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
Tag: NAT
Asterisk – installation and dial plans for WebRTC supported PJSIP clients
Asterisk is a framework or toolkit designed for VOIP systems . It can support Enterprise communication systems like PBXs, call distributors, VoIP gateways , conference bridges etc . It is open source and free to use . It is developed in C and runs in linux . Technically , Asterisk has protocol support for many … Continue reading Asterisk – installation and dial plans for WebRTC supported PJSIP clients
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.
Telecom R & D 