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IP Unified Communication Solutions  (VOIP)

Moving to an IP telephony infrastructure represents a significant shift from more traditional voice services. An understanding of the fundamental differences between circuit- and packet-based communication systems is key to successfully leveraging the new opportunities that unified communications provides.

IP telephony infrastructure solutions vary widely among vendors, but they all share core functionality and architecture design approaches that are important to understand when integrating unified communications solutions.

Traditional PBX systems were isolated from the data network and from other voice systems within the organization. Phone switches connected directly to desk sets and required a separate nest of cabling to hook everything together. Moving to IP and network-based telephony unifies voice and data on a common infrastructure, not only eliminating extra cable runs throughout an office building but opening the door to a level of integration not usually possible on dedicated voice systems. Understanding and making provision for the stringent requirements of this new enterprise communication system is key to delivering the benefits this system has to offer to every enterprise.

At netEDGE we understand the stringent performance and security requirements of your IP Telephony (VOIP) communications  system as they are a key component of your enterprise business infrastructure. From system and site analysis to design and implementation we are well equipped to deliver the best solution to address your business needs.


OEMS and Technology components of our IP Telephony (VOIP) solutions 

Cisco Systems

Bootstrap 4 has been noted

Designed for small and medium businesses, as well as large enterprises, Cisco UCM provides voice over IP technology that can replace a legacy telephone system.

Cisco's Unified Communications Manager (Cisco UCM) is an IP-based communications system that supports voice, conferencing, video, data, instant messaging and mobility services. With Cisco UCM, businesses can replace their legacy PBX telephone system with a modern, secure VoIP system that can scale beyond a legacy PBX and provide unified communications services to multiple locations.

Because of the modular design of Cisco UCM, it is suitable for virtually any size company. It uses a server-clustering design, which allows each cluster to reach 30,000 users. Organizations can deploy multiple clusters, significantly increasing this number to well over 100,000 users. Hardware requirements for a Cisco UCM (CUCM) deployment will depend on the organization's size and which add-ons are selected.

CUCM is available as a standalone product or part of the Cisco Collaboration Solutions -- Cisco Business Edition 6000 (BE6K) and 7000 (BE7K). The BE7K edition is a family of end-to-end collaboration services designed for fast-growing enterprises, while the BE6K is targeted toward enterprises ranging from 25 to 1,000 users, and provides the necessary applications to support business growth. Both BE7K and BE6K are offered on a virtualized platform running under the VMware hypervisor.


One of Bootstrap 4's big points

A VoIP Phone System / IP PBX system consists of one or more SIP phones / VoIP phones, an IP PBX server and optionally includes a VoIP Gateway. The IP PBX server is similar to a proxy server: SIP clients, being either soft phones or hardware based phones, register with the IP PBX server, and when they wish to make a call they ask the IP PBX to establish the connection. The IP PBX has a directory of all phones/users and their corresponding SIP address and thus is able to connect an internal call or route an external call via either a VoIP gateway or a VoIP service provider to the desired destination.

At the center we have, the IP PBX. Starting from the bottom, we see the Corporate Network. This is the company’s local network. Through that network, Computers running SIP clients such as the 3CX softphones, and IP Phones connect directly to the PBX. On the left, we see the company’s router/firewall connected to the internet. From there it can connect to remote extensions in the form of computers running the softphones, remote IP Phones, mobile devices running the 3CX Android and iOS clients, and Bridged PBX’s. Using a VoIP provider we can connect to the PSTN network. To the right a VoIP Gateway connects the PBX directly to the PSTN network.



Asterisk, open source PBX software, is richly featured and benefits from fast development in the open source community. Better yet, it can save you money.

Did you know that there is completely open source, Linux-based VoIP software? Welcome to Asterisk. Written by Mark Spencer of Digium (A Huntsville, Alabama-based company), its code has been modified by the open source community, along with the usual testing and bug fixes, which have really helped the development of this software in terms of enhancing the stability and increasing its feature set. 

Asterisk is open source PBX software that runs on Linux, BSD and MacOS X. It has pretty much all the features that one gets from mainstream PBX software and supports VoIP over many protocols. It can also give you the interoperability that one needs when using different types of telephony equipment. (Yes, it can work with Nortel.) Some features include voicemail service along with directory, voice response, call queuing and call conferencing, support for three-way calling, caller ID and SIP. From a hardware perspective, it needs no additional hardware to provide VoIP services. To connect to existing telephony equipment, it supports many different types of hardware devices, though it has a close partnership with Digium hardware, who is their sponsor.