Capacities

• Max. Signaling: 60
• Max. Registered Users: 200
• Max. RTP/SRTP Sessions: 60

Telephony Interfaces

• Digital: 1-4 BRI ports, network S/T interfaces, NT or TE termination
• Analog: Up to 4 FXS and 4 FXO ports
• Clock Source: 5 ppm High Precision

Network Interfaces

• Ethernet: 4 GE interfaces configured in 1+1 redundancy or as individual ports

Security

• Access Control: DoS/DDoS line rate protection, bandwidth throttling, dynamic blacklisting (Intrusion Detection System)
• VoIP Firewall: RTP pinhole management, rogue RTP detection and prevention, SIP message policy, advanced RTP latching
• Encryption/Authentication: TLS, SRTP, HTTPS, SSH, client/server SIP Digest authentication, RADIUS Digest
• Privacy: Topology hiding, user privacy
• Traffic Separation VLAN/physical interface separation for multiple media, control and OAMP interfaces

Interoperability

• SIP B2BUA: Full SIP transparency, mature and broadly deployed SIP stack, stateful proxy mode
• SIP Interworking: 3xx redirect, REFER, PRACK, session timer, early media, call hold, delayed offer and more
• Registration and Authentication: SIP Registrar, registration on behalf of users/servers, SIP Digest access authentication
• Transport Mediation: Mediation between SIP over UDP/TCP/TLS, IPv4/IPv6, RTP/SRTP (SDES)
• Header Manipulation: Add/modify/delete SIP headers and message body using simple WireShark-like language with powerful capabilities such as variables and utility functions
• Number Manipulations: Ingress and egress digit manipulation
• SIP Interworking: 3xx redirect, REFER, PRACK, session timer, early media, call hold, delayed offer
• Signal Conversion: DTMF/RFC 2833/SIP, T.38 fax, T.38 V3, packet-time conversion
• NAT: Local and far-end NAT traversal for support of remote workers

Voice Quality and SLA

• Call Admission Control: Limit number and rate of concurrent sessions and registers per peer for inbound and outbound directions
• Packet Marking: 802.1p/Q VLAN tagging, DiffServ, TOS
• Standalone Survivability: Maintains local calls in the event of WAN failure
• Voice Monitoring and Enhancement: Transrating, RTCP-XR, acoustic echo cancellation, replacing voice profile due to impairment detection, fixed and dynamic voice gain control, packet loss concealment, dynamic programmable jitter buffer, silence suppression/comfort noise generation, RTP redundancy, broken connection detection
• Image Enhancement BLC/3D DNR
• Direct Media: Hair-pinning (no media anchoring) of local calls to avoid unnecessary media delays and bandwidth consumption
• Test Agent: Ability to remotely verify connectivity, voice quality and SIP message flow between SIP UAs

SIP Routing

• Routing Criteria: Incoming SIP trunk, DID ranges, host names, any SIP headers, codecs, QoE, bandwidth
• Querying External Databases: Routing based on customized queries of ENUM, LDAP, HTTP server (REST API)
• Route To: Configured SIP peers, registered users, IP address, request URI
• Advanced Routing Features: Alternative routes, load balancing, least-cost routing, call forking, E911 emergency call detection and prioritization
• SIPREC: IETF standard SIP recording interface

Management

• OAM&P: Browser-based GUI, CLI, SNMP, INI Configuration file, REST API, EMS

Physical/ Environmental

• Dimensions: 51 x 296 x 160 mm (2 x 11.65 x 6.3 in.) (HxWxD)
• Weight: 670g
• Mounting: Desktop
• Power: Single universal AC power supply 100-240V, 50-60 Hz, 12V/3A or 12V/5A
• Environmental: Operational: 5 to 40° C (41 to 104°F); Storage: -25 to 85°C (-13 to 185°F) Relative Humidity: 10 to 90% non-condensing