VoIP monitoring is a practical solution for anyone whose network utilizes VoIP technology. VoIP is a form of voice communication over IP networks such as the Internet or wireless networks. It allows organizations to replace traditional telecommunications networks with data networks that can transmit voice and multimedia communications over any broadband connection. There are many benefits of VoIP, including: reduced capital expenditures, improved efficiency, elimination of long-distance charges, reduction in operational costs, easy configuration, and flexibility. In order to take advantage of all of these benefits, it is essential for businesses to monitor and evaluate their VoIP usage to ensure that they are using all of its features and that it is meeting its objectives.
VoIP monitoring is typically provided by specialized voip monitor applications. This kind of application allows an organization to receive alerts and performance reports in real time from their VoIP monitoring solutions. These applications are able to provide information on bandwidth usage, latency, jitter, packet loss, jostled call times, total number of calls, user agent, processing rate, etc. The specialized voip monitor also allows users to track their own calls and receive alerts when they exceed any bandwidth limits, exceed any latency standards, or experience any form of dropped calls.
One example of an application that allows a network administrator to monitor VoIP usage is Site24x7 APM. This monitoring solution works with Site Simulator, a freely available web-based tool that allows users to interact with various online applications. Users can simply visit a site, enter their log in information, and begin interacting with the application. Once they have begun browsing, they will be able to view their call logs, obtain statistics, and see their bandwidth and latency in real-time. In addition, Site Simulator offers advanced features such as Jitter Analyzer and Rate Limiting, which can be used with a Site Builder and Website Analytics.
In addition to the application tools mentioned above, Site Simulator offers a number of other tools that will help network administrators maximize their VoIP monitoring experience. The Site Simulator Dashboard is just one of these tools. The Site Simulator dashboard is designed to show a user how to customize their IP settings, which includes setting the connection to simulate a DSL, Cable, Fiber Optics, or wireless connection. A prime feature of the dashboard is the Site Simulator Round Trip Sensor. The Site Simulator round trip sensor is a feature that verifies the round-trip distance that the server has covered in the span of a specified time frame. This useful metric can help administrators reduce unnecessary bandwidth usage and jitter caused by traffic that is outside the scope of the primary server.
For businesses that are interested in availing of highly efficient calling features, consider SteelCentral UCEXpert and VOIP Spear Specialized VOIP Monitor for your business. These packages are bundled with high quality steel core routers and switches along with a robust and scalable switch and router design. With these components, a business can enjoy enhanced voice and data quality and a more cost-effective phone and data service plan. You also get automatic phone setup with a pre-configured number set up through the internet prior to your first call.
One of the most common features of Voice over Internet Protocol (VoIP) systems is packet loss prevention. With a high speed internet connection, many users do not notice a significant decrease in voice quality when on a shared network. However, when packets of data are lost, the quality of voice drastically decreases. Packet loss can occur due to a number of reasons such as: firewalls preventing VoIP calls, internet connections that are too slow, corrupt or damaged servers, and other communication protocol issues.
An ideal solution to prevent packet loss and improve voice quality while on a VoIP network, a company should deploy SAAJitter, a form of automated handling technology. Utilizing state-of-the-art technology, SAAJitter automatically handles VoIP call flow on its own, taking care of dropped calls and avoiding busy signal zones. Its highly efficient routing algorithm intelligently decides how to route voice traffic based on information from both the SIP server and the network. The resulting real-time traffic flow results in real-time call quality and far fewer dropped calls.
To ensure optimal performance of their services, many organizations today use voipmonitor. The automated handling technology implemented into voipmonitor delivers improved business performance by focusing on preventing lost calls as well as improving voice and data quality. In addition, it can also be used as a monitoring tool to check if the organization’s internal processes are properly handling the traffic. The metrics generated from the system can also be used for identifying bottlenecks and improving the quality of services. With so many benefits and features, this service has been a great help for businesses trying to boost their productivity.