AudioCodes Mediant 1000 4 FXS & 4 FXO (M1K-A1)
AudioCodes Mediant 1000 4 FXS & 4 FXO (M1K-A1) Summary
The Mediant 1000 is based on VoIPerfect, AudioCodes underlying, best-of-breed, media gateway core technology for all of its products. The Mediant 1000 provides superior voice-technology for connecting legacy telephone and PBX systems to IP networks, as well as seamless connection of the IP-PBX to the PSTN. In addition to operating as a pure media gateway, the Mediant 1000 can also host partner applications and serve as an IP-PBX platform. The Mediant 1000 is fully interoperable with multiple vendor gateways, Softswitches, gatekeepers, proxy servers, IP phones, Session Border Controllers and firewalls.
The Mediant 1000 matches the density requirements for small locations while meeting enterprises and service providers’ demands for scalability. The compact Mediant 1000 Modular Gateway is extremely scalable and supports multiples of 1, 2, or 4 E1/T1/J1 spans, 4 to 20 BRI ports or 1 to 24 analog ports in various FXO/FXS configurations. The Mediant 1000 also supports mixed digital/analog with media processing capabilities such as conferencing, play/record configurations.
The Mediant 1000 can support a variety of telephony interfaces. The digital module can be configured as regular E1/T1/J1 interfaces, with up to 1 or 2 paired spans acting as life-line interfaces for switching to the PSTN in case of power failure or network problems. The analog module is available as regular FXS or FXO interfaces, where 1 FXS line can be used as a life-line interface for switching to the PSTN.
Customers who connect to centralized IP-Centrex services, as well as branch offices of enterprises who use a centralized IP-PBX server may face a survivability challenge. Stand Alone Survivability (SAS), supported in the Mediant 1000 is based on the SIP B2BUA (Back to Back User Agent) functionality, and enables the backup of SIP clients such as SIP IP and Soft Phones in the case of a connectivity failure with the centralized SIP server.
AudioCodes Mediant 1000 4 FXS & 4 FXO (M1K-A1) Core Features
- Employs AudioCodes VoIPerfect technology for outstanding voice quality
- Scalable “pay-as-you-grow” modular architecture
- Rich offering of digital (E1/T1/J1), analog (FXS/FXO), and BRI interfaces
- Cost-efficient for low density gateways
- Lifeline fallback to PSTN in case of power failure or network degradation
- PSTN fallback for assured connectivity
- Internal OSN Server for hosting 3rd party application
- An ideal match as a platform for IP-PBX
- Media processing and conferencing option
- Stand Alone Survivability (SAS) for service continuity
AudioCodes Mediant 1000 4 FXS & 4 FXO (M1K-A1) Technical Specifications
- Interfaces
- Modularity and Capacity
- Voice interface: Equipped with 6 Slots that can host voice modules. Up to a maximum of 24 analog ports or 4 digital spans.
- Digital Modules
- 1, 2 or 4 E1/T1/J1 spans using RJ-48c connectors per module. Up to 4 digital modules (maximum 4 spans per gateway). Optional 1+1 or 2+2 fallback spans.
- Analog FXO and FXS Modules
- 4 ports using RJ-11 connectors per module; Up to 6 modules per gateway, Ground Start and Loop Start.
- BRI Module
- 4 BRI ports (8 calls) per module, up to 5 modules per gateway with S/T interfaces. Supports Euro ISDN, NI2, 5ESS or QSIG.
- Media Processing Module
- Hosting media processing features: conferencing, play/record over HTTP or NFS.
- I/O
- MOH (Music On Hold), NB (Night Bell).
- Ethernet
- Dual Redundant 10/100 Base-TX Ethernet ports via 2 RJ-45 connectors.
- RS-232
- Debugging and configuration.
- Media Processing
- Voice Coders
- G.711, G.726, G.727, G.723.1, G.729, GSM FR, MS GSM, iLBC, EG.711, EVRC, QCELP, AMR, GSM EFR, G.722 Independent dynamic vocoder selection per channel.
- Echo Cancellation
- G.165 and G.168-2002, with 32, 64 or 128 tail length.
- Quality Enhancement
- Dynamic programmable jitter buffer, VAD, CNG, 802.1p/Q VLAN tagging, DiffServ, voice quality monitoring, G.729B, RTCPXR.
- DTMF/MF Transport
- Packet side or PSTN side detection and generation, RFC 2833 compliant DTMF relay. Call Progress tones detection and generation.
- IP Transport
- VoIP (RTP/RTCP) per IETF RFC 3550 and 3551.
- Fax and Modem Transport
- T.38 compliant (real time fax), Automatic bypass to PCM or ADPCM.
- Signaling
- Digital –PSTN Protocols
- CAS: MF-R1: T1 CAS (E&M, Loop, Start, Feature Group-D, E911CAMA), E1 CAS (R2 MFC), R1.5 numerous protocol and country variants ISDN PRI: ETSI/EURO ISDN, ANSI NI2 and other variants (DMS100, 5ESS) QSIG (Basic and supplementary), IUA (SIGTRAN), VN3, VN4, VN6
- Analog Signaling
- FXS; Caller ID; polarity reversal; metering tones, distinctive ringing, visual message waiting indication, Loop Start, Ground Start
- Control & Management
- Control Protocols
- SIP, MSCML, H.323 (MEGACO – for digital trunks)1
- Operations & Management
- AudioCodes Element Management System Embedded HTTP Web Server, Telnet, SNMP V2, V3 Remote configuration and software download via TFTP, HTTP, HTTPS, DHCP and BootP, RADIUS, Syslog (for events, alarms and CDRs) Auto Update
- Security
- IPSEC, HTTPS, TLS (SIPS), SSL, Web access list, RADIUS login and SRTP2
- Hardware Specifications
- Power Supply
- 100-240V, 50-60Hz, 1.5A Max, Single (default) or redundant (optional) power supply configurations
- Physical
