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.
Tag: STUN
Kamailio DNS and NAT
DNS sub-system in Kamailio DNS failoverDNS load balancingNAT ( Network Address Translation)NAT ( Network Address Translation)Why is NAT is important in SIP?Types of NAT solutionsNAT behavioursRTP NATFixing NATNAT Traversal ModuleWhy use keepalive when Registrations are already there for NATing ?How keepalives work for NATing ?function nat_keepalive()ParamsFunctionsclient_nat_test()fix_contact()nat_keepalive()Pseudo VariablesStatisticsNATHelper ModuleNAT pinging typesUDP packetSIP requestparamsfunctionsPseudo VariablesRPC Commands In … Continue reading Kamailio DNS and NAT
Session Border Controller (SBC) for WebRTC
SBC became important part of comm systems developed over SIP and MGCP. SBC offer B2BUA ( Back to Back user agent) behavior to control both signalling and media traffic.
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
NAT traversal using STUN and TURN
STUN and TURN server protocols handle session initiations with handshakes between peers in different network environments . In case of a firewall blocking a STUN peer-to-peer connection, the system fallback to a TURN server which provides the necessary traversing mechanism through the NAT.
WebRTC Stack Architecture and Layers
WebRTC offers web application developers the ability to write rich, realtime multimedia applications (think video chat) on the web, without requiring plugins, downloads or installs. It's purpose is to help build a strong RTC platform that works across multiple web browsers, across multiple platforms.
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
Telecom R & D 