Polycom SIP 2.2.2 Telephone User Manual


 
Overview
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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.