Chapter 9. Native SIP Application and Interaction with WebRTC Clients

With the passage of time, better standards for mobile phone networks emerged that which were much faster than their predecessors. This led to the widespread adoption of VoIP protocols to communicate between different kinds of devices over wireless networks. As WebRTC is meant to be used not only over a LAN but also over mobile data packet networks, it is crucial for all the functionality, performance, and interoperability scenarios to be ascertained. WebRTC is intended to be a homogeneous technology for every browser; however, there are marginal differences between the browser types/versions for devices and operating systems (OSs). Intercommunicating between a native SIP client and WebRTC is also a challenge, considering the format of Media codecs and signaling used. While native SIP phones use SRTP or RTP for media, G.7xx, AMR-xx, Speex, GSM audio codecs, and H.263 and H.264 video codecs, WebRTC offers SRTP as a video...