應科院桌面SIP電話產品,
iSIP(暫時只提供英文版本) |
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iSIP
Phone is a SIP-compliant IP phone developed by ASTRI.
Equipped with a two-line display, it supports a
variety of advanced call features which are suitable
for both enterprise and residential use. As a standard-based
IP phone, it can interoperate with a number of 3rd
party SIP servers and phones.
I.
Standard Compliance
Being compliant with the latest Internet standards,
iSIP Phone supports the SIP call signaling protocol
(RFC 3261) and its extensions. Besides, to tackle
with the NAT traversal problem, iSIP Phone is capable
of using various solutions such as STUN and rport
for connecting to the different NAT networks. iSIP
Phone can also interoperate with a number of dominating
SIP service providers and products.
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II.
Toll-Quality Voice
Featuring multiple voice codecs like G.711A/µ, G.729A
and G.723.1, iSIP Phone delivers excellent voice quality.
Plus, its adaptive jitter buffering and echo cancellation
algorithms enable callers to enjoy uninterrupted conversation
even when delay and jitter are experienced over the IP
network. The fullduplex speakerphone ensures the voice
quality is optimal.
III.
Advanced Functions
Based on the latest VoIP technology, iSIP Phone supports
a variety of call features including caller ID display,
call waiting, call transfer, call forwarding, 3-way conferencing
and phonebook, which are necessary in residential and
business use.
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| IV.
Key Features |
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SIP-Compliant
(RFC3261) |
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Standard
PBX Functions |
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Two
Call Lines |
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3-Way
Conferencing |
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Call
History |
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Phonebook |
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Two
10/100 Ethernet Ports |
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2
x 20 Character LCD |
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24
Buttons For Keypad & Other Functions |
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4
LEDs For Status Indication |
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Handset |
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Speakerphone |
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NAT
Traversal |
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Phone
Menu And Web Configurations |
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Multiple
User Settings |
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Network
Upgrade |
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| Specifications |
| Call
Features |
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2
Call Lines |
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Call
Hold |
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Call
Waiting |
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Call
Transfer (Attended & Unattended) |
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Call
Forwarding (Unconditional, Busy & No Answer) |
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3-Way
Conferencing |
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Outgoing
Call Blocking |
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Call
Mute |
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DND |
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In-call
DTMF: Inband, RFC2833 |
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Caller
ID Display |
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Call
History |
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Phonebook |
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Redialing |
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Speed
Dialing |
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Direct
IP Call |
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| Audio
Features |
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G.711
A Law and μ Law (64 Kbps) |
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G.723.1
(5.3 Kbps) |
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G.729A
(8 Kbps) |
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Voice
Activity Detection (VAD) |
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Comfort
Noise Generation (CNG) |
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Adaptive
Jitter Buffering |
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Acoustic
Echo Cancellation (AEC) |
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Volume
Control (Ringer, Handset, Speakerphone, Hands-free) |
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| Protocol
Standards |
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SIP
(RFC3261) |
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SDP
(RFC2327) |
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RTP |
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Auto Media Negotiation (RFC3264) |
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SIP
REFER (RFC3515) |
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STUN (RFC3489) |
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rport
(RFC3581) |
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Digest Authentication (INVITE, BYE) |
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Network
Time (NTP) |
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TFTP (Firmware Upgrade) |
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HTTP
(For Web Administration) |
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| Configurations |
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Network:
Static IP, DHCP, PPPoE |
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Date/Time: Local, NTP (With Daylight Saving
& Timezone Settings) |
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Phone Menu Display |
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Remote Web Configuration |
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Web Password Protection |
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Easy
Setup Wizard |
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Multiple User Settings |
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SIP
Registration Tuning |
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Diagnostic Tools: PING, TRACEROUTE, SIP Log |
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Network Upgrade (TFTP) |
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| Hardware
Specifications |
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2
x 20 Character LCD Display |
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24 Buttons For Keypad & Other Functions |
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4 LEDs For Status Indication |
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Handset |
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Speakerphone |
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Hands-free
Support |
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Two-port 10/100 Ethernet Switch |
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Power Jack (9V DC @ 500 mA) |
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