Polycom SIP 2.2.2 manual Overview

Overview

—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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