WebRTC Live #70: Using Kamailio to Connect WebRTC to SIP and PSTN

Kamailio is an Open Source SIP Server released under GPLv2+, able to handle thousands of call setups per second. Kamailio can be used to build large platforms for VoIP and real-time communications - presence, WebRTC, instant messaging and other applications; allowing you to easily bridge WebRTC endpoints to the PSTN, enterprise, and more. Our guest, Fred Posner, will be discussing bridging WebRTC to SIP via Kamailio and use cases such as call centers, remote workers, PSTN connectivity, and more.
Fred Posner has been working with Kamailio for more than 15 years. He loves to show how Kamailio can help secure and enable communication using well deployed open source software. Fred actively participates in the VoIP Community, including Kamailio World, AstriCon, ClueCon, and other groups providing support for Open Source and branded communication software.

[@JohnJohn-sf1df]

Good stuff!

[@dt7472]

Thank you!