Analysis and testing of voip-subsystems of Ip Brick

Technology in which communication done using IP(Internet Protocol) alternative for the traditional analog systems is voip (voice over internet protocol) .One of the emerging or attractive communication systems in this era is voip. Several technologies within the voip are emerging and more come to near future, services offered by this technology needs internet connections and/or telephone connections. It offers services such as making audio and video calls. VoIP is a medium that converts the analog signal to digital signals[1]. In this dissertation mainly focused on ipbrick voip-subsystems such as webrtc ,asterisk13.8 pbx server and kamilio sip proxy. These are free and open-source technologies available in the marketplace. Integration of these technologies provides real-time communication between the systems such as voice,video and text. Webrtc enables web browsers to have native support for real-time voice, video and data capabilities. As such, end-users do not require additional add-ons or plug-ins to utilize real-time voice and video communication. To allow webrtc to make calls to non-webrtc voip applications, A initiation protocol is needed and one such protocol is session initiation protocol (SIP),which is the standard protocol used for initializing, changing and terminating sessions for multimedia today. It is particularly known for its use in voip applications. Asterisk13.8pbx supports webrtc and it acts as media gateway. In this thesis, we are evaluating and comparing previous and current versions of the ipbrick voip-subsystem which is previously having lab environment of ipbrick OS v6.1 with Asterisk v1.8 as a pbx server and webrtc application such as webrtc2sip and SIP Proxy software called Kamailio which connects two endpoints. In this thesis we are testing ipbrick voip-subsystem with current version of ipbrick OS v6.2 with Asterisk v13.8 as a new pbx server and Webrtc application (SIPML5) on the browser side and a phone on the server side (softphone) establish a phone call between them. The SIP Proxy server Kamailio will act as a intermediary connection, connecting the two endpoints using websockets