AudioCodes Mediant 1000 Datasheet
AudioCodes Session Border Controllers
DATASHEET
Mediant™ 1000
Hybrid SBC and Media Gateway
The AudioCodes Mediant 1000 enterprise session border
controller (E-SBC) and media gateway oers a complete
connecvity soluon 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 oers
superior performance in connecng any SIP to SIP environment.
In addion, the Mediant 1000 supports up to 192 voice channels in a 1U plaorm to enable versale connecvity
between TDM and VoIP networks, such as connecng legacy TDM PBX systems to IP networks and IP-PBXs to the PSTN.
150 SBC Sessions | 192 TDM Sessions | Modular | Extensive Vocoder Support | Certied SBC for Teams
Direct Routing supporting media optimization
Comprehensive interoperability
Proven interoperability with SIP trunks, SIP plaorms and IP cloud services
Hybrid functionality
True hybrid SBC and gateway plaorm for gradual migraon, low CAPEX and reduced space and power
footprints
Enhanced security
Robust perimeter defense against cyber, DoS and DDoS aacks, as well as eavesdropping, fraud and
service the
Superior voice quality
Advanced capabilies for opmizing and monitoring voice service quality
High resiliency
Local branch survivability and PSTN fallback with E911
AudioCodes Session Border Controllers
DATASHEET
Specicaons
Mediant™ 1000
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
BRI Module
Analog Module
Media Processing Module Up to 4 Media Processing modules (MPM), providing additional DSP resources
Various ISDN PRI protocols such as EuroISDN, North American NI-2, Lucent™ 4/5ESS, Nortel™ DMS- 100 and others. Dierent CAS protocols, including MFC R2,
E&M immediate start, E&M delay dial/start and others.
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)
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 start
Network Interfaces
Ethernet Up to 6 GE interfaces congured in 1+1 redundancy or as individual ports
Security
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/ser ver SIP Digest authentication, RADIUS Digest
Privacy 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 oer 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
Signal Conversion DTMF/RFC 2833/SIP, T.38 fax, V.34, packet-time conversion
NAT Local and far-end NAT traversal for support of remote workers
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
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, DiServ, TOS
Standalone Survivability Maintains local calls in the event of WAN failure. Outbound calls can use PSTN fallback (including E911).
Voice Monitoring and Enhancement
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 ow between SIP UAs
Transrating, RTCP-XR, acoustic echo cancellation, replacing voice prole due to impairment detection, xed and dynamic voice gain control, packet loss
concealment, dynamic programmable jitter buer, silence suppression/comfort noise generation, RTP redundancy, broken connection detection
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 Congured 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
Browser-based GUI, CLI, SNMP, INI Conguration le, REST API,
One Voice Operations Center (OVOC)
OSN Server Platform (Optional)
Single Chassis Integration Optional embedded, x86, Intel-based Open Solution Network platform for third-party applications
Physical/Environmental
Dimensions 1U x 444 x 355 mm (HxWxD) Weight Approx. 9.7lb (4.4kg)
Mounting Desktop or 19” mount Power 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
Internaonal Headquarters
1 Hayarden Street, Airport City
Lod 7019900, Israel
Tel: +972-3-976-4000
Fax: +972-3-976-4040
AudioCodes Inc.
200 Coontail 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
Beer, IPmedia, Mediant, MediaPack, What’s Inside Maers, OSN, SmartTAP, User
Management Pack, VMAS, VoIPerfect, VoIPerfectHD, Your Gateway To VoIP, 3GX,
VocaNom, AudioCodes One Voice, AudioCodes Meeng Insights, AudioCodes Room
Experience and CloudBond are trademarks or registered trademarks of AudioCodes
Limited. All other products or trademarks are property of their respecve owners.
Product specicaons are subject to change without noce.
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