AudioCodes Session Border Controllers
Hybrid SBC and Media Gateway

DATASHEETMediant™ 1000

The AudioCodes Mediant 1000 enterprise session bordercontroller (E-SBC) and media gateway offers a complete
connectivity solution for small-to-medium sized enterprises.
Scaling up to 150 concurrent sessions, the Mediant 1000
connects IP-PBXs to any SIP trunking service provider and offers
superior performance in connecting any SIP to SIP environment.
In addition, the Mediant 1000 supports up to 192 voice channels in a 1U 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.

150 SBC Sessions | 192 TDM Sessions | Modular | Extensive Vocoder Support | Certified SBC for Teams
Direct Routing supporting media optimization

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 theftSuperior 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™ 1000

Specifications
Capacities

Max. Signaling

150

Max. RTP/SRTP Sessions

120

Max. Transcoding Sessions

96

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

Digital PSTN Protocols

Various ISDN PRI protocols such as EuroISDN, North American NI-2, Lucent™ 4/5ESS, Nortel™ DMS- 100 and others. Different CAS protocols, including MFC R2,E&M immediate start, E&M delay dial/start and others.

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 interfaces, provided on 4-port FXS modules, ground/loop start
Up to 24 FXO interfaces, provided on 4 port-FXO modules, ground/loop startMedia Processing Module

Up to 4 Media Processing modules (MPM), providing additional DSP resources

Network Interfaces
EthernetSecurity

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 (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 DigestPrivacy

Automatic 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, 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

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, G.722, G.727, iLBC, QCELP, GSM EFR

Signal Conversion

DTMF/RFC 2833/SIP, T.38 fax, V.34, packet-time conversion

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

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 Call Handling
Criteria

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

Querying External Databases

Destinations based on customized queries of ENUM, LDAP, HTTP server (REST API)

Advanced Features

Alternative destinations, load balancing, LCR, call forking, E911 emergency call detection and prioritization

Available Destinations

Configured SIP peers, registered users, IP address, request URI

SBC Media Types

Audio\Video\Fax\Text\Message Session Relay Protocol (MSRP)\Binary Floor Control Protocol (BFCP)Management
OAM&P

OSN Server Platform (Optional)
Single Chassis Integration

Browser-based GUI, CLI, SNMP, INI Configuration file, REST API,One Voice Operations Center (OVOC)
Optional embedded, x86, Intel-based Open Solution Network platform for third-party applications

Physical/Environmental
Dimensions

1U x 444 x 355 mm (HxWxD)

WeightApprox. 9.7lb (4.4kg)

Mounting

Desktop or 19” mountPower

Dual power supply 100-240V, 50-60 Hz, 1.5A max

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

International Headquarters

AudioCodes Inc.

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

200 Cottontail Lane, Suite A101E,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.