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Mediant 2600 -ESBC

Mediant2600 Hybrid SBC and Media Gateway Benefits • Pure-IP SBC for medium-sized enterprise deployments • Offers comprehensive security, interoperability and reliability • Delivers high service performance and voice quality • Flexible licensing options for cost-effective scalability Key Features • Scalable to 600 SBC sessions • Extensive SIP mediation capabilities • Supports remote workers and mobile SIP clients • Perimeter defense against denial of service, fraud and eavesdropping • VoIP quality monitoring and enforcement • Branch survivability during WAN failure • Active/Standby High Availability • Advanced media handling including transcoding and wideband speech
Chi tiết Tài liệu Tin khuyến mãi

Capacities
Max. Signaling/Media Sessions 600
Max. SRTP/RTP Sessions 600
Max. Transcoding Sessions 600
Max. Registered Users 8,000
Networking Interfaces
Ethernet 8 Redundant 100/1000 Base-T Ethernet ports for physical separation between multiple LAN and WAN
between Media, Control and OA&M
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 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
Transcoding and Vocoders Coder normalization including: transcoding, coder enforcement and re-prioritization
Exetnsive vocoder support: G.711, G.723.1, G.726, G.729, GSM-FR, AMR-NB, AMR-WB, SILK-NB, SILKWB,
OPUS1
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
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
Transparent Media Low latency, unprocessed payload transfer (voice and video supported)
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
Gain Control Fixed & dynamic voice gain control
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 48 separate LANs
SIPRec IETF standard SIP recording interface
Physical / Environmental
Dimensions 1U x 444mm x 355mm (HxWxD)
Weight Approx. 11.7 lbs (5.3Kg)
Mounting Desktop or 19” rack mount
Power 100–240 V AC redundant dual feed
Operating Temperature 5°-40° C
Regulatory Compliance
Safety and EMC UL60950-1
FCC Part 15 Class A
ICES-003 Class A
CE makring: IEC60950-1, EN55024, EN55022 Class A, EN61000-3-2, EN61000-3-3, ETSI EN300 386

The AudioCodes Mediant 2600 Session Border Controller
(SBC) is a mid-range capacity member of AudioCodes’ fieldproven
hardware-based SBC product family, designed to offer
enterprises a reliable and scalable SBC solution. The Mediant
2600 SBC supports wide-ranging SIP interoperability,
delivering service assurance and enabling scalable, reliable
and secured connectivity between different VoIP networks.
The Mediant 2600 SBC provides a perfect solution for
enterprises and large organizations such as contact centers,
where security, reliability and high performance are critical.
Extensive Mediation Capabilities and Proven
Interoperability
The Mediant 2600 SBC includes comprehensive media
security and SIP normalization capabilities. It offers full
interoperability with an extensive list of IP-PBXs, unified
communications solutions and SIP trunking provider networks.
Security
The Mediant 2600 SBC provides robust protection for the IP
communications infrastructure, preventing fraud and service
theft and guarding against cyber-attacks and other serviceimpacting
events.
Reliability
The Mediant 2600 SBC 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 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
• Residential VoIP

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