Category: SIP servers

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

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

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

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

Mobicents SIP server platform

We know that SIP is in the p2p session layer of the OSI mode and used to setup voip sessions and that a SIP Servlets must be executed within a SIP Servlets Container, which implements the SIP Servlet specification. Mobicents sip servlets have been extensively used to create , deploy and manage VOIP services. Also … Continue reading Mobicents SIP server platform

Freeswitch Integration with Telecom Carrier

This articles is a follow up of the earlier freeswitch capability introduction and some generic usecases around dialplans and contexts. This post contains instructions on how to integrate your SIP VOIP Freeswitch server to ITSP ( Internet Telephony Service Providers) which are basically part of large telecommunications companies. First we check the external profile via … Continue reading Freeswitch Integration with Telecom Carrier

Freeswitch Modules

This section describes some of the popular and useful freeswitch module. Although there are many more modules, I have picked a few of commonly used one and divided them into categories Logger modules in FreeswitchXML Interfaces in FreeswitchEvent system and Event Handlers in Freeswitchmod_amqpmod_cdr_csvDataBaseApplicationsInfo, Intercept and eavesdropChannel operationsDialplan Tools ( DPTools)WaitSchedulePlayPreprocessRecordsayTimeLimitDTMFAPISocketLanguagesJitterBufferASR/TTSDialplan InterfacesCodec InterfacesFile Format Interfaces … Continue reading Freeswitch Modules

FreeSwitch SIP and Media Server

Architecture and Design of FreeswitchCoreThreaded Model State Machine in Freeswitch CoreChannel VariablesDialplanSpeak Time and Date on CallCall Routing based on destination number and forwarding to voice mail on no answerCall routing based on day and timeMatch incoming network IP address with pre configured IPStore captured values in standard variables PlaybackMedia recording and playback in audio (wav)Routing by … Continue reading FreeSwitch SIP and Media Server