Innovaphone IP110 User manual

Pure IP-Telephony
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Errors and omissions excepted. All information, technical specifications, and delivery times may be subject to change.
ts_IP110_en.doc, Issue: August 2008
Copyright © 2004-2008 innovaphone®AG
Technical Specification
IP phone IP110
Interfaces
Ethernet:2 x RJ 45 interface
10/100-BASE-TX (auto negotiation)
with internal 2 port switch
and Power over LAN, Class 1 (IEEE 802.3af)
Hardware
Power supply: Power over LAN, Class 1 or
Mains adapter
primary: 110-240 V, 50 Hz, 45mA,
secondary: 12V DC 800mA,
Memory: 8 MB DRAM
4 MB Flash
Operation environment: Operation temperature: 0°C to +45°C,
Humidity: 10% to 90% non-condensing
Storage temperature: -10°C to +70°C
Display: 128 x 64 Pixel - or - 7 lines with 21 characters
Keyboards: numeric,
2 arrow keys,
4 free programmable function keys
volume and free speech keys
Protocols
DHCP dynamic host configuration protocol - IP interfaces settings
ICMP Internet control message protocol - for Ping tests
SNTP simple network time protocol – to receive date and time values
LDAP lightweight directory access protocol – access for LDAP compatible database
PPPoE PPP over Ethernet – direct access to DSL modem
PPTP point-to-point tunneling protocol – to setup VPN tunnels to the corporate network
MPPE Microsoft point-to-point encryption - encryption in PPTP
NAT network address translation – to translate official IP addresses in unofficial addresses and vice versa
(inclusive SIP und H.323)
HTTP for configuration sing Web browser
SNMP sending traps to the management software

Pure IP-Telephony
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Voice over IP
H.323: H.323 version 5, H.245 fast connect, Enblock dialing, Overlapped sending
RAS Gatekeeper routed signaling (support for external Gatekeeper)
Security (encrypted password authentication H.235)
Supplementary Services: H.450.1
H.450.2 Call transfer
H.450.3 Call diversion
H.450.4 Call hold
H.450.5 Call Pick-up
H.450.6 Call waiting
H.450.7 Message Waiting Indication
H.450.8 Name identification
H.450.9 Call Completion busy (CCBS) and Call Completion no Reply (CCNR)
DTMF: H.245 “Alphanumeric” oder “Signal Type”
SIP: SIP version 2 (including HTTP digest authentication) conform RFC 3261
SIP over UDP, TCP, TLS (SIPS, V 7.0 or higher)
SIP services: MWI (rfc3842/rfc3265 "Subscription für message-summary")
DTMF (rfc2833 "RTP payload for DTMF")
Name Identification (Display String, rfc3325 "Asserted Identity")
Hold/Retrieve (rfc3264 "Offer/Answer Model for SDP")
Transfer (rfc3515 "REFER Method", rfc3891 "Replaces Header)
Coder Change, T.38 (rfc3264 "Re-Negotiation")
Call Forwarding (PBX internal, "183 Call Is Being Forwarded", "Diversion Header")
Overlap Dialing (rfc3578)
Dialoge State monitoring, partner Key (rfc 4235)
Instant Messaging (rfc 3428)
supported SIP-RFCs: RFC 1889 RTP: Real-Time Transport Protocol
RFC 2327 SDP: Session Description Protocol
RFC 2396 Uniform Resource Identifiers (URI): Generic Syntax
RFC 2543 SIP: Session Initiation Protocol (obsolete)
RFC 2616 Hypertext Transfer protocol (HTTP/1.1)
RFC 2617 HTTP Authentication: Basic and Digest Access Authentication
RFC 2782 A DNS RR for specifying the location of services (DNS SRV)
RFC 2976 The SIP INFO Method
RFC 3261 SIP: Session Initiation Protocol
RFC 3263 Session Initiation Protocol (SIP): Locating SIP Servers
RFC 3264 An Offer/Answer Model with the Session Description Protocol (SDP)
RFC 3265 SIP-Specific Event Notification
RFC 3326 The Reason Header Field for the Session Initiation Protocol
RFC 3389 RTP Payload for Comfort Noise
RFC 3515 The Session Initiation Protocol (SIP) Refer Method
RFC 3550 RTP: Transport Protocol for Real-Time Applications
RFC 3551 RTP Profile for A/V Conferences with Minimal Control
RFC 3555 MIME Type Registration of RTP Payload Formats
RFC 3578 Mapping of ISDN User Part (ISUP) Overlap Signalling to the SIP
RFC 3680 SIP Event Package for Registrations
RFC 3764 enumservice registration for SIP Adresses-of-Record
RFC 3824 Using E.164 numbers with SIP
RFC 3891 The Session Initiation Protocol ‘Replaces Header’
RFC 3892 The SIP Referred-By Mechanism -SIP-aware filtering (to prevent SIP attacks)
RFC 3842 SIP Message Waiting
RFC 3311 re-INVITE
RFC 2833 DTMF via RTP Channel , RTP payload for DTMF
RFC 3325 Name identification
RFC 3578 Overlap Dialing
RFC 3420 Internet Media Type message/sipfrag

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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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Call Diversion break through partner function, authorized subscriber can call an other subscriber with active call diversion
Busy lamp field partner function, shows the actual status of the partner phone
related key supports pickup when ringing and speed dial when free
needs IP 200 phone
partner key: partner status display „free“, „Ringing“ or“ busy“
speed dial in status „free“
pickup in status „Ringing“
DTMF DTMF tone detection & ringing
Security: password-protected configuration
PIN protected phone access
Ring tones: general MIDI compatible synthesizer / interpreter (RTTTL)
24 sequences downloadable
selective ring tones for internal, external and special phone connections specified in phone book
Call lists: last 100 outgoing calls,
last 100 incoming calls,
combined list for incoming and outgoing calls
information about date, time and success of the call
Multiple registrations: up to 6 subscribers can be registered to one phone
Multiple registrations of the same subscriber on different phones supported (supported by inno-
vaphone PBX)
Free speech: free speech in on hook mode
Mute: switch off the microphone
Phone book
internal: for private entities, only for this phone
notes and special ring tone for all phonebook entries
global: automatic access to all PBX subscribers
external: import from LDAP compatible database
Usability: search separate or global in all phone books
Character wise resolution during name entering
Name resolution of incoming phone numbers
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