E1 Basics

RAS96 RASCard User Guide

FXS: Foreign Exchange, Subscriber; the customer premises interface that, along with FXO, allows the phones to act as if connected locally to the main PBX, but without the cost.

G.704: A series of ITU standards for E1 frame formatting. (e.g., section 2.1.3.1).

ITU: (International Telecommunications Union) a United Nations standards agency.

IXC: Interexchange Carrier; a long-distance phone carrier, such as AT&T, MCI, or Sprint.

LBO: Line Build Out; the insertion of loss in a short transmission to make it act like a longer line.

PBX: A small, privately-owned switch within a company.

POP: Point of Presence; the inter-exchange carrier’s central office (CO).

Yellow Alarm: a type of carrier failure indicating a remote alarm condition.

E1 service provides a two-way digital telecommunications connection at 2.048M bps. The local PTT (a country’s national Postal, Telephone and Telegraph agency) provides the entire circuit within a country, and interconnects with one or more other circuits for international network connections.

E1’s higher equipment and leased line costs are more than offset by its inherent advantages: reduced phone bills (payback in months), increased control of the network, improved reliability, quick and cheap change implementation, vastly increased speed and improved voice quality due to the nature of digital vs. analog lines.

The CEPT E1 speed of 2.048M bps is derived from 30 channels at 64K bps each, plus 8K bps for synchronization. To be transmitted effectively, the normally incompatible voice and data must be “mixed” for compatibility. When digitizing the analog voice signal, there is a question of the number of bits that can be transmitted economically, and how to best represent the “smooth variation” in loudness. “Best” typically implies maximum voice quality, but there can also be tradeoffs for cost, circuit availability, bandwidth, and reliability. The current world-wide standard for digital voice is PCM (Pulse Coded Modulation). A “codec” (coder-decoder) selects the value closest to the true analog signal, minimizing the distortion, and making the voice transmission acceptable to the human ear. Technologies like aliasing (voice signal filtering) and non- linear sampling are used to overcome problems in performing PCM voice compression. The method of non-linear sampling used in Europe, Mexico and the U.K. is called A- law. Similar technology is used at the sending and receiving ends (i.e., analog-to- digital conversion at one end, and then digital-to-analog conversion at the other end).

The frequencies around 1000 Hz convey most of the information in a person’s voice. Several methods of non-linear sampling are available, including Pulse Code Modula- tion (PCM), Differential PCM (DPCM), Adaptive DPCM (ADPCM), CVSD, VQC, and HCV. Each has its own associated data transfer rate, cost, and quality factors.

42	CommPlete Communications Server