In the era of WebRTC, incredible efforts are being made to bring Real-Time Communications on every web browser. Hundreds of millions of users are one click away from a high quality videochat session with friends or customers.

 

Yet, pure peer-to-peer communications between browsers can be hard to achieve (depending on network configurations) or undesirable (e.g. if you need to record conversations or provide value added services).

 

Furthermore, there are many scenarios where users are reachable only via GSM, and you need to plan the interoperation between Web and Telephony worlds.

 

RTCSoft designs and implements solutions tailored to your needs, but using extensively best-of-breed Open Source applications like Kamailio, FreeSWITCH, Asterisk, ejabberd, node.js. This minimizes the time from idea to running service, while at the same time bringing flexibility, extensibility and ease of interoperation.

 

Want to know more? Interested in discussing your idea? gv@rtcsoft.net