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VoIP basics

When I start working on a new exam one of the first things I do is checking the contents of the books. My ONT Official Exam Certification Guide have three parts: VoIP, QoS and Wireless. Out of 10 chapters Six (6) go for QoS and three (3) for wireless which leave one small chapter for VoIP.

VoIP is why we’re here to begin with. It is the reason QoS is so important and understanding it is important. This post will go over the foundations of VoIP as reflected on the ONT exam. It does not cover the in-depth of things as ONT take a relatively brief look at this topic.

First, some of the main reasons to migrate a telephony system to IP-based systems:

  1. More efficient use of bandwidth and equipment
  2. Consolidated network expenses
  3. Improved employee productivity
  4. Access to new communication devices

Personally, I think the mobility of devices in an IP network is the single most important reason to convert a network.

There are two major steps in a network migration to VoIP environment:
Phase I migration – keep existing system and connect the PBX to the router. The router can convert the calls to VoIP and save cost, QoS with low-cost.
Phase II migration – IP systems replacing all phones and PBX. This is a full IP-based environment.

Few terms:
Call control = routing voice around the network.
Distribution= every device has brain, every router have to be configured with all details.
Centralized = call agent is the center point with database for all calls in the network.
Cisco Call Manager (CCM) is a centralized solution.

FXS (Foreign Exchange Connections) is an analog interfaces that connect with old devices. Each analog port can run one call only, not efficient and costly. FXS ports plug to station and generate dial tone.

FXO (Foreign Exchange Office) convert analog to VoIP. It receives the dial tone.

E&M (Ear and Mouth) Receive and Transmit, create direct trunk between PBXs or between PBX to a router.

FXS, FXO and E&M are all analog, they all use one call per line.

The next step of understanding VoIP is understanding how does voice become a packet. There is a four step recipe to turn voice into bits:

  1. Sampling – take many samples of the analog signal
  2. Quantization – calculate a number representing each sample
  3. Encoding – convert that number to binary
  4. Compression (optional) – compress the signal

I wrote about waveform, Nyquist Theorem – the man and the theory so I won’t repeat myself. This is the most important part, the basic required to master QoS – my next topic that will get most of my attention.

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