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What is OpenSIPS?

OpenSIPS serves as an open-source SIP proxy/server capable of managing voice, video, instant messaging, presence, and a variety of other SIP extensions. This adaptable signaling SIP server is tailored for carriers, telecom operators, and ITSPs, offering solutions for a broad range of applications including Class 4 and Class 5 residential platforms, trunking and wholesale services, as well as enterprise and virtual PBX systems. It also functions effectively as a session border controller, application server, load balancer, and is utilized in call centers among many other applications. With the ability to handle high throughput, it supports tens of thousands of calls each second along with millions of concurrent connections, making it a favored option for numerous SIP use cases. The strength of OpenSIPS lies in its flexible routing and integration capabilities, enabling users to implement custom routing logic via routing scripts and a wealth of interfacing APIs. Featuring over 120 modules, it supports robust application development for SIP management, backend operations, and routing processes. Furthermore, the 3.2 version of OpenSIPS has received a comprehensive security audit, bolstering its safety and dependability for users. As a resilient framework, it not only caters to present requirements but also adapts and evolves to meet the ever-changing challenges within the telecommunications sector, ensuring its relevance for years to come. This commitment to continuous improvement underscores the importance of innovation in maintaining a competitive edge in the industry.

What is AudioCodes Software-Defined Voice Network?

Transforming communication and collaboration through digital means is crucial for contemporary businesses. AudioCodes has introduced the Software-Defined Voice Network (SDvN) by leveraging Software-Defined Networking (SDN) principles, offering a versatile solution specifically designed for large enterprises embarking on their digital transformation journey. This cutting-edge framework enhances voice network management and optimizes call routing efficiency by dismantling communication barriers, integrating seamlessly with SIP trunk services, and utilizing the corporate IP network for enhanced call handling. The solution features AudioCodes' session border controllers (SBCs) along with an array of tools for global call routing, policy management, user oversight, and voice network administration, which together create a unified network that effortlessly connects both on-premises and cloud-based communication systems. Additionally, SDvN empowers organizations to upgrade their communication infrastructures without the need to discard existing platforms, thereby facilitating a more seamless transition into the evolving landscape of business communications. This adaptability not only aids in future-proofing their communication strategies but also supports improved collaboration across teams and departments.

Media

Media

Integrations Supported

Ansible
Docker
Fonimo
Netdata

Integrations Supported

Azure Marketplace
Brekeke CCS
Chatbot Builder AI
Check Point IPS
Check Point Infinity
Cocohub
Junos Traffic Vision
Oreka TR
Rasa

API Availability

Has API

API Availability

Pricing Information

Free
Free Version

Pricing Information

Pricing not provided

Supported Platforms

On-Prem
Linux

Supported Platforms

SaaS
On-Prem

Customer Service / Support

Web-Based Support

Customer Service / Support

Standard Support
Web-Based Support

Training Options

Documentation Hub
Webinars

Training Options

Documentation Hub
Webinars
Online Training

Company Facts

Organization Name

OpenSIPS

Company Location

United States

Company Website

opensips.org

Company Facts

Organization Name

AudioCodes

Date Founded

1993

Company Location

Israel

Company Website

www.audiocodes.com/solutions-products/solutions/software-defined-voice-network-digital-transformation

Categories and Features

PBX

Not specified

Categories and Features

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