AudioCodes Session Border Controllers

DATASHEETMediant™ 800
Hybrid SBC and Media Gateway

The AudioCodes Mediant 800 enterprise session bordercontroller (E-SBC) and media gateway offers a complete
connectivity solution for small-to-medium sized enterprises.
Scaling up to 400 concurrent sessions, the Mediant 800 connects
IP-PBXs to any SIP trunking service provider and offers superior
performance in connecting any SIP to SIP environment.
In addition, the Mediant 800 supports up to 124 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.

400 SBC Sessions | 124 TDM Sessions | 1+1 High Availability | Certified SBC for Teams Direct Routing |Supports OPUS and SILK
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
High availability using 1+1 redundancy, local branch survivability and PSTN fallback

AudioCodes Session Border Controllers

DATASHEETMediant™ 800

Specifications
Capacities
Max. Signaling

Max. RTP/SRTP Sessions

Max. Transcoding Sessions

Max. Registered UsersMediant 800B

250

250/250

57

1500

Mediant 800C

400

400/250

114

2000

Telephony Interfaces
Analog

4 FXS/FXO ports

Digital

Up to 4 E1/T1 interfaces; 4/8 BRI Ports

Clock Source

5 ppm High Precision

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.

Network Interfaces
Ethernet

Security
Access Control
VoIP Firewall
Encryption/Authentication
Privacy
Traffic Separation
Intrusion Detection System

Interoperability
SIP B2BUASIP Interworking
Registration and Authentication
Transport Mediation
Header Manipulation
Number Manipulations
Transcoding and Vocoders
Signal Conversion
WebRTC Gateway
NAT

4 GE or 4 GE + 8 FE interfaces configured in 1+1 redundancy or as individual ports
DoS/DDoS line rate protection, bandwidth throttling, dynamic blacklisting (Intrusion Detection System)

RTP pinhole management, rogue RTP detection and prevention, SIP message policy, advanced RTP latching
TLS, DTLS, SRTP, HTTPS, SSH, client/server SIP Digest authentication, RADIUS Digest
Automatic topology hiding, user privacy

VLAN/physical interface separation for multiple media, control and OAMP interfaces
Detection and prevention of VoIP attacks, theft of service and unauthorized access
Full SIP transparency, mature and broadly deployed SIP stack, stateful proxy mode

3xx redirect, REFER, PRACK, session timer, early media, call hold, delayed offer and more
SIP Registrar, registration on behalf of users/servers, SIP Digest access authentication

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

Add/modify/delete SIP headers and message body using simple WireShark-like language with powerful capabilities such as variables and utility functions
Ingress and egress digit manipulation

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, AMR-WB (G.722.2), SILK-NB/WB, Opus-NB/WB, iLBC
DTMF/RFC 2833/SIP, T.38 fax, T.38 V3, V.34, packet-time conversion

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

High Availability

SBC high availability with two-box redundancy, active calls preserved

Test Agent

Ability to remotely verify connectivity, voice quality and SIP message flow between SIP UAs

SIP Call Handling
Criteria
Querying External Databases
Available Destinations
Advanced Features
SBC Media Types
SIPRECManagement
OAM&P

Physical/Environmental
Dimensions
Mounting
Power

Incoming SIP trunk, DID ranges, host names, any SIP headers, codecs, QoE, bandwidth
Destinations based on customized queries of ENUM, LDAP, HTTP server (REST API)
Configured SIP peers, registered users, IP address, request URI

Alternative destinations, load balancing, LCR, call forking, E911 emergency call detection and prioritization
Audio\Video\Fax\Text\Message Session Relay Protocol (MSRP)\Binary Floor Control Protocol (BFCP)IETF standard SIP recording interface, supporting both audio and video SBC sessions
Browser-based GUI, CLI, SNMP, INI Configuration file, REST API,One Voice Operations Center (OVOC)
1U x 345mm x 320mm (HxWxD)

Operating Temperature

Internal AC power supply rated: 100-240 VAC ~50- 60Hz 1.5A maximum(Optional) Additional 12V 10A DC power, via an AudioCodes external AC/DC power adaptor

1 Hayarden Street, Airport City, Lod, 7019900, Israel
Tel: +972-3-976-4000
80 Kingsbridge Rd - Piscataway, NJ 08854
Tel: +1-732-469-0880
www.audiocodes.com

WeightDesktop or 19” rack mount

Approx. 5.95lb (2.7kg) loaded with OSN
5°-40° C©2023 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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