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This paper presents the findings of two experiments that were performed on the Redundancy in Wireless Connection Model (RiWC) using the 802.11b standard. The experiments were simulated using OPNET 11.5 Modeler software. The first was aimed at finding the maximum number of simultaneous Voice over Internet Protocol (VoIP) users the model would suppor...
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Three different queuing mechanisms...
In this paper, we describe architecture for VoIP over WLAN. We discuss the issue of providing good QoS support for VoIP applications when run on WLAN. We practically analyze the performance of VoIP over WLAN by using resource reservation protocol (RSVP). We use SIP as a signaling protocol for VoIP. By means of a real tested implementation, we provi...
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Wireless LAN connects two or more devices using Orthogonal Frequency Division Multiplexing (OFDM) or Direct Sequence Spread Spectrum (DSSS) modulation techniques to establish communication between devices within a limited range. This paper mainly aimed to study wireless local area network WLAN and analyze VoIP over WLAN and explore the evaluation o...
Citations
... The MOS standard considers any of the values between 4 and 5 as a clear phone conversation and anything below 4 as poor quality [24]. The acceptable jitter's value is between 0 ms to 50 ms, and it is unacceptable above 50 ms [25]. The average interpacket arrival time comes from jitter. ...
... VoIP interpacket arrival time should be between 20 or 30 ms [27]. The acceptable data loss value falls between 0% and 1.5% and unacceptable above this [25]. The maximum tolerable latency value falls between 80 ms and 120 ms [28]. ...
Voice over Internet Protocol (VoIP) phone service has several factors that can contribute to its degraded quality. Some of these factors include delay, packet loss, and jitter. This research study focuses on VoIP phone services offered to customers by cable companies that utilize broadband hybrid fiber coaxial (HFC) networks. The value of this research comes from the nature of the HFC broadband networks where they deliver three types of traffic on the same network links. Traffic types include voice, data, and video. Voice is unlike data and video, some delays or packet loss can result in a noticeable degraded impact on a VoIP's phone conversations. This research will analyze and assess VoIP traffic prioritization and its impact on VoIP's quality of service (QoS) based on the concept of differentiated services code point (DSCP) markings. This study examines two call types that include VoIP-based and signaling system 7 (SS7) based networks. This research mimics the production environment. This study provides some DSCP markings configurations that can clearly improve VoIP's conversations quality.
... Figures 10, 1112, and 13 show the delay performance of HP and LP traffics, respectively, in our scheme (marked as " new delay " on the graph) in comparison to IEEE 802.11e (marked as " old delay " on the graph) for network scenarios I (Figures 10 and 12) and II (Figures 11 and 13). Our proposed scheme leads to about a 6% increase in delay for HP traffic, as expected, since the CFP phase is now utilized to serve the LP traffic as well in addition to the HP traffic, thereby leading to a longer wait time for the system to revert back to the CP where HP traffic can start their transmission again [20,21]. Specifically speaking, in our simulation setup (given our data rate of 2 Mbps and CBR packet arrival), we had set the delay bound of HP traffic to a reasonable 0.5 s for both traffic scenarios I and II. ...
With the tremendous boom in the wireless local area network arena, there has been a phenomenal spike in the web traffic which has been triggered by the growing popularity of real-time multimedia applications. Towards this end, the IEEE 802.11e medium access control (MAC) standard specifies a set of quality-of-service (QoS) enhancement features to ensure QoS for these delay sensitive multimedia applications. Most of these features are unfair and inefficient from the perspective of low priority (non-real time) traffic flows as they tend to starve the non-real time flows depriving them of appropriate channel access, hence throughput. To that extent, this article proposes a MAC protocol that ensures fairness in the overall network performance by still providing QoS for real-time traffic without starving the "underdog" or non-real-time flows. The article first presents analytical expressions supported by Matlab simulation results which highlight the performance drawbacks of biased protocols such as 802.11e. It then evaluates the efficiency of the proposed "fair MAC protocol" through extensive simulations conducted on the QualNet simulation platform. The simulation results validate the fairness aspect of the proposed scheme.
... For the VoIP case, we use PDR and delay jitter as performance metric, since these two parameters largely determine the QoS of VoIP sessions. A maximum packet loss of 1.5% and a delay jitter of less than 50 ms is generally considered acceptable for a standard VoIP session [36]. Not surprisingly, the VoIP performance results of all four AODV variants in terms of PDR are similar to our other CBR UDP results shown inFig. ...
Reliable broadband communication is becoming increasingly important during disaster recovery and emergency response operations. In situations where infrastructure-based communication is not available or has been disrupted, an Incident Area Network needs to be dynamically deployed, i.e. a temporary network that provides communication services for efficient crisis management at an incident site. Wireless Mesh Networks (WMNs) are multi-hop wireless networks with self-healing and self-configuring capabilities. These features, combined with the ability to provide wireless broadband connectivity at a comparably low cost, make WMNs a promising technology for incident management communications. This paper specifically focuses on hybrid WMNs, which allow both mobile client devices as well as dedicated infrastructure nodes to form the network and provide routing and forwarding functionality. Hybrid WMNs are the most generic and most flexible type of mesh networks and are ideally suited to meet the requirements of incident area communications. However, current wireless mesh and ad-hoc routing protocols do not perform well in hybrid WMN, and are not able to establish stable and high throughput communication paths. One of the key reasons for this is their inability to exploit the typical high degree of heterogeneity in hybrid WMNs. SafeMesh, the routing protocol presented in this paper, addresses the limitations of current mesh and ad-hoc routing protocols in the context of hybrid WMNs. SafeMesh is based on the well-known AODV routing protocol, and implements a number of modifications and extensions that significantly improve its performance in hybrid WMNs. This is demonstrated via an extensive set of simulation results. We further show the practicality of the protocol through a prototype implementation and provide performance results obtained from a small-scale testbed deployment.
... Our model has been tested for two types of applications (PRINT and EMAIL) in two different sites each comprising of 20 users and found that among a set of other parameters Media Access Delay and Load were highly affected by the number of users per application with and without load balancing. Simulation performed shows the behavior of load balancing on wire-line and wireless network for both applications (print and email) [5][6][7][8][9][10][11][12][13][14][15]. ...
... The IEEE 802.11 [12,13] WLAN architecture is built around a Basic Service Set (BSS). A BSS is a set of stations that communicate with one another. ...
Network parameters are vital ingredients of today's data communication scenario especially Media access delay, Throughput and CPU utilization. This Paper has taken into consideration the modeling and implementation of Wireless Local Area Network (WLAN) based on OPNET simulator and evaluated the performance of the wireless local area network in a campus/university environment. Our model has been tested for two types of applications (PRINT and EMAIL) in two different sites each comprising of 20 users and found that among a set of other parameters Media Access Delay and Load were highly affected by the number of users per application with and without load balancing. Simulation performed shows the behavior of load balancing on wire-line and wireless network for both applications (print and email) [5-15].
Voice Over Internet Protocol (VoIP) properties are vital for its reliability in mission-critical applications. This research aims to find network topology, call signalling and voice codecs property combinations that meet reliability targets of VoIP communication in a Small Office Home Office (SOHO) environment where network resources may be limited but reliable and secured operation is essential. Local Area Network (LAN) and Wireless LAN (WLAN) scenarios are evaluated using Quality of Service (QoS) and Mean Opinion Score (MOS) measurements to find which property combinations satisfy predefined classes; best quality and best performance. The research extended Roslin et al. [1] on LAN VoIP to WLANs, and validated Khiat et al. [2] s and Guy [3]’s work that argued SIP was effective in optimal set up. This research found that VoIP combinations offer some desirable characteristics, but at the cost of other properties required, leading to categorisation being based on the interpretation of the results, concluding that though, not ideal for mission-critical applications, combinations function well in replicating real-world scenarios. The analysis also established VoIP's scalability for application-based configurations, impact of VoIP’s modularity and ease of configuration in achieving user expectations. Further property testing can solidify VoIP’s capabilities to function for mission-critical environments.
Visual and Vocal communication can be transferred through Circuit switched Network or Packet Switched Network. Public Switched Telephone Network (PSTN) is not an affordable option therefore over existing packet switched network, Voice over Internet Protocol (VoIP) has become a preferable alternative due to its reduced cost. However, despite its reduced cost it has so many challenges which affect its successful deployment. This is because; the quality of VoIP is mainly affected by jitter, delay, packet loss and some other parameters. This research was carried out to evaluate voice quality in VoIP experimentally under different scenarios using OPNET network simulator. A VoIP network was simulated using Riverbed modeler academic edition 17.5 and the behavior and quality of VoIP was studied and analyzed under different scenarios. The results of the analysis and the performance evaluation are presented in this paper. This work can guide researchers and designers to design a network for VoIP services and its deployment. It can also guide the operators to choose speech compression technique for better voice quality.
Paper investigates that how the performance of the wireless local area networks (WLANs) is effected by the redundancy in a office environment. Environment has been simulated, where many applications are in use at a time and their mutual effects thereof. Here, OPNET has been used to develop a new model suitable for office/company environment. Performance of the WLANs has been evaluated against two types of applications (FTP and DATABASE) for a office/company environment. This model provides us the significant advantage of not having any single point of failure. It has been shown here that there is no degradation of the performance if one of the path has been failed for the communication.
has become an interesting topic of research in both the internet and the telecommunication industry. The tremendous increase in popularity of VoIP services is a result of huge growth in broadband access. In wired as well as in wireless communication, VoIP is expected to completely replace the traditional telephony approaches. To provide a good quality speech through VoIP applications, certain QoS parameters must be analyzed. These QoS parameters help us to evaluate the performance of various networking protocols, voice encoding schemes, etc. Numbers of QoS techniques like IntServ, DiffServ and RSVP are adopted to ensure good quality in IP based networks. In this paper, our main contribution is to analyze and evaluate the performance of various voice encoding schemes using RSVP, in a VoIP based wireless LAN. The network model designed in this paper is based on OPNET IT GURU Academic Edition. Various scenarios for different voice encoding schemes using RSVP are setup in the OPNET simulation environment. Different parameters that indicate the QoS like throughput, end to end delay, delay variations, traffic send, traffic received, etc. are calculated and analyzed in WLAN. KeywordsWLANs, QoS, IntServ, DiffServ, RSVP.
