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 (SCCP)

 

 

 

 

GL-6

 

OL-3141-01

 

 

 

 

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ATA 188, ATA 186 specifications

The Cisco Systems ATA 186, or Analog Telephone Adapter, revolutionized the way traditional telephony interacted with Voice over Internet Protocol (VoIP) systems. Designed primarily for home and small office use, the ATA 186 allows users to connect standard analog phones and fax machines to a network, enabling them to take advantage of the benefits of VoIP technology.

One of the defining features of the ATA 186 is its dual port architecture. It includes two FXS ports, allowing users to connect up to two analog telephones. This functionality means that multiple devices can leverage VoIP services simultaneously without the need for separate adapters for each phone. The flexibility of the ATA 186 helps streamline the user experience, facilitating voice communication over an IP network while ensuring users can still use their existing phone equipment.

The ATA 186 employs various technologies to maintain high-quality voice calls. It supports standard voice codecs such as G.711 and G.729, which ensure efficient bandwidth usage while preserving call clarity. The adaptive jitter buffer technology further enhances call quality, compensating for network variations and minimizing latency, which is crucial for clear and uninterrupted conversations.

Additionally, the ATA 186 provides users with advanced calling features that were traditionally available only on PBX systems. These features include caller ID, call waiting, and voicemail functionality, integrating seamlessly with typical telephony services. The device also supports T.38 fax relay, allowing users to send and receive faxes over the internet, thus addressing the needs of environments where fax communication remains essential.

The security of VoIP conversations is also a priority for the ATA 186. It employs encryption protocols such as Secure Real-time Transport Protocol (SRTP) and Transport Layer Security (TLS), ensuring that voice data is securely transmitted across the network. This focus on security helps protect sensitive information transmitted during calls.

Installation and configuration of the ATA 186 is streamlined, with an intuitive web-based interface that simplifies the setup process. This accessibility makes it suitable for users with varying levels of technical expertise, as frequently required adjustments, such as network configurations and firmware updates, can be easily managed.

In conclusion, the Cisco Systems ATA 186 stands out as a versatile and robust solution for users looking to integrate analog phones into a VoIP environment. With its dual port capabilities, high-quality voice codecs, advanced call features, and security measures, it offers a compelling choice for both residential and commercial users seeking seamless telephony integration. As technology evolves, devices like the ATA 186 remain cornerstones in bridging traditional telephony with modern communication systems.
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