CS-830 IP phone is an internet based voice network phone terminal. PHONE series IP phone adopts multiple voice control protocols and voice compression coding methods to directly convert analog voice into IP packet for internet transport, thus effectively using the existing bandwidth to provide PSTN quality voice service.
CS-830 IP phone supports SIP protocols,Support Bridge and Router model. offers Two Ethernet interface and is compatible with various softswitch systems and VoIP voice gateways to provide broadband IP voice service..
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Supported specification and applications
data networking:
MAC Address
TCP: Transmission Control Protocol
DHCP: Dynamic Host Configuration Protocol
PPPoE: PPP Protocol over Ethernet
SNTP, Simple Network Time Protocol
STUN - Simple Traversal of User Datagram ...
MD5 Message-Digest Algorithm
TOS:Type of service
QoS: Quality of service control option
DNS: Domain Name Server
Support Bridge and Router model
RTP: Real-time Transport Protocol
RTCP: Real-time Control Protocol
Telnet:I nternet's remote login protocol
HTTP: Hyper Text Transfer protocol
FTP: File Transfer protocol
TFTP: Trivial File Transfer Protocol
call control /voip Features
SIP RFC3261
Tone generation and Local DTMF re-generation according with ITU-T
voice algorithms
G.711(A-law or u-law)
G.723.1(6.3kbps,5.3 kbps, high/low)
G729 ,G.726
AGC(Auto Gain Control)
G.168/165 compliant 16ms echo cancellation
AEC(Auto Echo Cancellation)
VAD (Voice Activity Detection)
CNG(Comfort Noise Generation) -
Key features
voice Features
call hold
Unattended/ attended call transfer
call waiting and per call-waiting blocking
call forward (Always, busy, no answer, power off)
voip speed dial
VoIP digital map
user define ring tone
E.164 dial plan and customize dial rules
caller number/name display
MWI(Message wait Indication)
voice promote
configurable jitter buffer size
configurable audio frame
Support Bridge and Router model -
data features
Static/Dynamic WAN-IP-Addressing
PPPoE
sip voip phone