kamailio has a modular architecture with core components and modules to extend the functionality, this article will be discussing few of the essential modules in Kamailio.
Category: Session Initiation Protocol (SIP) Frameworks
Telephony Solutions with Kamailio
Rich features set suiting to telephony domain that includes IMS extensions for VoLTE; ENUM; DID and least cost routing; load balancing; routing fail-over; Json and XMLRPC control interface, SNMP monitoring. To integrate with a carrier grade telecom network as SBC / gateway / inbound/outbound proxy , it can act as IPv4-IPv6 gateway , UDP/TCP/SCTP/WS translator and even had NAT and anti DOS attack support .
Proxying Media Streams via Kamailio’s RTP Proxy
Kamailio is a SIP server which does not play any role by itself in media transmission path. this behaviour leads to media packets having to attempt to stream peer to peer between caller and callee which in turn many a times causes them to get dropped in absence of NAT management To ensure that media … Continue reading Proxying Media Streams via Kamailio’s RTP Proxy
Service Creation Environment (SCE ) for SIP Applications
Develop a SCE ( Service Creation Environment ) to addresses all aspects of lifecycle of a Service, right from creation/development, orchestration, execution/delivery, Assurance and Migration/Upgrade of services.
OfficeSIP
This post describes the installation , setup and configuration of Office SIP server to provide a registrar to our SIP based WebRTC application .
BEA Weblogic SIP server
Bea server is a old SIP servlet container ie application server which is used to embed control logic in a program. 1. Install Bea Weblogic 2. Follow the Installation steps Make domain 3. Goto the installation directory . Usually C:/bea/user_projects/mydomain/ . click on startweblogic.cmd in windows. In case the system is linux run startweblogic.sh script … Continue reading BEA Weblogic SIP server
SIP Presence
Subscribe SIP Notify SIP PUBLISH We have already learned about Sip user agent and sip network server. SIP clients initiates a call and SIP server routes the call . Registrar is responsible for name resolution and user location. Sip proxy receives calls and send it to its destination or next hop. Presence is user’s reachability … Continue reading SIP Presence
Interoperability between WebRTC, SIP phones and softphones
SIP Server convert the SIP transport from WebSocket protocol to UDP, TCP or TLS which are supported by all legacy networks. It also facilitates the use of rich serves such as phonebook synchronisation , file sharing , oauth in client .
Kamailio Call routing and Control
Kamailio SIP server evolved from SER and OpenSER. Written in ANSI C , primarily it is an open source proxy SIP server. RFC 3261 compliant and has support for various Operating system to install and run on as alpine , centos , deb , fedora , freebsd , netbsd , obs , openbsd , opensuse … Continue reading Kamailio Call routing and Control
SIP and SDP Messages Explained
SIP is a widely adopted application layer protocol used in VoIP calls and confernecing applciations and in IMS architeture or pure packet switched networks . More on SIP , its packet structure , transaction and dialogs , loose and strict record routing , location service , near and far end nating , and commonly used … Continue reading SIP and SDP Messages Explained
Kamailio Transaction management and Transaction Module tm
Kamailio is basically only a transaction stateful proxy, without any dialog support build in. Here the TM module enables stateful processing of SIP transactions. State is a requirement for many complex logic such as accounting, forking , DNS resolution
RCS ( Rich Communication Suite )
RCS ( Rich Communication Suite ) What is special about RCS ? RCS releases RCS APIs RCS e ( enhanced ) What is the difference between RCS-e and RCS? RCSe Features RCS-e Customer Value Proposition RCSe Characteristics RCSe Release WebRTC/RCS Client WebRTC RCS Addon Features IMS integartion with RCS Monitizations with RCS Context based communications … Continue reading RCS ( Rich Communication Suite )
JAINSLEE – Developer and business benefits
JAIN SLEE is the Java open standard for a SLEE ( Service Logic Execution Environment ). It is a Java programming language API for developing and deploying network services. Evolution of Open- Standard Platform (JAINSLEE) There is a strong evolution being seen in CSP space. Now operators are looking forward to implement the open standard for intelligent networks. It reduces their dependency on … Continue reading JAINSLEE – Developer and business benefits
JAIN SLEE
•Jain SLEE :- JAIN is a Sun Java standards initiative and part of the Java Community Process. JAIN specifies a comprehensive range of APIs that target converged IP and PSTN networks, including APIs for - High-level application development (such as service provider APIs and the Service Logic Execution Environment (SLEE)) - call control - signalling … Continue reading JAIN SLEE
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 