Thiết Bị SBC Mediant 800C

Thiết bị SBC Mediant 800 kết nối các tổng đài IP với các nhà cung cấp dịch vụ SIP Trunking, với 400 phiên đồng thời. Nó cung cấp hiệu suất vượt trội trong việc kết nối mọi môi trường SIP to SIP, các hệ thống tổng đài dựa trên TDM kế thừa với các mạng IP và tổng đài IP với PSTN.

  • Max. Signaling: 400
  • Max. RTP/SRTP Sessions: 400/300
  • Max. Transcoding Sessions: 114
  • Max. Registered Users: 2000
AvailabilityIn stock
Mediant 800C
Thương hiệu:
MODEL Mediant 800C
Max. Signaling 400
Max. RTP/SRTP Sessions 400/300
Max. Transcoding Sessions 114
Max. Registered Users 2000
Telephony Interfaces
Analog 4/8/12 FXS ports; 4/8/12 FXO ports
Digital Up to 4 E1/T1 interfaces; 4/8 BRI Ports
Clock Source 5 ppm High Precision
Digital PSTN Protocols Various ISDN PRI protocols such as EuroISDN, North American NI-2, Lucent™ 4/5ESS, Nortel™ DMS- 100 and others. Different CAS protocols, including MFC R2, E&M immediate start, E&M delay dial/start and others.
Network Interfaces
Ethernet 4 GE or 4 GE + 8 FE interfaces configured in 1+1 redundancy or as individual ports
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, DTLS, SRTP, HTTPS, SSH, client/server SIP Digest authentication, RADIUS Digest
Privacy Automatic topology hiding, user privacy
Traffic Separation VLAN/physical interface separation for multiple media, control and OAMP interfaces
Intrusion Detection System Detection and prevention of VoIP attacks, theft of service and unauthorized access
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/WebSocket, IPv4/IPv6, RTP/SRTP (SDES/DTLS)
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
Transcoding and Vocoders Coder normalization including transcoding, coder enforcement and re-prioritization, extensive vocoder support: G.711, G.723.1, G.726, G.729A/B, GSM-FR, AMR-NB, AMR-WB (G.722.2), SILK-NB/WB, Opus-NB/WB, iLBC
Signal Conversion DTMF/RFC 2833/SIP, T.38 fax, T.38 V3, V.34, packet-time conversion, V.150.1
WebRTC Gateway Interworking between WebRTC endpoints and SIP networks. Supports WebSocket, Opus, VP8 video coder, lite ICE, DTLS, RTP multiplexing.
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. Outbound calls can use PSTN fallback (including E911).
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
Direct Media Hair-pinning (no media anchoring) of local calls to avoid unnecessary media delays and bandwidth consumption
High Availability SBC high availability with two-box redundancy, active calls preserved
Test Agent Ability to remotely verify connectivity, voice quality and SIP message flow between SIP UAs
SIP Call Handling
Criteria Incoming SIP trunk, DID ranges, host names, any SIP headers, codecs, QoE, bandwidth
Querying External Databases Destinations based on customized queries of ENUM, LDAP, HTTP server (REST API)
Available Destinations Configured SIP peers, registered users, IP address, request URI
Advanced Features Alternative destinations, load balancing, LCR, call forking, E911 emergency call detection and prioritization
SBC Media Types Audio\Video\Fax\Text\Message Session Relay Protocol (MSRP)\Binary Floor Control Protocol (BFCP)
SIPREC IETF standard SIP recording interface, supporting both audio and video SBC sessions
OAM&P Browser-based GUI, CLI, SNMP, INI Configuration file, REST API, One Voice Operations Center (OVOC)
More Information
Thương Hiệu Audiocodes
SKU Mediant 800C
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