Overview

Multilingual User Interface—All phones except SoundPoint IP 301 have multilingual user interfaces.

Multiple Call Appearances—The phone supports multiple concurrent calls. The hold feature can be used to pause activity on one call and switch to another call.

Multiple Line Keys per Registration—More than one line key can be allocated to a single .

Multiple Registrations—SoundPoint IP phones support multiple registrations per phone. (SoundStation IP 4000 supports a single registration.)

Network Address Translation—The phones can work with certain types of network address translation (NAT).

Presence—Allows the phone to monitor the status of other users/devices and allows other users to monitor it. Requires call server support.

Real-Time Transport Protocol Ports—The phone treats all real- time transport protocol (RTP) streams as bi-directional from a control perspective and expects that both RTP end points will negotiate the respective destination IP addresses and ports.

Recording and Playback of Audio Calls — Recording and playback allows the user to record any active conversation using the phone on a USB device. The files are date and time stamped for easy archiving and can be played back on the phone or on any computer with a media playback program what supports the .wav format.

Server Redundancy—Server redundancy is often required in VoIP deployments to ensure continuity of phone service for events where the call server needs to be taken offline for maintenance, the server fails, or the connection from the phone to the server fails.

Shared Call Appearances—Calls and lines on multiple phones can be logically related to each other. Requires call server support.

Synthesized Call Progress Tones—In order to emulate the familiar and efficient audible call progress feedback generated by the PSTN and traditional PBX equipment, call progress tones are synthesized during the life cycle of a call. Customizable for certain regions, for example, Europe has different tones from North America.

Voice Mail Integration—Compatible with voice mail servers.

Audio Features

Acoustic Echo Cancellation—Employs advanced acoustic echo cancellation for hands-free operation.

Audio Codecs—Supports the standard audio codecs.

Automatic Gain Control—Designed for hands-free operation, boosts the transmit gain of the local user in certain circumstances.

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Polycom SIP 3.0.2 manual Overview