AudioCodes Mediant 1000 Enterprise Session Border Controller and Media Gateway (M1KB)
![AudioCodes Mediant 1000 Enterprise Session Border Controller and Media Gateway (M1KB) AudioCodes Session Border Controllers VoIP Supplies]()
![AudioCodes Mediant 1000 Enterprise Session Border Controller and Media Gateway (M1KB) AudioCodes Session Border Controllers VoIP Supplies]()
AudioCodes Mediant 1000 Enterprise Session Border Controller and Media Gateway (M1KB)
Available:In Stock
- Category:ALL PRODUCTS,AudioCodes Session Border Controllers,Best selling products,Intel: Revolutionizing Computing Technology,New products
£1,592.91
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 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.
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
Understanding Our Returns and Refunds Policy
Easy Returns for Your Convenience
At our company, we believe that shopping should be hassle-free. This is why we have made our returns and refunds process as easy as possible. If you receive a product that doesn’t meet your expectations or is defective, you can return it within 30 days for a full refund. Our team is always here to assist you, so you don’t have to worry about complicated steps when returning an item.
Refund Process Made Simple
Once we receive your returned item, we initiate the refund process immediately. Typically, refunds are processed within 5-7 business days. However, there might be variations due to your bank's policies. We understand that getting your money back promptly is essential, which is why we prioritize quick refunds. Because of this efficiency, customers can trust that their satisfaction is our highest priority.
Customer Support for Returns and Refunds
If you have questions about our returns and refunds, our customer support team is always ready to help. You can reach out via email or phone for assistance with your return. We want you to feel confident when making a purchase, and knowing that we value customer support makes the process smoother. So, don’t hesitate to contact us if you need clarification or help!
