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Mediant 800B

Mediant800 Hybrid E-SBC and Media Gateway Benefits • Fully integrated device for secured SIP trunking and PSTN access • Hybrid SBC and Media Gateway platform lowers CAPEX and reduces space and power footprints • Extensive interoperability and partnerships that extend across multiple vendor devices and protocol implementations • Offers comprehensive security, interoperability and reliability • Delivers high service performance and voice quality • Branch office survivability in the event of a WAN outage Key Features • Rich and powerful SIP normalization and routing mechanisms for seamless interoperability • Hybrid SBC enables seamless migration and PSTN fallback • Support for analog and digital TDM interfaces • Perimeter defense against denial of service, fraud and eavesdropping • VoIP quality monitoring and enforcement • High Availability using two box redundancy • Media Processing for Transcoding, Gain Control, DTMF/Fax, etc. • Optional Open Solution Network (OSN) Platform for hosting value-added applications
Chi tiết Tài liệu Tin khuyến mãi

Specifications
Capacities
Max. Signaling/Media Sessions 250
Max. SRTP/RTP Sessions 180
Max. Transcoding Sessions 45
Max. Registered Users 800
Telephony Interfaces
Analog 4/8/12 FXS ports; 4/8/12 FXO ports
Digital 1/2 span E1/T1; 4/8 BRI ports, network S/T interfaces, NT or TE termination
Clock Source 5 ppm High Precision
Networking 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
VoIP Firewall RTP pinhole management, Rogue RTP detection and prevention, SIP message policy
Encryption and Authentication TLS, 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 OAM interfaces
Intrusion Detection System Detect and mitigate VoIP attacks, prevent Theft of Service and unauthorized access.
Interoperability
SIP B2BUA Full SIP transparency, mature & broadly deployed SIP stack
SIP interworking 3xx redirect, REFER, PRACK, Session Timer, Early media, Call hold, Delayed offer
Registration Registration and authentication on behalf of an IP-PBX
Transport Mediation SIP over UDP to SIP over TCP or SIP over TLS, IPv4 to IPv6, RTP to SRTP, V.34 Fax
Header Manipulation Ability to add/modify/delete headers using advanced regular expressions
URI and Number URI User and Host name manipulations. Ingress & Egress Digit Manipulation
Manipulations
Signal Conversion DTMF/RFC 2833, Inband/T.38 Fax, Packet-time Conversion, V.150.1
NAT Local and Far End NAT traversal for support of remote workers
Hybrid PSTN mode Connect to TDM PBXs or PRI/CAS trunks for least-cost routing or fallback. Also useful for gradual
enterprise migration to SIP, Support for analog, BRI and T1/E1/J1
Transcoding and Vocoders Coder normalization, including transcoding, coder enforcement and re-prioritization. Extensive vocoder
support: Narrowband: SILK, G.711a/mu, G.723.1, G.729A/B, iLBC, AMR, G.726. Wideband: G.722,
AMR-WB and SILK WB
Voice Quality and SLA
Call Admission Control Based on bandwidth, session establishment rate, number of connections/registrations
Packet marking 802.1p/Q VLAN tagging, DiffServ, TOS
Intelligent Voice Multiple queues for granular prioritization of VoIP over other non-real time traffic types, Integrated
Queuing and scheduling schemes (Strict Priority, Class based Prioritization queuing, fairness)
Standalone Survivability Maintain local calls in the event of WAN failure. Outbound calls use PSTN Fallback for external
connectivity (including E911)
Transparent Media Low latency, unprocessed payload transfer
Impairment Mitigation Packet Loss Concealment, Dynamic Programmable Jitter Buffer, Silence Suppression/Comfort Noise
Generation, RTP redundancy, broken connection detection
Voice Enhancement Transrating, RTCP-XR, Acoustic echo cancellation
Media De-anchoring Hair-pinning of local calls to avoid unnecessary media delays and bandwidth consumption
Voice Quality Monitoring AudioCodes Session Experience Manager (SEM)
Redundancy High availability with two box redundancy, active calls preserved
Quality of Experience Access control and media quality enhancements based on QoE and bandwidth utilization
Test agent Ability to remotely verify connectivity, voice quality and SIP message flow between SIP UAs
SIP Routing
Routing Methods Request URL, IP Address, FQDN, ENUM, advanced LDAP
Advanced Routing Criteria QoE, bandwidth, SIP message (SIP request, Coder type etc.)
Redundancy Detect proxy failures and route to alternative proxies
Routing Features Least cost routing, call forking, load balancing
Multiple LANs Support for up to 12 separate LANs
SIPRec IETF standard SIP recording interface
OSN Server Platform (Optional)
Single Chassis Integration Embedded, open Network Solution Platform for third-party services
Memory Up to 16 GB RAM Storage HHD or SSD
Physical / Environmental
Dimensions 1U x 320mm x 345mm (HxWxD) Weight Approx. 5.95lb (2.7kg) loaded with OSN
Mounting Desktop or 19” rack mount Power 100-240V 1.5A 50-60 Hz
Operating Temperature 5°-40° C
Regulatory Compliance
Telecommunications TIA/EIA-IS-968 (FXO, T1) interface, ETSI ES203 021 (FXO interface), TBR-4 (ISDN over E1 interface),
TBR13/13 (E1 lines), TBR-3 (BRI interface)
Safety and EMC IEC60950-1, UL60950-1, FCC Part 15 Class A, EN55022 Class A, EN55024, EN300 386
Environmental Storage ETS300019-2-1 class T1.2
Transportation ETS300019-2-2 class T2.3 Operating ETS300019-2-3

The AudioCodes Mediant 800 Enterprise Session Border
Controller (E-SBC) and Media Gateway offers a complete
connectivity solution for small-to-medium sized enterprises.
The Mediant 800 connects IP-PBXs to any SIP trunking
service provider, scaling up to 250 concurrent SBC sessions.
It offers superior performance in connecting any SIP to SIP
environment, legacy TDM-based PBX systems to IP networks,
and IP-PBXs to the PSTN, supporting up to 60 voice channels
in a 1U platform.
Vast mediation capabilities and proven interoperability
The Mediant 800 supports a wide range of voice coders and
is capable of transcoding between narrowband and wideband
voice coders, providing SIP normalization, fax handling, gain
control and numerous additional media processing features.
It offers certified interoperability with leading unified
communications solutions and SIP trunking providers.
Security
The Mediant 800 provides robust protection for the IP
communications infrastructure, preventing Denial of Service,
fraud and service theft and guarding against cyber-attacks
and other service-impacting events.
Reliability
The Mediant 800 offers active/standby high availability and
maintains high voice quality to deliver reliable enterprise
VoIP communications. Advanced call routing mechanisms,
network voice quality monitoring and branch survivability
capabilities (including PSTN fallback with E911) result in
minimum communications downtime.
Applications
• SIP trunking
• Hosted PBX & UC as a Service
• IP contact centers
• Remote and mobile worker support
• SIP mediation between UC and IP-PBX systems

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Hổ trợ kinh doanh ĐT: 0905710588
Hổ trợ kinh doanh ĐT: 0986883886
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