AudioCodes Mediant 1000 Enterprise Session Border Controller and Media Gateway (M1KB)
Review: 5 - "A masterpiece of literature" by , written on May 4, 20020
Great
Categories

AudioCodes Mediant 1000 Enterprise Session Border Controller and Media Gateway (M1KB)

Available:In Stock
£1,473.72
Image

AudioCodes Mediant 1000 Enterprise Session Border Controller and Media Gateway (M1KB)

Please Note: Additions including sessions, remote implementation support and licenses/ software are available. If you require any of these additions, please enquire using the button above.

The AudioCodes M1KB offers a complete connectivity solution for small-to-medium sized enterprises.

AudioCodes M1KB Key Features

  • 150 SBC Sessions
  • 192 TDM Sessions
  • Modular
  • Extensive Vocoder Support
  • Comprehensive interoperability, proven interoperability with SIP trunks, SIP platforms and IP cloud services
  • Hybrid functionality, true hybrid SBC and gateway platform for gradual migration, low CAPEX and reduced space and power footprints
  • Enhanced security, robust perimeter defense against cyber, DoS and DDoS attacks, as well as eavesdropping, fraud and service theft
  • Superior voice quality, advanced capabilities for optimising and monitoring voice service quality
  • High resiliency, high availability using 1+1 redundancy, local branch survivability and PSTN fallback

Scaling up to 150 concurrent sessions, the Mediant 1000 connects IP-PBXs to any SIP trunking service provider and offers superior performance in connecting any SIP to SIP environment.


In addition, the Mediant 1000 supports up to 192 voice channels in a 1U platform to enable versatile connectivity between TDM and VoIP networks, such as connecting legacy TDM PBX systems to IP networks and IP-PBXs to the PSTN.

AudioCodes M1KB Technical Specifications

Capacities

  • Max Signalling: 150
  • Max. RTP/STRP Sessions: 120
  • Max. Transcoding Sessions: 96
  • 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, LucentTM 4/5ESS, NortelTM 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, DTLS, 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 System: 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 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 normalisation including transcoding, coder enforcement and re-prioritization, extensive vocoder support: G.711, G.723.1, G.726, G.729, GSM-FR, AMR-NB, 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 Routing

  • Routing Criteria: Incoming SIP trunk, DID ranges, host names, any SIP headers, codecs, QoE, bandwidth
  • Querying External Databases: Routing based on customised 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 prioritisation

Management

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

Physical/Environmental

  • Dimensions: 1U x 444 x 355 mm (HxWxD)
  • Weight: Approx. 9.7lb (4.4kg)
  • Mounting: Desktop or 19” rack mount
  • Power: Dual power supply 100-240V, 50-60 Hz, 1.5A max
  • Environmental: Operational: 0 to 40° C (32 to 104°F); Storage: -20 to 70°C (-4 to 158°F), Relative Humidity: 10 to 85% non-condensing

OSN Server Platform (Optional)

  • Single Chassis Integration: Optional embedded, x86, Intel-based Open Solution Network platform for third-party applications

AudioCodes Mediant 1000 Enterprise Session Border Controller and Media Gateway (M1KB)

Please Note: Additions including sessions, remote implementation support and licenses/ software are available. If you require any of these additions, please enquire using the button above.

The AudioCodes M1KB offers a complete connectivity solution for small-to-medium sized enterprises.

AudioCodes M1KB Key Features

  • 150 SBC Sessions
  • 192 TDM Sessions
  • Modular
  • Extensive Vocoder Support
  • Comprehensive interoperability, proven interoperability with SIP trunks, SIP platforms and IP cloud services
  • Hybrid functionality, true hybrid SBC and gateway platform for gradual migration, low CAPEX and reduced space and power footprints
  • Enhanced security, robust perimeter defense against cyber, DoS and DDoS attacks, as well as eavesdropping, fraud and service theft
  • Superior voice quality, advanced capabilities for optimising and monitoring voice service quality
  • High resiliency, high availability using 1+1 redu