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Mediant 1000B

Mediant1000 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 • Scalable “pay-as-you-grow” modular architecture • 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 • Modular support for analog and digital TDM interfaces • Perimeter defense against denial of service, fraud and eavesdropping • VoIP quality monitoring and enforcement • Media Processing for Transcoding, Gain Control, DTMF/Fax, etc. • Optional Open Solution Network (OSN) server module for hosting value-added applications
Chi tiết Tài liệu Tin khuyến mãi


Max. Signaling/Media Sessions           150 Max.
Transcoding Sessions                        96
Max. SRTP/RTP 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
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/FXO interfaces, provided on 4 ports FXO / FXS 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
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/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
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 Manipulations          URI User and Host name manipulations. Ingress & Egress Digit Manipulation
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
Transcoding and Vocoders               Coder normalization, including transcoding, coder enforcement and re-prioritization. Extensive
                                                   vocoder support: NarrowBand: G.711a/mu, G.723.1, G.729A/B, iLBC, EVRC, AMR, G.726,
                                                   G.727, GSM FR, MS GSM, GSM EFR, QCELP. Wideband: G.722                                  
Media Processing                            60 Conferencing legs (3 way or N-way), play, record to IP or PSTN using Netann (RFC4240),
                                                   MSCML (RFC5022)
Voice Quality and SLA
Call Admission Control                     Based on bandwidth, session establishment rate, number of connections/registrations
Standalone Survivability                  Maintain local calls in the event of WAN failure
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
Media De-anchoring                       Hair-pinning of local calls to avoid unnecessary media delays and bandwidth consumption
Voice Quality Monitoring                 AudioCodes Session Experience Manager (SEM)
Test agent Remote                      verification of 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)
Routing Features                           Least cost routing, call forking, load balancing
OSN Server Platform
Single Chassis Integration Embedded, Open Network Solution Platform for third-party services
Memory                                      Up to 8GB RAM Storage HHD or SSD
Hardware Specifications
Dimensions                                 1U x 320mm x 345mm (HxWxD)
Weight Approx.                          5.95lb (2.7kg) installed with OSN
Mounting                                   Desktop or 19” mount
Power                                       Single power supply 100-240V, 50-60 Hz, 1.5A max. optional redundant power supply
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
Regulatory Compliance
Telecommunication Standards      TIA/EIA-IS-968, TBR-4, TBR-13, and TBR-21
Safety and EMC Standards           UL60950-1; FCC 47 CFR part 15 Class B
                                               CE Mark (EN55022 Class B, EN60950-1, EN55024, EN300 386, EN61000-3-2/3-3)
Environmental Specifications         ETS 300019-2-1 Storage T1.2, ETS 300019-2-2 Transportation T2.3, ETS 300019-2-3 Operating T3.2

The AudioCodes Mediant 1000 Enterprise Session Border
Controller (E-SBC) and Media Gateway offers a complete
connectivity solution for small-to-medium sized enterprises.
The Mediant 1000 connects IP-PBXs to any SIP trunking
service provider, scaling up to 150 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 192 voice channels
in a modular 1U platform.
Vast mediation capabilities and proven interoperability
The Mediant 1000 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.
The Mediant 1000 provides robust protection for IP
communications infrastructure, preventing Denial of Service,
fraud and service theft and guarding against cyber-attacks
and other service-impacting events.
The Mediant 1000 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.
• 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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