VoIP Operators : From a Carrier Point of View (original) (raw)
Related papers
VoIP Systems Management using Internet Protocol Detail Records
2012
The increasing demand for latency sensitive services through the Internet imposes the development of networks capable of delivering quality of service. These networks require the use of enhanced traffic management tools. This paper performs an analysis of IP telephony or VoIP traffic considering Internet Protocol Detail Record (IPDRs). When a VoIP call occurs upon the Internet, a ticket (a file record) is generated to produce information regarding that specific call. These files are called Internet Protocol Detail Record. The IPDR, which is generated for every VoIP call, contains information related to the history of the call. The full set of information in the IPDRs carries a very comprehensive description of what happened to the call and can provide valuable information about the state of the network during the history of the call. Therefore, IPDRs can be used to establish network traffic baselines. This paper presents the development of a baseline that supports VoIP traffic management in Open Access MANs. Our main conclusion is that this method can be used to manage VoIP networks.
On the deployment of VoIP in Ethernet networks: methodology and case study
Computer Communications, 2006
Deploying IP telephony or voice over IP (VoIP) is a major and challenging task for data network researchers and designers. This paper outlines guidelines and a step-by-step methodology on how VoIP can be deployed successfully. The methodology can be used to assess the support and readiness of an existing network. Prior to the purchase and deployment of VoIP equipment, the methodology predicts the number of VoIP calls that can be sustained by an existing network while satisfying QoS requirements of all network services and leaving adequate capacity for future growth. As a case study, we apply the methodology steps on a typical network of a small enterprise. We utilize both analysis and simulation to investigate throughput and delay bounds. Our analysis is based on queueing theory, and OPNET is used for simulation. Results obtained from analysis and simulation are in line and give a close match. In addition, the paper discusses many design and engineering issues. These issues include characteristics of VoIP traffic and QoS requirements, VoIP flow and call distribution, defining future growth capacity, and measurement and impact of background traffic.
Telecollaboration: a case study for the performance of VoIP systems
… Conference WWW/Internet, 2004
The evolving nature of complex traffic issues of IP (Internet Protocol)-based Telecollaboration (TC) business system technology requires an intuitive understanding for the participants not just to make sense of design issues involved but also to provide an insight of guaranteed ...
Determining the Issues to Consider When Deploying VoIP onto a Small Enterprise Network
In this paper, we report some lessons learnt when deploying VoIP onto a small enterprise network. While these lessons resulted from a small network environment, some of the issues could have a more general applicability. Some theoretical analyses were first conducted, followed by some experimental work. An experiment was conducted for about 24 hours in order to capture the network measurement of a small LAN. The results obtained from the measurement were used in analyzing the whole network based on queuing theory. This provides ground for making judicious modifications to the network in order to support the deployment of VoIP. The modified LAN was simulated using OPNET for a period of 3 minutes after which the router began to drop some packets. With this, the simulation was terminated and results of the simulation were collected. The results obtained from queuing analysis agree with those obtained at the end of the simulation.
Implementing Enterprise VoIP Deployment
Voice over Internet Protocol (VoIP) is perceived as the best example of collaboration technology in today’s telecommunication space. There have being a paradigm shift from regular legacy telephone systems to VoIP over the last decade by most telecommunication and multinational companies globally. One of the huge benefit of VoIP is noticeable reduction in international call rates. In addition to the potential cost savings, VoIP offers the prospect of integrating voice with data and video applications, this goes a long way in increasing workers’ productivity within organizations. Many organizations are already embracing VoIP as replacement for their legacy PBX, others are integrating the existing PBX system with VoIP. This write up will focus on deployment of PBX-VoIP converged solution in enterprise organizations, where cost saving is one of the primary objectives
Voice over Internet Protocol, otherwise known as VoIP, has seen a lot of interest and patronage in the 20 th century. This is due to the development of its underlying network infrastructure and internet as a whole. Also, it brings to board a communication system that is much cheaper to use and maintain than the traditional legacy communication system. KNUST boasts of an efficiently functional VoIP deployment over its local network, allowing for users, mainly administrators and other faculty and staff members to make calls over the VoIP system. Though this system is reliable and serves its purpose well, it has failed to fully exploit other capabilities a VoIP system can employ. There is no integration of the Public Switched Telephone Network (PSTN) system, hence users can reach other users when they are away from their IP phones through the VoIP system. The reason for its absence is history of users misusing the PSTN service for long, personal calls. Also, calls to external networks contend for bandwidth on the router, delivering low quality call experience during peak hours. This project addresses all the above issues, with recommendations for future improvements. A proposed design is made for integration of PSTN with the existing VoIP system. This design incorporates control mechanisms to reduce misuse of the service by users. The design also tackles externally made calls, especially to satellite campuses and other universities. The new design passes all simulation tests for ITU-T recommended metrics. It is also low-cost and highly feasible, making it a suitable improvement to be implemented.
Deploying VoIP over a Small Enterprise Network
The possibility of voice communications traveling over the Internet, rather than the PSTN, first became a reality in February 1995 when Vocaltec, Inc. introduced its Internet Phone software. This technology has advanced rapidly. This paper is aimed at making suggestions on things to consider when deploying VoIP onto a small enterprise network. Though the suggestions to be presented are centered on small network, some of the issues are applicable to a bigger network. These suggestions are based on some theoretical analysis, which are supported with some experimental work. An experiment was conducted for about 24 hours in order to capture the network measurement of a small LAN. The results obtained from the measurement were used in analyzing the whole network based on queuing theory; this provides ground for making judicious modifications to the network in order to support the deployment of Voice over IP.
Comparison of Voice over Internet Protocol (VoIP) Performances in Various Network Topologies
Buletin Pos dan Telekomunikasi, 2020
VoIP is a digital communication technology that is currently developing because VoIP can be implemented on several network topologies, such as bus, star, and ring. Each of these topologies has advantages and disadvantages. So, a study is required to find out in which topology can VoIP be implemented optimally. In this research, VoIP is implemented in several topologies and furthermore the performance measurements are carried out for each topology. VQ manager is installed in order to measure the VoIP performance. For the server, we used Elastix and for the node implementation network topologies, we use several access points. From the results of the research, the performance of VoIP implemented in the star topology produces QoS that is better than other topologies with a delay value of 185 ms, 18 ms jitter, and 1% packet loss. This happens because in the star topology, all packets are distributed centrally. It is expected that the results of this research can be used as a reference in the application of VoIP technology in several types of topologies.
Scalable Resource Management Architecture for VoIP
This paper presents a new approach for scalable Quality of Service (QoS) based on a resource manager with the focus on Voice over IP (VoIP) in corporate networks. We first review the requirements of VoIP and consider typical network scenarios as well as existing approaches. In our approach several network domains have local resource managers (RM), which are in charge of the local network resources. This architecture has the advantage that only one new network element is needed, which also simplifies QoS signaling. We furthermore discuss ways for automatic configuration of the resource manager and enhance VoIP signaling (H.323) by QoS signaling with the resource manager. We show that the architecture fulfills the main requirements for VoIP, in particular minimal call setup delay and low management and installation effort.