AudioCodes Session Border Controllers

DATASHEETMediant™ 3100
Hybrid SBC and Media Gateway

The AudioCodes Mediant 3100 session border controller (SBC) and media
gateway is a complete connectivity solution for medium-to-large sizedenterprises, contact centers and service providers.
Scaling up to 5,000 concurrent SBC sessions, the Mediant 3100 connects IP-PBXs
to any SIP trunking service provider and offers superior performance in connecting
any SIP to SIP environment.
In addition, the Mediant 3100 supports up to up to 64 E1/T1 spans in a 2U platform 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.

5,000 SBC Sessions | 64 E1/T1 Spans | Extensive Vocoder Support |Certified for Microsoft Teams Direct Routing with local media optimization

Comprehensive interoperability

Proven interoperability with SIP trunks, unified communications solutions, PBXs and IP cloud services

Hybrid functionality

True hybrid SBC and gateway platform for gradual migration to IP communications, low CAPEX andreduced 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

Service resilience

Local branch survivability and PSTN fallback

AudioCodes Session Border Controllers

DATASHEETMediant™ 3100

Specifications
Capacities

Max. Signaling

5,000

Max. RTP/SRTP Sessions

5,000

Max. Registered Users

20,000Max. Transcoding Sessions

3,200

Telephony Interfaces
Digital PSTN Protocols

Various ISDN PRI protocols such as EuroISDN, North American NI-2, Lucent™ 4/5ESS, Nortel™ DMS- 100 and others.

PSTN

8 to 64 E1/T1 interfaces

Network Interfaces
EthernetSecurity

8 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, DTLS, 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 more

Registration and Authentication

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

Transport Mediation

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

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

Transcoding and Vocoders

Coder normalization including transcoding, coder enforcement and re-prioritization, extensive vocoder support: G.711, G.723.1, G.726, G.729A/B, GSM-FR,
AMR-NB/WB, Opus-NB/WB, SILK-NB/WB

Signal Conversion

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

WebRTC Gateway

Interworking between WebRTC endpoints and SIP networks. Supports WebSocket, Opus, VP8 video coder, DTLS, RTP multiplexing, secure RTCP withfeedback.

NATLocal and far-end NAT traversal for support of remote workers, ICE full and lite support (RFC 8445)

Voice Quality and SLA
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. Outbound calls can use PSTN fallback (including E911).

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

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

2U high 19-inch rack wide
(H x W x D) 88 x 438 x 490 mm (3.5 x 17.24 x 19 inches)Weight

11.5 kg (25.3 lbs.) for fully-populated chassis

Operating Temperature

Operational: 0° - 40° C (41° - 104° F)
Storage: -25° - 70° C (-13° - 158° F)
Humidity: 5% - 90% non-condensing

PowerRedundant Dual Feed, 100-240 V AC/9-4A, 50-60Hz or 48VDC 18A max

Naimi Park, 6 Ofra Haza Street, Or Yehuda, Israel
Tel: +972-3-976-4000
80 Kingsbridge Rd - Piscataway, NJ 08854
Tel: +1-732-469-0880
www.audiocodes.com©2024 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 and AudioCodes One Voice are trademarks or registered trademarksof AudioCodes Limited. All other products or trademarks are property of their respective owners. Productspecifications are subject to change without notice.
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