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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.

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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

Disk Space

10 GB

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About Genesys

Every year, Genesys^® delivers more than 70 billion remarkable customer experiences for organizations in over 100 countries. Through the power of the cloud and AI, our technology connects every customer moment across marketing, sales and service on any channel, while also improving employee experiences. Genesys pioneered Experience as a Service^℠ so organizations of any size can provide true personalization at scale, interact with empathy, and foster customer trust and loyalty. This is enabled by Genesys Cloud^™, an all-in-one solution and the world's leading public cloud contact center platform, designed for rapid innovation, scalability and flexibility.