Sync & Sense can accurately measure the one-way network delay of the VoIP packets (without propagation delay). While time skew is very common in any system involving a clock, we also discuss this issue and how Sync & Sense can deal with the time skew without requiring a synchronized clock.

Application-aware scheduling for VoIP in Wireless Mesh Networks

Wireless Mesh Networks (WMNs) are seen as a means to provide last mile connections in Next Generation Networks (NGNs). Because of their auto-configuration capabilities and the low deployment cost WMNs are considered to be an efficient solution for the support of multiple voice, video and data services in NGNs. This paper looks at the optimal provision of resources in WMNs for Voice over IP (VoIP) traffic, which has strict performance requirements in terms of delay, jitter and packet loss. In WMNs, because of the challenges introduced by wireless multi-hop transmissions and limited resources, providing performance quality for VoIP comparable to the voice quality in the traditional circuit-switched networks is a major challenge. This paper analyses different scheduling mechanisms for TDMA-based access control in mesh networks as specified in the IEEE 802.16-2004 WiMAX standard. The performance of the VoIP applications when different scheduling mechanisms are deployed is analysed on a variety of topologies using ns-2 simulation and mathematical analysis. The paper concludes that on-demand scheduling of VoIP traffic - typically deployed in 802.11-based WMNs - is not able to provide the required VoIP quality in realistic mesh WiMAX network scenarios and is therefore not optimal from a network operator's point of view. Instead, it is shown, that continuous scheduling is much better suited to serve VoIP traffic. The paper then proposes a new VoIP-aware resource coordination scheme and shows, through simulation, that the new scheme is scalable and provides good quality for VoIP service in a wide range of network scenarios. The results shown in the paper prove that the new scheme is resilient to increasing hop count, increasing number of simultaneous VoIP sessions and the background traffic load in the network. Compared to other resource coordination schemes the VoIP-aware scheduler significantly increases the number of supported calls.

Research on the algorithm for identifying VoIP session

In this letter, a novel algorithm is proposed for identification of one popular type of VoIP (Voice over IP) application, MSN VoIP, based on both behavior pattern and protocol characteristics. In addition, the experiment for validation is implemented. Since there exists no method identifying MSN VoIP session and furthermore the IP addresses, domain names, ports and even the protocols MSNP works over, are all alterable during the application, which makes it difficult to identify MSN VoIP session, this work will be helpful to advance the research on VoIP identification.

Solutions to Performance Problems in VoIP Over a 802.11 Wireless LAN

Voice over Internet Protocol (VoIP) over a wireless local area network (WLAN) is poised to become an important Internet application. However, two major technical problems that stand in the way are: 1) low VoIP capacity in WLAN and 2) unacceptable VoIP performance in the presence of coexisting traffic from other applications. With each VoIP stream typically requiring less than 10 kb/s, an 802.11b WLAN operated at 11 Mb/s could in principle support more than 500 VoIP sessions. In actuality, no more than a few sessions can be supported due to various protocol overheads (for GSM 6.10, it is about 12). This paper proposes and investigates a scheme that can improve the VoIP capacity by close to 100% without changing the standard 802.11 CSMA/CA protocol. In addition, we show that VoIP delay and loss performance in WLAN can be compromised severely in the presence of coexisting transmission-control protocol (TCP) traffic, even when the number of VoIP sessions is limited to half its potential capacity. A touted advantage of VoIP over traditional telephony is that it enables the creation of novel applications that integrate voice with data. The inability of VoIP and TCP traffic to coexist harmoniously over the WLAN poses a severe challenge to this vision. Fortunately, the problem can be largely solved by simple solutions that require only changes to the medium-access control (MAC) protocol at the access point. Specifically, in our proposed solutions, the MAC protocol at the wireless end stations does not need to be modified, making the solutions more readily deployable over the existing network infrastructure.