Polycom SOUNDPOINT SIP 2.2.0 User Manual

Page 27

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Overview

2 - 11

Multiple Registrations

—SoundPoint IP phones support multiple s per

phone. (SoundStation IP 4000 supports a single .)

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.

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.

Background Noise Suppression

—Designed primarily for hands-free

operation, reduces background noise to enhance communication in

noisy environments.

Comfort Noise Fill

—Designed to help provide a consistent noise level

to the remote user of a hands-free call.

DTMF Event RTP Payload

—Conforms to RFC 2833, which describes

a standard RTP-compatible technique for conveying DTMF dialing

and other telephony events over an RTP media stream.

DTMF Tone Generation

—Generates dual tone multi-frequency

(DTMF) tones in response to user dialing on the dial pad.

IEEE 802.1p/Q

—The phone will tag all Ethernet packets it transmits

with an 802.1Q VLAN header.

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