JavaScript Session Establishment Protocol (JSEP) in WebRTC handshake

SEP is used during signalling via w3c's recommended RTCPeerConnectionAPI interface to set up a multimedia session. The multimedia session description specifies the critical components of setting up a session between local and remote such as transport ports, protocol, profiles. It also handles the interaction with the ICE state machine.

TURN server for WebRTC – RFC5766-TURN , Coturn, Xirsys , Twillio

STUN (Session Traversal Utilities for NAT) and TURN (Traversal Using Relays around NAT) are protocols that can be used to provide NAT traversal for VoIP and WebRTC. These projects provide a VoIP media traffic NAT traversal server and gateway. TURN Server is a VoIP media traffic NAT traversal server and gateway. This article describes working … Continue reading TURN server for WebRTC – RFC5766-TURN , Coturn, Xirsys , Twillio

SIP VoIP system architecture basics

Infrastructure RequirementsIntegral Components of a VOIP SIP based architectureSIP GatewaysRegistrar ServerProxy ServerRedirect ServerApplication Server Adding Media ManagementDTMF( Dual tone Multi Frequency )TTS ( Text to Speech ) Developing SIP based applicationsBasic SIP methodsExtending SIP headersCall routing ScriptsSIP platform DevelopmentCollecting and Processing PCAPS NAT and DNS Near-end NAT traversal Far-end NAT traversalCDR Processing and BillingData Streams … Continue reading SIP VoIP system architecture basics