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 .
Author: altanai
Specialized in CPaaS, carrier-grade WebRTC-SIP telecom platforms for Unified communication-collaboration, signaling gateways, SBC, soft turrets, IoT-surveillance and telecom integrations.
Ardent contributor to Open Source software, avid freelancer, innovator and technical writer (telecom.altanai.com ).
Inventor of "RamuDroid" an IOT Road-Cleaning robot
Author of book "WebRTC Integrator's Guide" published by Packt
Internet Telephony Convergence- JAINSLEE Platform
Convergence : Telephone networks and computer networks converging into single digital network using Internet standards. Components in a Network Client computer Server computer Network interfaces (NICs) Connection medium Network operating system Hub or switch Routers- Device used to route packets of data through different networks, ensuring that data sent gets to the correct address Figure :simple computer … Continue reading Internet Telephony Convergence- JAINSLEE Platform
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
Business Challenges for a telecom service provider
With the fast pace of telecom evolution both towards the access network front ( ie GSM , UMTS , 3G , 4G , LTE , VOLTE ) to core network side ( ie application servers , registrar , proxies , gateway , media server etc ) a CSP ( content service provider ) is trying hard to keep up with the user expectation . The user expects a plethora of services , reduced cost and high speed bandwidth . If this was not enough a CSP also has competition OTT ( Over The Top ) Players who provide communication and messaging for FREE .
WebRTC web service to PSTN – diagram
IMSSF and RIMSSF
This post particularly describes the gateways in IMS which communication back and forth with a legacy endpoints.To get a overview of IMS itself click here and to get a detailed description of IMS and its architecture click here . What is IM-SSF ? IP Multimedia Service Switching Function is a gateway to provide IN service such … Continue reading IMSSF and RIMSSF
Realtime Communication (RTC) platform
How easier would it be if there was an infrastructure that could connect all devices all communication protocols, old legacy PSTN phones, VOIP ones like SIP all the PBXs of the world, GSM handsets even the very new WebRTC! I have assembled my thoughts on an ideal communication platform in this blog post. Functional Req … Continue reading Realtime Communication (RTC) platform
Tools for a Telecom software Engineer
Evernote for notekeeping Eclipse to do real programming Github to upload download code MySQL workbench to take care of Database Management Technologies to Work with IETF W3C WebRTC HTML Java GSMS standards Frameworks Struts Hibernate Spring EJB
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
WebRTC communication diagrams
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
VoIP/ OTT / Telecom Solution startup’s strategy for building a scalable flexible SIP platform
Scalable and Flexible platform. Let's go in-depth to discuss how can one go about achieving scalability in SIP platforms. ulti geography Scaled via Universal Router, Cluster SIP telephony Server for High Availability, Multi-tier cluster architecture, Role Abstraction / Micro-Service based architecture, uted Event management and Event Driven architecture , Containerization, autoscaling , security , policies and market differentiator, ticketing and issue tracking.
E-Learning
e-learning platform which harness the power of Internet for the purpose of distance education and where students around the world volunteer to teach each other any subject they wish to. This will be made possible through a combination of real time communication technologies like WebRTC and plethora of knowledge repositories.
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 )
WebRTC SIP / IMS solution
We started in winters on 2012 with Webrtc . At time time it just looked like a new tech jargon that might fade away when new ones comes . In many many WebRTC's buzz has died down since its massive adoption. But i nevertheless still see a lot of potential and development around it. What … Continue reading WebRTC SIP / IMS solution
