Warren Romanow
Warren Romanow has been in the interconnect industry since 1984, selling, distributing and networking telecommunications equipment. He constructs systems inclusive of; the telephone (VoIP), voice mail system, unified messaging, screen synchronization, database look-up, facsimile networks, LAN/WAN, data & desk top environments, pricing for local service, lata, 1+ dialing, T1/T3 and point-to-point communications.
Warren explains; “There were four objectives in the composition of this book. 1. If you want to know from Alexander Graham Bell to present technology, I have detailed it in an easy to understand language. 2. Real life examples of customers who should use IP telecommunications. 3. How to put the entire network together making it run smoothly and seamlessly. 4. Who to use to construct and achieve this entire network extraordinaire and why.”
And here’s an offer you’ll never get from any other author. You see, Warren Romanow doesn’t just write about VoIP networks, he actually builds them. Warren Romanow and his team of consultants are so confident in their abilities to provide you a successful solution that they are now available to hire as a consultative team for your project. You can reach Warren Romanow at:
DataNet Telesys Group
8670 Dresden Court
Alta Loma, CA 91701
ipnetworks@charter.net
Warren Romanow has been in the interconnect industry since 1984, selling, distributing and networking telecommunications equipment. He constructs systems inclusive of; telephones (IP), voice mail systems, unified messaging, screen synchronization and database look-up, facsimile networks, LAN/WAN, data and desk top environments, and pricing for local and long distance services.
He has held the positions of, Vice President of Telrad Canada, President of the Comtel Data Group and C.E.O. of Comtel International. Warren has worked with companies and installed telecommunications equipment and services in Canada, the United States, Egypt and Saudi Arabia.
Warren Romanow is now residing in Alta Loma, California where he owns and operates DataNet Telesys Group. DataNet specializes in seamless IP telecommunications, system networks, software integration, screen synchronization, innovative line restructuring, facsimile networks and cost justifications. Warren is a member of the International High IQ Society (membership number 3378 / I.Q. of 138) and is extremely proficient in customer cost analysis and often pioneering new telecommunications strategies.
There are four objectives in writing this book. 1. If you want to know from Alexander Graham Bell up until today, Warren explains it in an easy to understand language. 2. Actual real life examples of customers who should use IP telecommunications. 3. How to actually put the entire network together making it run smoothly and seamlessly. 4. Who to use to construct and achieve this entire network extraordinaire and why.
Warren explains, “I thought it best to put together a guide for the average business to follow. Roughly 86% of all business'' fall into the small to medium size range, and although their requirements vary, their most basic needs remain the same. I let individuals, businesses, and vendors know whom they should deal with for the bulk of what they need. That is, if you want to sleep at night.”
Design Considerations
During the past 15 years data networking has moved from a hierarchical design to a distributed network. In the present scenario, it is considered to be the most viable solution to provide a reasonable approach for communication. Today’s data networks provide two main functions:
a) Physically connecting to the end user (PC) or the WAN links.
b) Switching the data traffic between them.
Some of the major applications of data networks provide facilities with add-ons, such as: e-mail, spreadsheets, video, and so on.
For example, in the data-networking arena, how we have moved from large and exclusive IBM SNA networks to inexpensive routed networks that connect everyone, potentially, with everyone else. Unpicking the building blocks that make up the network and providing something with which everyone can participate, is how this was created. The same is true of voice systems. All we need to do is look at how we unpick the building blocks and review the technical considerations.
Call Management
Call management (or call control) and directory services are key elements in the network design. Whether this is built into the software on a PBX or a Soft PBX, this function is critical. It also defines the capability of the network in terms of availability, performance, scalability and administration. Taken as a fundamental building block, its performance as a switching engine is paramount.
Voice Quality
Originally, IP was conceived as a transport protocol for data traffic, not real-time voice and video traffic. As a result, when IP networks are heavily loaded, real-time delivery of voice is affected by serious quality issues due to the lack of provisioning for voice traffic. These issues include delay, jitter, and packet loss.
As discussed earlier, delay is measured by the amount of time it takes for a packet to travel from point ‘A’ to point ‘B’. High end-to-end delay will result in echo and talker overlap – the problem of one talker stepping on the other talker’s speech). Jitter is the variation in the inter-packet arrival time. As voice is sampled, it is separated into multiple packets. These packets may not arrive at their destination in the same sequence in which they were created, because some of the packets may have taken a longer route to reach their destination. Limiting jitter requires holding packets long enough to allow the slowest packets to arrive in time to play in the correct sequence. Packet loss occurs at times of peak loads and congestion, which is due to the link failure or inadequate bandwidth, whereby packets are dropped.
Research shows that users face difficulty in dealing with a delay in voice transmission of more than 150 to 200 milliseconds or a packet loss greater than 10%. There are numerous methods under development to improve voice quality in IP networks. Echo cancellers and jitter buffers promise to control delay and packet loss in IP voice transmissions. Engineering an IP network that minimizes end-to-end delay and packet loss is essential in providing acceptable quality of service.
Unfortunately, poor quality of service is a common occurrence in an IP Telephone System because of the dynamic traffic loads on the LAN. Efforts are underway to improve voice quality by developing features that use bandwidth allocation techniques to reserve the necessary bandwidth required for voice traffic. ESI’s IVX is a prime example of a company who has solved this problem by installing a separate circuit board in each IP telephone that ‘reserves’ bandwidth to ensure voice quality. In addition, development is underway to improve the efficiency of standards used to packetize and compress voice. However, the wide-scale implementation of these features for most company’s is still one to two years away. During this time, progress will be made so that reliable interoperability between different manufacturers’ systems is assured. The bottom line is this; know what you are buying, its limitations, and current operating functionality within the scope of your environment. Then you will be able to set up your system to function, as you would have it.
Interoperability
Since standards for IP voice transmission have not yet matured, interoperability issues exist at all points in the IP network. Today, mainly proprietary voice compression algorithms are used to packetize voice. Standards such as Session Initiated Protocol (SIP) and H.323 are maturing. In the future a standard algorithm for packetized voice will solve the problem of interoperability between IP Telephone Systems, if adopted and adhered to. ESI’s IVX has solved this problem with proprietary software that allows a simple ‘open standards’ protocol if you will, which allows them to operate with any current standard being used.
Capacity
Voice and Video both require a minimum bandwidth, including compression techniques to be useable. Therefore, network capacity must be calculated to cope with peak capacity under all circumstances.
Latency
As with capacity there are strict requirements for network latency. The voice and video packets must traverse the network in 50-100ms. Hardware latency as well as link latency must be calculated beforehand.
Quality of Service (QoS)
Different Applications; For example, if we treat voice and video as applications for the time being, then they require minimum standards as well. QoS standards such as 802.1q, 802.1p, DiffServ, MPLS must all be rigorously adhered if the goal is to be achieved.
