AudioCodes Session Border Controllers
Mediant CE/VE/SE Software Session Border Controller (SBC)DATASHEETMediantTM CE/VE/SE

AudioCodes Mediant software session border controller (SBC) isa highly scalable SBC solution supporting broad SIP interoperability,
advanced media handling and robust security. AudioCodes Mediantsoftware SBC enables enterprises and service providers to deliver
voice services, such as SIP trunking and unified communications, via
private or public clouds.
The Mediant software SBC is available in three variants to meet different customer deployment needs:
Mediant CE | a cloud-native SBC delivering high scalability and elasticity in virtualized cloud environments
Mediant VE | built for deployment in virtualized data centers, public clouds and NFV environments
Mediant SE | designed to run on commercial off-the-shelf servers (COTS) in high-scale communications environments

Comprehensive SBC functionality and SIP interoperability
Shared code base with AudioCodes field-proven, hardware-based SBCs
Rapid cloud deployment
Optimized resource consumption for private and public clouds such as Microsoft Azure and Amazon
Web Services, Google Cloud Platorm
NFV-readyProven interoperability with leading NFV orchestrators
Enhanced scalability
Easily scale from tens up to tens of thousands of concurrent sessions
High availability
1:1 active-standby configuration for business continuity
High performance and robust security
Built-in software-based media transcoding with support for encryption and protection from attacks
Qualified for leading UC and hosted telephony platforms
Certified SBC for Teams Direct Routing supporting media optimization
Integrated WebRTC gateway
Simple and secure WebRTC deployment, supporting both signaling and media

AudioCodes Session Border Controllers

DATASHEETMediantTM CE/VE/SE

Specifications

About AudioCodes

Capacities
Mediant CE

Mediant VEMediant SE

Max. Signaling Sessions

40,00024,00070,000Max. Media Sessions

40,00024,00070,000Max. SRTP-RTP Sessions

40,00010,00040,000Max. Transcoding Sessions

27,00012,000 1

30,000 1

Max. Registered Users

100,000

75,000500,000

Security
Access Control

DoS/DDoS line rate protection, bandwidth throttling, dynamic blacklisting

VoIP Firewall

RTP pinhole management, rogue RTP detection and prevention, SIP message policy, advanced RTP latching

Encryption and Authentication

TLS, DTLS, 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

Intrusion Detection SystemDetection and prevention of VoIP attacks, theft of service and unauthorized access

STIR/SHAEKNSTIR/SHAKEN support. Interworking with STI-AS/VSInteroperability
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

Registration and Authentication

User registration restriction control, registration and authentication on behalf of users, SIP authentication
serverTransport Mediation

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

Header Manipulation

Ability to add/modify/delete SIP headers and message body using advanced regular expressions (regex)

URI and Number Manipulations

URI user and host name 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, AMR-WB (G.722.2), SILK-NB/WB, Opus-NB/WB

Signal Conversion

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

WebRTC Gateway

Interworking between WebRTC devices and SIP networks. Supports WebSocket, Opus, VP8 video coder,
DTLS, RTP multiplexing, secure RTCP with feedback

NATLocal and far-end NAT traversal for support of remote workers, ICE full and lite support (RFC 8445)

Voice Quality and SLA
Call Admission Control

Based on bandwidth, session establishment rate, number of connections/registrations

Packet Marking

802.1p/Q VLAN tagging, DiffServ, TOS

Standalone Survivability

Maintains local calls in the event of WAN failure.

Impairment Mitigation

Packet Loss Concealment, Dynamic Programmable Jitter Buffer, Silence Suppression/Comfort Noise
Generation, RTP redundancy, broken connection detection

Voice Enhancement

Transrating, RTCP-XR, Acoustic echo cancellation, replacing voice profile due to impairment detection,
Fixed & dynamic voice gain control

Direct Media

Hair-pinning of local calls to avoid unnecessary media delays and bandwidth consumption while avoidingmedia anchoring

Voice Quality Monitoring

RTCP-XR, AudioCodes One Voice Operations Center (OVOC)High Availability

SBC 1+1 high availability with active calls preservation

Quality of Experience

Access control and media quality enhancements based on QoE and bandwidth utilization

Test Agent

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

SIP Routing
Routing Methods

Request URL, IP address, FQDN, ENUM, advanced LDAP, third-party routing control through REST API

Advanced Routing Criteria

QoE, bandwidth, SIP message (SIP request, coder type, etc.), Layer-3 parameters

Redundancy

Detection of proxy failures and subsequent routing to alternative proxies

Routing Features

Least-cost routing, call forking, load balancing, E911 gateway support, emergency call detection andprioritization

SBC Media Types

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

Recording Solutions

Lawful Interception (LI2), SIPREC for both audio and video sessions

ManagementOAM&P

Browser-based GUI, CLI, SNMP, INI Configuration file, REST API, HTTP reverse proxy
One Voice Operations Center (OVOC), Session Detail Records (SDRs)

Multi-Tenancy

Advanced multi-tenant SBC partitioning

Deployment Tools

VNFM/Stack manager (Mediant CE), HEAT templates, Cloud Formation

Auto-scaling CE

Automatic, REST API, CLI, Web UI

Cloud Environments
Public Cloud

Azure, AWS, GCP

Private Cloud

OpenStack, VMware® vSphere

Mediant VE SBC Minimum Requirements
Hypervisor
1

With media transcoding cluster

2Requires a dedicated software build

VMware® vSphere ESXi™ 5.5,
& 6.5 and above, Linux KVM,
Microsoft Hyper-V

Virtual Resources

1 vCPU; 2 GB RAM; 10 GB Disk;
Virtual NICS - 2/3(Standalone/HA)

AudioCodes Ltd. (NasdaqGS: AUDC)
is a leading vendor of advanced voice
networking and media processing
solutions for the digital workplace.
With a commitment to the humanvoice deeply embedded in its DNA,
AudioCodes enables enterprises andservice providers to build and operate
all-IP voice networks for unified
communications, contact centers andhosted business services. AudioCodes’
wide range of innovative products,
solutions and services are used by large
multinational enterprises and leadingtier one operators worldwide.

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

AudioCodes Inc.

80 Kingsbridge Rd - Piscataway, NJ 08854
Tel: +1-732-469-0880
Contact us: www.audiocodes.com/contact
Website: www.audiocodes.com©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 trademarks of AudioCodes Limited. Allother products or trademarks are property of theirrespective owners. Product specifications are subject
to change without notice.
Ref. 10/23