FXO

FXS

Glossary

Foreign Exchange Office. An FXO interface connects to the public switched telephone network (PSTN) central office and is the interface offered on a standard telephone. Cisco FXO interface is an RJ-11 connector that allows an analog connection at the PSTN central office or to a station interface on a PBX.

Foreign Exchange Station. An FXS interface connects directly to a standard telephone and supplies ring, voltage, and dial tone. Cisco's FXS interface is an RJ-11 connector that allows connections to basic telephone service equipment, keysets, and PBXs.

G

G.711

Describes the 64-kbps PCM voice coding technique. In G.711, encoded voice is already in the correct

 

format for digital voice delivery in the PSTN or through PBXs. Described in the ITU-T standard in its

 

G-series recommendations.

G.723.1

Describes a compression technique that can be used for compressing speech or audio signal

 

components at a very low bit rate as part of the H.324 family of standards. This Codec has two bit

 

rates associated with it: 5.3 and 6.3 kbps. The higher bit rate is based on ML-MLQ technology and

 

provides a somewhat higher quality of sound. The lower bit rate is based on CELP and provides

 

system designers with additional flexibility. Described in the ITU-T standard in its G-series

 

recommendations.

G.729A

Describes CELP compression where voice is coded into 8-kbps streams. There are two variations of

 

this standard (G.729 and G.729 Annex A) that differ mainly in computational complexity; both

 

provide speech quality similar to 32-kbps ADPCM. Described in the ITU-T standard in its G-series

 

recommendations.

gateway

A gateway allows SIP or H.323 terminals to communicate with terminals configured to other protocols

 

by converting protocols. A gateway is the point where a circuit-switched call is encoded and

 

repackaged into IP packets.

H

H.245

An ITU standard that governs H.245 endpoint control.

H.323

H.323 allows dissimilar communication devices to communicate with each other by using a standard

 

communication protocol. H.323 defines a common set of CODECs, call setup and negotiating

 

procedures, and basic data transport methods.

I

ICMP

Internet Control Message Protocol

Cisco ATA 186 and Cisco ATA 188 Analog Telephone Adaptor Administrator’s Guide (H.323)

 

OL-4008-01

GL-3

 

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ATA 188 specifications

The Cisco Systems ATA 188 is a versatile Analog Telephone Adapter designed to facilitate the integration of traditional telephone systems with Voice over Internet Protocol (VoIP) networks. This device has been key in bridging the gap between legacy telephony and modern IP-based communication, allowing users to leverage their existing analog phones while enjoying the benefits of digital connectivity.

One of the main features of the ATA 188 is its ability to connect regular analog phones to a VoIP network, enabling users to make and receive calls over the internet. This significantly reduces calling costs, especially for long-distance and international calls. The ATA 188 supports two phone lines, allowing simultaneous voice calls. This dual-line capability makes it a suitable choice for small businesses or home offices that require multiple lines without the need for extensive infrastructure.

The device is equipped with various technologies that enhance its functionality. It supports the Session Initiation Protocol (SIP) and H.323, making it compatible with a wide range of VoIP service providers. Additionally, the ATA 188 features Quality of Service (QoS) settings, which prioritize voice traffic over the internet, ensuring clear voice quality without interruptions or delays. This is essential for maintaining a professional communication experience, especially in business environments.

Another characteristic of the ATA 188 is its user-friendly configuration interface. It allows users to easily set up and manage their devices through a web-based portal. The configuration process is straightforward, with options to adjust settings such as codec selection, call features including call waiting, and call forwarding functionalities.

Security is also a priority for the ATA 188, as it provides robust protocols to protect call data. The device supports Secure Real-time Transport Protocol (SRTP) and Transport Layer Security (TLS) to encrypt voice traffic and manage signaling securely. This ensures that sensitive conversations remain confidential.

Overall, the Cisco Systems ATA 188 is a reliable and efficient solution for users looking to transition from traditional telephony to VoIP. Its dual-line capacity, compatibility with multiple VoIP standards, user-friendly configuration, and built-in security features make it a valuable asset for both personal and professional communication solutions. In an ever-evolving telecommunications landscape, the ATA 188 remains a relevant and practical choice for integrating legacy telephony with modern internet-based services.
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