Benefits
Key features
Max. Signaling/Media Sessions 60
Max. SRTP/RTP Sessions 60
Max. Registered Users 200
Telephony Interfaces
Analog Up to 4 FXS ports
1-4 BRI ports, network S/T interfaces , NT or TE termination
Digital
Clock source 5 ppm High Precision
Network Interfaces
Ethernet 4GE interfaces configured in 1+1 redundancy or as individual ports
Security
Access Control
DoS/DDoS line rate protection
bandwidth throttling
dynamic blacklisting
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
Intrusion Detection Detection and prevention of VoIP attacks
theft of service and unauthorized access
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
Registration and Authentication
User registration restriction control
registration and authentication on behalf of users
SIP authentication server for SBC users
Transport Mediation SIP over UDP/TCP/TLS, IPv4/IPv6, RTP/SRTP (SDES)
Message Manipulation Ability to add/modify/delete SIP headers and message body using advanced regular expressions (regex)
URI and Number Manipulations URI user and host name manipulations, ingress and egress digit manipulation
Vocoders Coder normalization including coder enforcement and re-prioritization, extensive vocoder support: G.711, G.723.1, G.726, G.729,
GSM-FR, AMR-NB, AMR-/WB (G722.2), SILK-NB/WB, Opus-NB/WB
Signal Conversion DTMF/RFC 2833/SIP, T.38 fax, packet-time conversion
NAT Local and far-end NAT traversal for support of remote workers
Voice Quality and SLA
Call Admission Control Based on bandwidth, session establishment rate, number of connections/registrations
Packet marking 802.1p/Q VLAN tagging, DiffServ, TOS
Standalone Survivability Maintains local calls in the event of WAN failure
Impairment Mitigation Packet Loss Concealment, Dynamic Programmable Jitter Buffer, Silence Suppression/Comfort, Noise Generation,
RTP redundancy, broken connection detection
Voice Enhancement Transrating, RTCP-XR, Acoustic echo cancellation, replacing voice profile due to impairment detection, Fixed & dynamic voice gain control
Direct Media (No Media Anchoring) Hair-pinning of local calls to avoid unnecessary media delays and bandwidth consumption
Voice Quality Monitoring RTCP-XR
Quality of Experience Access control and media quality enhancements based on QoE and bandwidth utilization
Test agent Ability to remotely verify connectivity, voice quality and SIP message flow between SIP UAs
SIP Routing
Routing Methods Request URL, IP address, FQDN, ENUM, advanced LDAP, third-party routing control through REST API
Advanced Routing Criteria QoE, bandwidth, SIP message (SIP request, coder type, etc.), Layer-3 parameters
Routing Features Least-cost routing, call forking, load balancing, E911 gateway support, 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 670 g
Power Single universal AC power suppy 100-240V, 50-60 Hz, 12V/3A or 12V/5A
Mounting Desktop
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
1. ทางบริษัทฯ ขอสงวนสิทธิ์ในการรับผิดชอบใดๆ อันเนื่องจากข้อผิดพลาดทางการพิมพ์ (ผิด/ตก)
2. ทางบริษัทฯ ขอสงวนสิทธิ์ในการเปลี่ยนแปลงราคาและโปรโมชั่นโดยไม่แจ้งให้ทราบล่วงหน้า
กลุ่ม
ระบบตู้สาขาโทรศัพท์
แบรนด์
NEC
รุ่น
BX500
สินค้าของแท้ มีประกัน
100%
มีสินค้าในสต็อก
พร้อมส่งทันที