AudioCodes Session Border Controllers
Hybrid SBC and Media Gateway

DATASHEETMediant™ 500L

The AudioCodes Mediant 500L enterprise session bordercontroller (E-SBC) and media gateway is a compact, highperformance VoIP connectivity solution for small enterprises andbranch office locations.
Scaling up to 60 concurrent sessions, the Mediant 500L connects
IP-PBXs to any SIP trunking service provider and offers superior
performance in connecting any SIP to SIP environment.
In addition, the Mediant 500L supports up to 8 voice channels 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.

60 SBC Sessions | 8 TDM Sessions | Branch Survivability

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 andservice theft

Superior voice quality

Advanced capabilities for optimizing and monitoring voice service quality

High resiliency

Local branch survivability and PSTN fallback with E911

AudioCodes Session Border Controllers

DATASHEETMediant™ 500LSpecifications
Capacities

Max. Signaling

60

Max. Registered Users

20060

Max. RTP/SRTP Sessions

Telephony Interfaces
Digital1-4 BRI ports, network S/T interfaces, NT or TE termination

AnalogUp to 4 FXS and 4 FXO ports

Network Interfaces
EthernetSecurity

4 GE interfaces configured in 1+1 redundancy or as individual ports

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, 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

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 moreRegistration and Authentication

SIP Registrar, registration on behalf of users/servers, SIP Digest access authentication

Transport Mediation

Mediation between SIP over UDP/TCP/TLS, IPv4/IPv6, RTP/SRTP (SDES)

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

SIP Interworking

3xx redirect, REFER, PRACK, session timer, early media, call hold, delayed offer

Signal Conversion

DTMF/RFC 2833/SIP, T.38 fax, T.38 V3, packet-time conversion

NATLocal and far-end NAT traversal for support of remote workers

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

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

Test Agent

Ability to remotely verify connectivity, voice quality and SIP message flow between SIP UAs

Voice Quality and SLA

SIP Routing
Routing Criteria

Incoming SIP trunk, DID ranges, host names, any SIP headers, codecs, QoE, bandwidth

Querying External Databases

Routing based on customized 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 prioritization

SIPRECIETF standard SIP recording interface

Management
OAM&P

Physical/Environmental

Browser-based GUI, CLI, SNMP, INI Configuration file, REST API, EMS

Dimensions

51 x 296 x 160 mm (2 x 11.65 x 6.3 in.) (HxWxD)

Weight670g

Mounting

Desktop

PowerSingle universal AC power supply 100-240V, 50-60 Hz, 12V/3A or 12V/5AEnvironmental

Operational: 5 to 40° C (41 to 104°F); Storage: -25 to 85°C (-13 to 185°F)
Relative Humidity: 10 to 90% non-condensing

International Headquarters

AudioCodes Inc.

1 Hayarden Street, Airport City
Lod 7019900, Israel
Tel: +972-3-976-4000
Fax: +972-3-976-4040

27 World’s Fair Drive,Somerset, NJ 08873
Tel:+1-732-469-0880
Fax:+1-732-469-2298

Contact us: www.audiocodes.com/contact
Website: www.audiocodes.com©2021 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, AudioCodes Meeting Insights, AudioCodes RoomExperience 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.