To address the need for interconnecting a SIP network for enriched communication (voice, video, instant messaging, and videoconferencing) with the traditional telephone network (Public Switched Telephone Network), we developed theFlexisip B2BUA module some time ago. This allows a user, for example, to call a contact on their landline or mobile phone anywhere in the world from their SIP account. It should be noted that our company does not provide PSTN lines, so it is necessary to turn to a third-party service.
Until now, only outgoing calls were possible. For more information, see the following article: Improvements for the transmission module to traditional telephony and Flexisip enables interconnection with PSTN networks.
Following the improvement made, the system now also works in the other direction: a user can be called on their SIP account from a regular phone number (incoming calls). This is possible thanks to a new mapping table between a SIP account and a virtual number. Updates to this mapping table (additions, deletions, and user account updates) are notified to the B2BUA without needing a restart.
Additionally, to adapt to growing services, the B2BUA server has been optimized to support significant load increases. Multiple instances of the B2BUA can be connected for large-scale networks with, for example, 30,000 users and 2,000 simultaneous calls.
For system administrators wishing to configure their Flexisip B2BUA, we provide detailed documentation: SIP Bridge – XWiki.
Have you already deployed Flexisip for its unified communication features? You can now add a feature for integration with traditional telephony.
