SIP Phone,VoIP Phone
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„ Support IAX2 and SIP RFC3261 synchronously
„ Codec: G.711A/u,G.7231high/low,G.729
„ Echo cancellation: Support G.168,and Hands-free can support96ms
„ SupportVoice GainSetting,JitterBuffer,VAD, CNG, dualGK,call forward,and Peer to Peer
„ NAT transverse: support STUN client, Citron,AVS
„ SIP supportSIP domain, SIP authentication (none, basic,MD5), DNSname of server,Peerto Peer
and Private SIP server at the same time
„ SIP supportdualpublic server
„ DTMF: Support SIP info, DTMF Relay,RFC2833
„ SIP application: supportCall forward/transfer/holding/waiting
„ Callcontrolfeatures: Flexible dial map, support Hotline,EmptycallingNo.Rejectservice,Blacklistfor reject authenticated call,No disturb, Caller ID.
„ Support conference calland voice record
„ English, Spanish, Czechoslovakalternative
„ Networktoolsintelnetserver:Includingping,raceroute, telnetclient
„ CanupgradefirmwareandconfigurationfilethroughHTTP, FTP,TFTP
" Network Protocol
SIP v1 ( RFC2543),v2 ( RFC3261 ).
TFTP Client/DHCP Client/PPPoE Client .
Telnet/HTTP Server .
G.711: 64k bit/s ( PCM ) .
G.723.1:6.3k/5.3k bit/s .
G.726:16k/24k/32k/40k bit/s ( ADPCM ) .
G.729A:8k bit/s ( CS-ACELP) .
G.729B:adds VAD & CNG to G.729 .
VAD : Voice activity detection .
CNG: Comfortable noise generator .
LEC: Line echo canceller .
Packet Loss Compensation .
Adaptive Jitter Buffer .
3-way conference .
Out-of Band DTMF .
SIP Info .
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