AudioCodes Software Enterprise Session Border Controller

The AudioCodes Software Enterprise Session Border Controller (E-SBC) Server Edition and Virtual Edition provide a flexible and scalable SBC solution that meets the requirements of today’s datacenter infrastructures. E-SBC supports wide-ranging SIP interoperability, ensures high-quality service, and enables scalable, reliable, and secure connectivity between different VoIP networks.

Download PDF
Audiocodes software enterprise session border controller

Benefits

  • Designed for deployment in standardized datacenter environments
  • Supports network function virtualization
  • Simplifies and accelerates SBC deployments
  • Offers comprehensive security, interoperability, and reliability
  • Delivers excellent service performance and voice quality
  • Flexible licensing options for cost-effective scalability
  • Runs on dedicated COTS servers and in virtualized environments

Key Features

  • Scalable to thousands of SBC sessions
  • Extensive SIP mediation capabilities
  • Supports remote workers and mobile SIP clients
  • Perimeter defense against denial of service, fraud, and eavesdropping
  • VoIP quality monitoring and enforcement
  • Branch survivability during WAN failure
  • Active/standby high availability

Server Edition

  • Runs on dedicated, commercial, off-the-shelf (COTS) servers
  • Aimed at highly scalable environments

Virtual Edition

  • Runs in virtualized datacenter environments
  • Supports VMware, Hyper-V, and KVM systems
  • Cloud environments: OpenStack, Amazon Web Services, and Cloudband
  • Extensive Mediation Capabilities and Proven Interoperability

E-SBC includes comprehensive media security and SIP normalization capabilities. It provides full interoperability with an extensive list of IP PBX systems, unified communications solutions, and SIP-trunking services.

Security and Reliability

E-SBC offers robust protection for an IP communications infrastructure. It prevents fraud and service theft and guards against cyber-attacks and other events that could impact service. E-SBC also maintains the high quality needed for enterprise VoIP communications with active/standby high availability. Advanced call routing mechanisms, network voice quality monitoring, and branch survivability capabilities minimize communications downtime.

Applications

  • SIP trunking
  • Hosted PBX and UCaaS
  • IP contact centers
  • Remote and mobile worker support
  • SIP mediation between UC and IP PBX systems
  • Residential VoIP

Capacities

Virtual Edition

Server Edition

Max Signaling Sessions

6,000

32,000

Max Media Sessions

6,000

24,000

Max SRTP–RTP Sessions

4,000

16,000

Max Registered Users

30,000

120,000

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 Digest

Privacy

Topology hiding, user privacy

Traffic Separation

VLAN/physical interface separation for multiple media, control, and OAMP interfaces

Intrusion Detection System

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

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 offer

Registration and Authentication

User registration restriction control, registration, and authentication on behalf of users, SIP authentication server for SBC users

Transport Mediation

SIP over UDP/TCP/TLS/WebSocket, 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 hostname 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.729, GSM-FR, AMR-NB/WB, SILK-NB/WB, Opus-NB/WB

Signal Conversion

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

WebRTC Controller

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

NAT

Local and far-end NAT traversal for support of remote workers

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 (No Media Anchoring)

Hairpinning of local calls to avoid unnecessary media delays and bandwidth consumption

Voice Quality Monitoring

RTCP-XR, AudioCodes Session Experience Manager (SEM)

High Availability (Redundancy)

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

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

Management

OAM&P

Browser-based GUI, CLI, SNMP, INI configuration file, REST API, EMS

Multi-Tenancy

Advanced multi-tenant SBC partitioning

SIP Routing

Routing Methods

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

Advanced Routing

802.1p/Q VLAN tagging, DiffServ, TOS

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 and prioritization

SIPRec

IETF standard SIP-recording interface

Virtual Edition VE SBC Minimum Requirements

Hypervisor

VMware vSphere ESXi 5.x, Linux KVM, Microsoft Hyper-V

Virtual NICs

2 (standalone) 3 (high availability)

Memory

2 GB

Virtual CPUs

1

Disk Space

10 GB

Take this information to go.

Download a PDF
Genesys.com
+1.888.436.3797

Contact us

Products

Company

About Genesys

Every year, Genesys® orchestrates more than 70 billion remarkable customer experiences for organizations in more than 100 countries. Through the power of our cloud, digital and AI technologies, organizations can realize Experience as a Service℠, our vision for empathetic customer experiences at scale. With Genesys, organizations have the power to deliver proactive, predictive, and hyper personalized experiences to deepen their customer connection across every marketing, sales, and service moment on any channel, while also improving employee productivity and engagement. By transforming back-office technology to a modern revenue velocity engine Genesys enables true intimacy at scale to foster customer trust and loyalty.

Copyright © 2021 Genesys. All rights reserved. Terms of Use | Privacy Policy | Email Subscription |

Genesys ChatLive Assistance
Close Widget

How can we help you today?
Start ChatChat with Sales
Start DemoSchedule a Demo
Start CareCustomer Care
Start CareerGenesys Careers

Powered by

Genesys ChatLive Assistance