Thiết bị SBC Mediant 1000 Gateway

Thiết bị SBC Mediant 1000 Gateway và bộ điều khiển E-SBC được thiết kế để cung cấp giải pháp kết nối hoàn chỉnh cho các địa điểm doanh nghiệp vừa và nhỏ.

  • Max. Signaling: 150
  • Max. RTP/SRTP Sessions: 120
  • Max. Transcoding Sessions: 96
  • Max. Registered Users: 600
AvailabilityIn stock
SKU
Mediant 1000
Thương hiệu:
MODEL Mediant 1000
Capacities
Max. Signaling 150
Max. Transcoding Sessions 96
Max. RTP/SRTP Sessions 120
Max. Registered Users 600
Telephony Interfaces
Modularity and Capacity 6 slots for hosting voice processing and PSTN termination modules (up to 192 channels)
Digital Module Up to 6 E1 or 8 T1/J1 spans provided on trunk modules. Each module supports 1, 2, or 4 E1/T1/J1 spans, with an option of PSTN fallback
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
BRI Module Up to 20 BRI ports provided on BRI modules. Each module supports 4 BRI ports, with PSTN fallback. Providing S/T interfaces; NT or TE termination; 2W per port (power supplied)
Analog Module Up to 24 FXS interfaces, provided on 4-port FXS modules, ground/loop start
Up to 24 FXO interfaces, provided on 4 port-FXO modules, ground/loop start
Media Processing Module Up to 4 Media Processing modules (MPM), providing additional DSP resources
Network Interfaces
Ethernet Up to 6 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 Automatic 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
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, G.722, G.727, iLBC, QCELP, GSM EFR
Signal Conversion DTMF/RFC 2833/SIP, T.38 fax, V.34, 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. 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
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)
Advanced Features Alternative destinations, load balancing, LCR, call forking, E911 emergency call detection and prioritization
Available Destinations Configured SIP peers, registered users, IP address, request URI
SBC Media Types Audio\Video\Fax\Text\Message Session Relay Protocol (MSRP)\Binary Floor Control Protocol (BFCP)
Management
OAM&P Browser-based GUI, CLI, SNMP, INI Configuration file, REST API, One Voice Operations Center (OVOC)
OSN Server Platform (Optional)
Single Chassis Integration Optional embedded, x86, Intel-based Open Solution Network platform for third-party applications
More Information
Thương Hiệu Audiocodes
SKU Mediant 1000
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