Flexisip is a complete suite of modular and scalable SIP servers built around a central module, the SIP proxy. Since its launch in 2015, Flexisip has enabled companies to deploy SIP networks for voice/video/text communications over the Internet. Faced with the growing need of our customers to be able to interconnect their SIP networks with traditional telephony (known as PSTN), we decided to develop a new server component allowing the Flexisip proxy to be interconnected with a PSTN gateway provided by a third party operator. From a technical point of view, this new component is a back-to-back user agent (Flexisip B2BUA), which offers several applications including one that allows SIP trunking functionality (sip-bridge).
This new Flexisip module meets the different needs of our customers, developers of telephony solutions for companies, or call centres :
- It allows their users to be reachable on the SIP network via classic numbers (inbound calls), or to reach classic numbers with their SIP account (outbound calls)
- It makes it possible to affiliate a certain number of PSTN lines to a service, and pick from these available lines to receive incoming/outgoing calls from the company's network
- It enables a call to be transferred to a PSTN line when it cannot be completed via the SIP protocol for any reason, such as if the recipient does not have access to the Internet, the number entered by the caller corresponds to a standard number, etc.
These different rules allow you to determine when the call must be transmitted from the SIP proxy to the B2BUA and it can be configured according to your needs.
Please note that we do not yet offer Linphone users with a sip.linphone.org account the means to configure their PSTN line in the application. This is because we would have to work with a PSTN line provider, such as an Internet telephony service provider or carrier. On the other hand, we offer companies the possibility to deploy our software to provide this kind of functionality in their own Linphone-derived applications. If you are interested, then it will be up to you to purchase these PSTN lines from a third party provider. On our side, however, we can deploy a complete network including Flexisip proxy and Flexisip B2BUA and configure your PSTN lines in the B2BUA server for you.
Examples of use cases:
- Phone reception: allowing people outside the company network to use their standard phone number to contact your call centre or the staff connected to your SIP infrastructure.
- Unified communications: allowing your employees to use their Linphone-derived softphone to communicate with each other while benefiting from the app's advanced features (video conferencing, chat, etc.), and allowing them to contact standard numbers via this softphone.
- Intercom: as a company selling IP intercoms, you can offer your users a mobile application to answer and open the door via their smartphone. Thanks to Flexisip's SIP trunking feature, you can now relay the call to the user's standard number via the standard phone network.
The Flexisip B2BUA configuration procedure is available on our public Wiki. It indicates in particular:
- how to install this component and activate the SIP trunking mode (sip-bridge application)
- how to configure the authentication info of the PSTN gateway of your third party provider
- how to configure the call relay rules from the proxy to the B2BUA (forward module)
- how to configure the PSTN lines from which the B2BUA can pick when it needs a PSTN line for an outgoing call
The source code is publicly available on our Gitlab, in the Flexisip project.
Flexisip B2BUA is part of the official Flexisip releases since version 2.2.0. The list of packaged Linux distributions is available here.
If you wish to subcontract to us the installation of this server in your infrastructure or subscribe to a support contract, do not hesitate to consult our sales department via the contact form.