OpenSIPS 3.1 by Bogdan-Andrei Iancu

OpenSIPS 3.1, on the Verge of Class 5 Enrichment by Bogdan-Andrei Iancu, Founder and Developer at OpenSIPS Project

Abstract:

The upcoming 3.1 release of OpenSIPS will focus on complex call crafting, that is increasing OpenSIPS’s ability to create and handle complex calling scenarios where multiple SIP calls are mixed and able to interact. Put simply, OpenSIPS 3.1 will address the Class 5 specific calling features and how to control such calling features via APIs.

Description:

Historically, OpenSIPS started as a SIP proxy and for many years all the additions were focused on the routing capabilities. Nevertheless, we are a dynamic project and the needs of our users (and their services) are dictating the evolution path of the projects. We’ll review the history of OpenSIPS so you understand its beginnings, evolution, many common applications, and the vibrant ecosystem supporting OpenSIPS today.

Even if OpenSIPS already developed capabilities transcending the definition of a SIP proxy, like UAC (User Agent Client) authentication, Topology Hiding, Back2Back, Call Recording, Gatewaying, the upcoming 3.1 release is taking a huge step for enabling the Class 5 capabilities into OpenSIPS.

Without aiming to transform OpenSIPS into a SIP Back-to-Back UA (User Agent), OpenSIPS 3.1 philosophy is to enhance the dialog, UAC and UAS (User Agent Server) related capabilities all the way to a point where you can combine and use them to build complex Class 5 specific features, like Call Pickups, Call Transfers, Call Parking, Music On Hold, Call Listing and Barging, Call Recording, RingBack Tones, Pre-Call Announcements or N-Way Conferencing. Again, without any Back2Back, just a skillful crafting of proxied calls, of SIP UACs and UASs (User Agent Client and User Agent Server).

Even more, for the first time, OpenSIPS 3.1 will provide a Calling API, an API that will allow external apps/services to start, monitor and control SIP calls via OpenSIPS, without touching or even understanding the SIP protocol. Several features, like MoH (Music on Hold) or Call Transfers will be provided directly via API. Isn’t that cool to enrich any dummy audio-enabled phone to the rank of PBX flavoured terminal by simply using the API via Web or Apps?

Asia Case Studies:

From the OpenSIPS Solutions side (business, undisclosed):

  • Infrastructure SBC to replace existing outdated components
  • Infrastructure Load-Balancer to replace existing outdated components
  • Integrators and Solution Providers
  • Virtual Hosted PBX system

From the OpenSIPS community you can learn about the hundreds of applications around the world. Check out this link to see just a few examples of who is using OpenSIPS:

https://www.opensips.org/About/WhoIsUsing

You can contact Bodgan-Andrei here, and OpenSIPS here.

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