Call Control Features
SIP inbound and outbound call support
SIP call redirection (SIP 302 status)
SIP call rejection
SIP call routing and transfers
SIP call leg bridging
SIP REINVITE audio re-routing
SIP REFER support
SIP registrar server support
SIP proxy server support
SIP authentication support
Configurable SIP port range
NAT IP translation support
Multi-ethernet card bridging support
Runs on standard x86 platforms
Media and IVR Features
Audio prompt/announcement playback
Audio bridging
Audio mixing/conferencing
Audio noise removal
Audio fixed gain control
Audio dynamic gain control
Audio voice activity detection (VAD)
Audio call recording
Audio codec transcoding
Audio speech recognition (ASR)
Audio speech synthesis (TTS)
Audio playout jitter buffer
DTMF tone detection
DTMF tone generation
Configurable RTP port range
NAT IP translation support
Multi-ethernet card bridging support
Supports VoiceXML 2.x IVR
Supports CallXML 3.0 IVR
Runs on standard x86 platforms
SIP Appliance Hardware Features
Highly reliable x86 platform
120/240v AC
19" full-depth rack mount server
Up to 40 servers in a 4 post rack
Quad Core 2.33 GHz, 4GB RAM
1U w/ 1 PCI slot, 3U w/ 5 PCI slots
SIP Appliance Built-in PSTN/SIP Gateway
Optional add-on to SIP Appliance
Supports analog POTS lines
Supports T1 and E1 trunks/lines
NI1/2, QSIQ, and Euro ISDN
DMS100/2500 and 4/5ESS ISDN
CAS robbed bit signaling
Echo cancel with 32 to 128 MS tail
Standard size PCI 3.3v card
Up to 96 ports in 1U server
Over 4,000 ports per 4 post rack
|
|
Compatability Features
Works with Level(3) SIP
Works with BroadVoice SIP
Works with Delta3 SIP
Works with SER/OpenSER SIP
Works with Asterisk SIP
Works with Cisco SIP
Works with Lucent TNT SIP
Works with Sipura SIP
Works with Sonus SIP
Works with most other SIP services
Works with most other SIP devices
Other Features
Premise or hosted solution
No single points of failure
Scales from one to thousands of ports
First platform with XML call control
First platform with XML conferencing
First shipping CCXML implementation
First SIP/VoIP IVR platform
IETF Standards Support
RFC 3261 SIP
RFC 3310 SIP Authentication
RFC 1889 RTP Media
RFC 1890 RTP Audio
RFC 2327 SDP
RFC 3264 SDP negotiation
RFC 2883 DTMF and events
RFC 3263 SRV DNS records
RFC 3761 ENUM URI DNS records
RFC 3764 ENUM SIP DNS records
RFC 3164 UDP Syslog logging
RFC 3195 TCP Syslog logging
RFC 2865 RADIUS metering
RFC 2616 HTTP protocol
RFC 2617 HTTP authentication
RFC 2964 HTTP state management
RFC 2965 HTTP state management
RFC 3927 Dynamic IP config
RFC 2136 Dynamic DNS updates
DNSEXT DNS Service Discovery Draft
MMUSIC RTSP Draft
SPEECHSC MRCP Draft
|
|
W3C Standards Support
VoiceXML 2.0 speech/IVR media
VoiceXML 2.1 extensions
CCXML 1.0 call control
SRGS 1.0 speech grammars
SSML 1.0 speech markup
SISR speech semantic interpretation
Extensible Markup Language (XML) 1.1
Namespaces in XML 1.1
XML Document Object Model
XML Path Language (XPath) 1.0
XML Event Syntax
SOAP Web Services
WSDL Web Service Description
ANSI Standards Support
T1/DS1 Electrical Interface
T1/DS1 Robbed Bit Signaling
T1/DS1 Primary Rate ISDN
T3/DS3 Electrical InterfaceBearer Services for ISDN PRI
SS7 SCCP/MTP/IUP/TCAP
Telecom Voltage Levels
Electrical protection for CO's
ITU Standards Support
G.711 uLaw Audio Codec
G.712 aLaw Audio Codec
G.726 ADPCM Audio Codec
G.729 CS-ACELP Audio Codec
G.729 (b) Silence Detection
GSM RPE-LTP Audio Codec
|
