Brooks Communication, Inc. - Internet Protocol Telephony

Internet Protocol is a "packet" protocol. Data in a file is broken up into data packets of fixed or varying size, sent to the receiving computer, and reassembled into a duplicate of the original file. There are error checking schemes to make sure the received file is identical to the sent file. If a packet does not arrive intact or is missing, the receiving computer asks the sending computer to resend the packet. This simplistic explanation overlooks the complex tasks of breaking data into packets and reassembling them in the correct sequence at a remote location and making sure each packet goes to the correct computer or server on the Internet. For the most part, packets do not even take the same route to a destination: Each packet is treated as an individual file and my be sent on different routes depending upon network traffic. That process is PFM (pure magic).

Everything in a packet network takes time. It takes time (a very, very short time) to take data from a file and fill the individual packets. It takes time to transfer the packets from one computer to another. It takes more time to transfer packets across the Internet than it does to transfer packets within your local area network. It takes time to resend packets that have become corrupt, are missing, or have collided with another data packet. All these times add to the total delay. The total delay is called latency. A number of programs are available which measure total latency and will give you an indication of how quickly data is getting from one place to another.

How does Voice work?

Telephone voice networks work in "real time". Even though each telephone call is digitized, transmitted, and reassembled just like IP calls, the protocol is completely different. Voice calls are digitized "on the fly" so there is no time waiting for packets to be filled: Each digital element of voice is transmitted immediately upon being digitized. Each voice call uses a dedicated transmission path that remains for the entire conversation. As the digital information is converted to analog speech at the receiving end, we hear it immediately. Although there is some latency in voice transmission, the latency is generally under 10 ms (10/1000 second). As a result, we perceive voice telephone calls to be real time, with no interruptions, and with the ability for both parties to talk at the same time (full duplex).

How well does Voice over IP work?

Compromises must be made when transmitting voice signals over IP transmission medium. Current standards require about 10 ms to fill each packet and about 10 ms to empty the packet at the other end. A good latency time for the Internet is 50-60 ms. This gives a total latency of 70-80 ms. At this level conversations over IP are quite good - about as good as the quality on a cell phone call. You probably can't expect full duplex operation and you shouldn't expect to be able to use a speakerphone effectively. At 100-120 ms of latency you can start to have echoes on your conversation. These are usually annoying but do not interfere with the conversation. Greater than 120 ms of latency generally results in dropped syllables and can make conversation difficult to impossible.

If you implement IP Telephony over a private network where latency time can be controlled and where voice packets can be given priority over data packets, results can be quite good. If IP Telephony is implemented over the Internet, results can vary from good to abysmal - all depending upon the route taken by individual packets and the overall speed of the Internet at a given time.

IP Telephone Applications

Converged Communications Platforms unify technologies such as wireless, cellular, LAN/WAN, Internet and PDA into a single enterprise. With analog, digital, wireless and Voice over IP (VoIP) built into the same platform, you have the power to choose from a variety of solutions at a pace that's right for your organization. Whether you want to blend traditional and IP solutions or deploy full IP, we offer a collection of applications and endpoints that will enable you to harness the power and benefits of Voice over IP.

Seamlessly connecting multiple locations together has never been easier or more cost-effective. The Inter-Tel Converged Communications Platforms are designed around a distributed open architecture for maximum efficiency and reliability. Whether you have 2 sites or 60, all of your locations can interact in a completely seamless fashion over your IP data network.

Your phone system is everywhere you want it to be. With feature-rich, IP-based endpoints, your data network connects with your remote employees, satellite offices and warehouse facilities as if they were all in the same building. Because IP phones are fully integrated with Mitel's Converged Communications Platforms, users anywhere have access to the functions of the phone system such as transferring calls, conferencing, accessing voice mail, record-a-call and much more. Even call center agents working offsite do not sacrifice functionality. They can be part of hunt groups or call routing patterns, and supervisors can monitor their calls as if they were in the office. Additionally, the IP PhonePlus and IP SoftPhone eliminate the need for a separate phone system in each location, and they provide you with function, mobility and flexibility like you've never experienced.

Brooks Communication is committed to delivering standards-based IP telephony solutions. Mitel's Converged Communications Systems use standard Media Gateway Control Protocol (MGCP) to communicate with third-party gateways. This allows Call Processing to use remote gateways as access points for inbound and outbound call functions. MGCP also provides local dial tone and 911 capabilities to remotely deployed IP phones. Trunk interfaces can be deployed remotely, without a remote telephone system. Mitel's SIP (Session Initiation Protocol) Server provides a foundation for standards-based communications within a converged IP environment. SIP allows Call Processing to communicate with products, such as the Cisco 7960 IP Phone, Cisco voice-enabled routers, Windows® XP, Microsoft Passport® network and a series of upcoming Mitel SIP-enabled IP phones.

Unified Communicator enables your mobile and remote associates to have full control of their Inter-Tel endpoints from virtually anywhere through multiple user interfaces including: Speech recognition, touchtone, PC Web browser and Wireless Application Protocol (WAP) devices such as cellular phones and handheld computers (PDAs). You can also forward calls to cells phones, pagers, voice mail and alternate numbers so important matters can be handled anytime, anywhere. Increase your efficiency with access your Personal Address Book and the System Directory, control your availability and location, check availability of fellow associates and initiate calls from a Web browser or WAP-enabled device.

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