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Digital and Analog Media Gateways

Mediant 1000

Mediant 1000
Modular, hybrid TDM/SIP SBC for small-to-medium sized enterprise locations.
Mediant 1000
Open Modular digital TDM interfaces tooltip

Modular digital TDM interfaces

Open Up to 7 10/100/1000 Base-T interfaces tooltip

Up to 7 10/100/1000 Base-T interfaces

Open Seamless migration and PSTN fallback tooltip

Seamless migration and PSTN fallback

  • For small-to-medium sized enterprises
  • Modular hybrid SBC and gateway
  • Comprehensive SIP interoperability
  • Supports digital TDM interfaces


The AudioCodes Mediant 1000 hybrid media gateway and enterprise session border controller (E-SBC) is designed to provide a complete connectivity solution for small-to-medium sized enterprise locations.

Supporting up to 192 concurrent voice sessions in a 1U modular platform, the Mediant 1000 provides versatile connectivity between TDM and VoIP networks.

The modular Mediant 1000 connects IP-PBXs to any SIP trunking service provider, scaling to 150 concurrent SBC sessions. It offers superior performance in connecting any SIP to SIP environments, legacy TDM-based PBX systems to IP networks, and IP-PBXs to the PSTN.

The Mediant 1000 series is compatible with the Microsoft SIP Gateway to enable the direct integration of analog devices with Microsoft Teams Phone System.


  • Support a wide variety of qualified SIP trunks, SIP platforms and IP cloud services
  • Hybrid VoIP media gateway and E-SBC lowers CAPEX and reduces space and power footprints
  • Simplified integrated management reduces OPEX
  • A highly integrated device for secured SIP Trunking and PSTN access, forming a single and managed point of demarcation for VoIP networks
  • Scalable “pay-as-you-grow” modular architecture


  • Rich and powerful SIP normalization and routing mechanism for seamless interoperability
  • Support for digital TDM interfaces
  • Hybrid SBC enables seamless migration and PSTN fallback
  • Enhanced perimeter defense against DoS attacks
  • Advanced monitoring tools help analyze and optimize VoIP service quality-of-experience
  • Embedded DSP performs media handling (codec transcoding, DTMF, fax, call progress announcements, etc.)
  • E1/T1/J1 bypass and dual power supplies for high-availability


Max. SBC Sessions 150 sessions
Max. Transcoding Sessions 96 sessions
Max. Registered Users 600
Max. Voice Channels
  • 192 - supporting E1/T1/J1 trunks
  • Up to 6 E1 and 8 T1/J1 trunks
LAN Interfaces Up to 7 10/100/1000 Base-T interfaces configured in 1+1 redundancy or as individual ports
Server Platform (Optional) Embedded Intel-based Server platform for third-party services
SIP Survivability Up to 600 users
IP Media Conferencing and Announcements server Support via Netann and MSCML
Supported DSP Capabilities IBS, Echo-Cancellation (EC), Caller-ID (CID), Silence-Compression (SC), VAD, CNG, Automatic Gain Control (AGC) and Answer Detector (AD), RTCP XR, T.38, G.711, G.726, G.727, G.729, G.723.1, G.722, GSM FR, GSM EFR, MS GSM, iLBC, EVRC, QCELP, AMR
Physical Dimensions (W) 345 mm, (D) 320 mm, (H) 1U
Powering Single/Dual-redundant 100-240V, 50-60Hz, 1.5A power supply

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