Glossary

SIP

SIP endpoint

SLIC

SOHO

Session Initiation Protocol. Protocol developed by the IETF MMUSIC Working Group as an alternative to H.323. SIP features are compliant with IETF RFC 2543, published in March 1999. SIP equips platforms to signal the setup of voice and multimedia calls over IP networks.

A terminal or gateway that acts as a source or sink of Session Initiation Protocol (SIP) voice data. An endpoint can call or be called, and it generates or terminates the information stream.

Subscriber Line Interface Circuit. An integrated circuit (IC) providing central office-like telephone interface functionality.

Small office, home office. Networking solutions and access technologies for offices that are not directly connected to large corporate networks.

T

TCP

Transmission Control Protocol. Connection-oriented transport layer protocol that provides reliable

 

full-duplex data transmission. TCP is part of the TCP/IP protocol stack.

TFTP

Trivial File Transfer Protocol. Simplified version of FTP that allows files to be transferred from one

 

computer to another over a network, usually without the use of client authentication (for example,

 

username and password).

TN power systems A TN power system is a power distribution system with one point connected directly to earth (ground).

 

The exposed conductive parts of the installation are connected to that point by protective earth

 

conductors.

TOS

Type of service. See CoS.

U

UAC

UAS

User agent client. A client application that initiates the SIP request.

User agent server (or user agent). A server application that contacts the user when a SIP request is received, and then returns a response on behalf of the user. The response accepts, rejects, or redirects the request.

UDP

User Datagram Protocol. Connectionless transport layer protocol in the TCP/IP protocol stack. UDP

 

is a simple protocol that exchanges datagrams without acknowledgments or guaranteed delivery,

 

requiring that error processing and retransmission be handled by other protocols. UDP is defined in

 

RFC 768.

user agent

See UAS.

V

VAD

 

Voice activity detection. When enabled on a voice port or a dial peer, silence is not transmitted over

 

 

 

the network, only audible speech. When VAD is enabled, the sound quality is slightly degraded but

 

 

 

the connection monopolizes much less bandwidth.

 

 

 

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

 

 

 

 

GL-6

 

OL-4008-01

 

 

 

 

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Cisco Systems ATA 188 manual GL-6

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.