Mediant 500 SBC

Mediant 500 SBC
AudioCodes Session Border Controller (SBC) Products
Mediant 500 Session Border Controller
• A highly integrated device for secured SIP
Trunking and PSTN access, forming a single and
managed point of demarcation for VoIP networks
• Compact, high performance VoIP connectivity
device for small enterprises and branch offices
• Extensive interoperability and partnerships that
extend across multiple vendor devices and
protocol implementations
The AudioCodes Mediant 500 Enterprise Session Border Controller
(E-SBC) is a compact, high performance VoIP connectivity solution
for small enterprises and branch office locations. The Mediant 500
connects IP-PBXs and unified communications platforms 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 a single E1/T1 interface with 30 voice channels in
a 1U platform. It also ensures secure and reliable communications for
branch offices in distributed enterprise communications deployments.
Vast mediation capabilities and proven interoperability
The Mediant 500 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.
• Offers comprehensive security and reliability
• Delivers high service performance and voice
The Mediant 500 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
• Branch office survivability in the event of a WAN
Key Features
• Rich and powerful SIP normalization and routing
mechanisms for seamless interoperability
• Support for E1/T1 digital TDM interface
The Mediant 500 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.
• Supports remote workers and mobile SIP clients
• Perimeter defense against denial of service,
fraud and eavesdropping
• VoIP quality monitoring and enforcement
• High Availability using two box redundancy
• SIP trunking
• Hosted PBX & UC as a Service
• IP contact centers
• Remote and mobile worker support
• SIP mediation between UC and IP-PBX systems
AudioCodes Session Border Controller (SBC) Products
Mediant 500
About AudioCodes
Max. Signaling/Media Sessions
Max. Registered Users
Max. SRTP/RTP Sessions
Telephony Interfaces
Single E1/T1 interface
Clock Source
5 ppm High Precision
Digital PSTN Protocols
Supporting various ISDN PRI protocols such as EuroISDN, North American NI-2, Lucent™ 4/5ESS, Nortel™ DMS100 and others. It also supports different variants of CAS protocols, including MFC R2, E&M immediate start, E&M
delay dial / start and others.
Network Interfaces
4 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, advanced RTP latching
TLS, SRTP, HTTPS, SSH, client/server SIP Digest authentication, RADIUS Digest
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
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
Registration and Authentication
User registration restriction control, registration and authentication on behalf of users, SIP authentication server
for SBC users
Transport Mediation
Message Manipulation
Ability to add/modify/delete SIP headers and message body using advanced regular expressions (regex)
URI and Number Manipulations
URI user and host name manipulations, ingress and egress digit manipulation
Coder normalization including 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
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
Standalone Survivability
Impairment Mitigation
Voice Enhancement
Maintains local calls in the event of WAN failure. Outbound calls can use PSTN fallback for external connectivity
(including E911)
Packet Loss Concealment, Dynamic Programmable Jitter Buffer, Silence Suppression/Comfort
Noise Generation, RTP redundancy, broken connection detection
Transrating, RTCP-XR, Acoustic echo cancellation, replacing voice profile due to impairment detection, Fixed &
dynamic voice gain control
Direct Media
(No Media Anchoring)
Hair-pinning of local calls to avoid unnecessary media delays and bandwidth consumption
Voice Quality Monitoring
RTCP-XR, AudioCodes Session Experience Manager (SEM)
High Availability (Redundancy)
SBC 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, third-party routing control through REST API
Advanced Routing Criteria
QoE, bandwidth, SIP message (SIP request, coder type, etc.), Layer-3 parameters
Routing Features
Least-cost routing, call forking, load balancing, E911 gateway support, emergency call detection and prioritization
IETF standard SIP recording interface
Browser-based GUI, CLI, SNMP, INI Configuration file, REST API, EMS
Physical / Environmental
43.7 (1U) x 310 x 210 mm (HxWxD)
4.4 lb (2.0kg)
Desktop or 19” rack mount
100-240V, 50-60 Hz, 0.8A
Operational: 0 to 40°C (41 to 104°F); Storage: -25 to 70°C (-13 to 185°F)
Relative Humidity: 10 to 90% non-condensing
Regulatory Compliance
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
ETS300019-2-2 class T2.3
AudioCodes Ltd. (NasdaqGS: AUDC) designs,
develops and sells advanced Voice-over-IP
(VoIP) and converged VoIP and Data networking
products and applications to Service Providers
and Enterprises. AudioCodes is a VoIP
technology market leader, focused on converged
VoIP and data communications, and its products
are deployed globally in Broadband, Mobile,
Enterprise networks and Cable. The Company
provides a range of innovative, cost-effective
products including Media Gateways, MultiService Business Routers, Session Border
Controllers (SBC), Residential Gateways, IP
Phones, Media Servers, Value Added Applications
and Professional Services. AudioCodes’
underlying technology, VoIPerfectHD™, relies on
AudioCodes’ leadership in DSP, voice coding and
voice processing technologies. AudioCodes’ High
Definition (HD) VoIP technologies and products
provide enhanced intelligibility and a better
end user communication experience in Voice
International Headquarters
1 Hayarden Street,
Airport City
Lod 7019900, Israel
Tel: +972-3-976-4000
Fax: +972-3-976-4040
AudioCodes Inc.
27 World’s Fair Drive,
Somerset, NJ 08873
Contact us:
©2016 AudioCodes Ltd. All rights reserved.
AudioCodes, AC, HD VoIP, HD VoIP Sounds Better,
IPmedia, Mediant, MediaPack, What’s Inside
Matters, OSN, SmartTAP, User Management
Pack, VMAS, VoIPerfect, VoIPerfectHD, Your
Gateway To VoIP, 3GX, VocaNom, AudioCodes
One Voice and CloudBond are trademarks or
registered trademarks of AudioCodes Limited.
All other products or trademarks are property of
their respective owners. Product specifications
are subject to change without notice.
Ref. # LTRM-30034 06/16 V.6
Was this manual useful for you? yes no
Thank you for your participation!

* Your assessment is very important for improving the work of artificial intelligence, which forms the content of this project

Download PDF