By using the sip:provider Community Edition, an open source VoIP soft-switch leveraging powerful and widely used open source components, we will in 15 minutes create a VoIP deployment from scratch in order to provide future-proof voice and video communication services, preceded by an introduction into the system architecture.
There are various open source components (e.g. Kamailio, Asterisk, Freeswitch and Sems) available to build a reasonably sized VoIP service from scratch. There are also some administrative web interfaces (e.g. Webmin, Siremis), allowing you to control the most basic things in such a system. And in order to escape the lab stage, you can even put an open source billing engine (e.g. a2billing) into the mix. Since your customers would like to manage their accounts and features, you will then develop a customer self-care interface as well. Time to market: at least two months. Feature set: basic. Service quality: uncertain at best. Future-proof: not. There is a solution to that though. During this talk, I will present how to deploy a solid VoIP Provider platform from scratch in just a few minutes, using the Sipwise sip:provider CE (SPCE) v2.4. The SPCE is a free and open source soft-switch based on Kamailio, Sems and Asterisk, providing fully featured and seamlessly integrated administrative and customer-self-care web interfaces, SOAP/XMLRPC provisioning APIs, a flexible rating engine and a huge load of subscriber features like conferencing, voicemail, call forwards, block lists etc. On reasonable hardware, the SPCE can serve 50k subscribers and more. As an introduction, I will outline the basic elements involved in a VoIP deployment. Then I will dive into the architecture of the SPCE, showing the building blocks and their interaction. Finally, I am going to do a live presentation on how to configure an SPCE instance for a typical deployment.
Speakers: Andreas Granig