Pure IP-Telephony
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RFC 4235 Dialoge State monitoring (partner Key)
RFC 3428 Extension for Instant Messaging
Quality of Service: priorisation of IP packets via TOS and DiffServ
VLAN priority (IEEE 802.1p / 802.1q)
RTP real time protocol – for speech transport
SRTP – secure speech transport (V 7.0 or higher)
RTCP real time control protocol
Voice Codecs: G.711 A-law / µ-law (64 kbps
),
G.723.1 (5.3 und 6.3 kbps),
G.729A (16 kbps)
VAD (Voice Activity Detection),
CNG (Comfort Noise Generation),
Dynamic Jitter Buffering
Echo Compensation: G.168
Administration
Access: via Web browser with HTML
Password protected with secure authentication
Bug tracking: log and trace files
Display of interface and connection status
Ping – connection test for Internet Protocol
sending SNMP Traps
Update: save and reload config files
boot code and firmware update via HTML-Upload
automatic update via update server
Supplementary services
Call Transfer with and without consultation
Call Diversion / Redirection
Call Hold / Retrieve with music on hold
Call waiting acoustic and optical signaling of waiting calls
Message Waiting displays a waiting message on the phone
Partner Pickup displays on the phone the possibility for pickup
Pickup-List displays on the phone the list on pickup able calls
Name Display additional name signaling on the phone
Call Completion on busy and no reply
3party conference conference with 3 subscribers, inclusive external subscriber
Calling line identification for different ring tones depending an the call (external, VIP, …)
Direct speech partner function, the remote phone device accepts the call automatically in loudspeaker mode
with short signal tone before conference start
remote microphone in mute
needs IP 200 phone
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