AudioCodes Mediant 800 Enterprise Session Border Controller and Media Gateway (M800B-ESBC)
![AudioCodes Mediant 800 Enterprise Session Border Controller and Media Gateway (M800B-ESBC)]()
![AudioCodes Mediant 800 Enterprise Session Border Controller and Media Gateway (M800B-ESBC)]()
AudioCodes Mediant 800 Enterprise Session Border Controller and Media Gateway (M800B-ESBC)
Available:In Stock
£711.48
AudioCodes Mediant 800 Enterprise Session Border Controller and Media Gateway (M800B-ESBC)
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 M800B-ESBC offers a complete connectivity solution for small-to-medium sized enterprises.
AudioCodes M800B-ESBC Key Features
- 400 SBC Sessions
- 124 TDM Sessions
- 1+1 High Availability
- Supports OPUS and SILK
- 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 400 concurrent sessions, the Mediant 800 connects IP-PBXs to any SIP trunking service provider and offers superior performance in connecting any SIP to SIP environment.
In addition, the Mediant 800 supports up to 124 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 M800B-ESBC Technical Specifications
Capacities
- Max Signalling: 250
- Max. RTP/STRP Sessions: 250/250
- Max. Transcoding Sessions: 57
- Max. Registered Users: 1500
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, LucentTM 4/5ESS, NortelTM DMS- 100 and others. Different CAS protocols, including MFC R2,
Network Interfaces
- Ethernet 4 GE or 4 GE + 8 FE 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/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 normalisation including transcoding, coder enforcement and re-prioritization, extensive vocoder support: G.711, G.723.1, G.726, G.729, GSM-FR, AMR-NB, AMR-WB (G.722.2), SILK-NB/WB, Opus-NB/WB
- 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 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
- SIPREC IETF standard SIP recording interface
Management
- OAM&P: Browser-based GUI, CLI, SNMP, INI Configuration file, REST API, EMS
Physical/Environmental
- Dimensions: 1U x 320mm x 345mm (HxWxD)
- Weight: Approx. 5.95lb (2.7kg) loaded with OSN
- Mounting: Desktop or 19” rack mount
- Operating Temperature: 5°-40° C
- Power: Internal AC power supply rated: 100-240V 4A 50- 60 Hz
OSN Server Platform (Optional)
- Single Chassis Integration: Optional embedded, x86, Intel-based Open Solution Network platform for third-party applications
AudioCodes Mediant 800 Enterprise Session Border Controller and Media Gateway (M800B-ESBC)
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 M800B-ESBC offers a complete connectivity solution for small-to-medium sized enterprises.
AudioCodes M800B-ESBC Key Features
- 400 SBC Sessions
- 124 TDM Sessions
- 1+1 High Availability
- Supports OPUS and SILK
- 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 400 concurrent sessions, the Mediant 800 connects IP-PBXs to any SIP trunking service provider and offers superior performance in connecting any SIP to SIP environment.
In addition, the Mediant 800 supports up to 124 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: 250
- Max. RTP/STRP Sessions: 250/250
- Max. Transcoding Sessions: 57
- Max. Registered Users: 1500
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, LucentTM 4/5ESS, NortelTM DMS- 100 and others. Different CAS protocols, including MFC R2,
Network Interfaces
- Ethernet 4 GE or 4 GE + 8 FE 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/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 normalisation including transcoding, coder enforcement and re-prioritization, extensive vocoder support: G.711, G.723.1, G.726, G.729, GSM-FR, AMR-NB, AMR-WB (G.722.2), SILK-NB/WB, Opus-NB/WB
- 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 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
- SIPREC IETF standard SIP recording interface
Management
- OAM&P: Browser-based GUI, CLI, SNMP, INI Configuration file, REST API, EMS
Physical/Environmental
- Dimensions: 1U x 320mm x 345mm (HxWxD)
- Weight: Approx. 5.95lb (2.7kg) loaded with OSN
- Mounting: Desktop or 19” rack mount
- Operating Temperature: 5°-40° C
- Power: Internal AC power supply rated: 100-240V 4A 50- 60 Hz
OSN Server Platform (Optional)
- Single Chassis Integration: Optional embedded, x86, Intel-based Open Solution Network platform for third-party applications
Related items
£488.76
