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
Author: altanai
WebRTC Media Streams and Quality metrics
Media Stream Tracks in WebRTCVideo StreamsVideo Capture insync with hardware's capabilitiesCapture ResolutionSDP attributes for resolution, frame rate, and bitrateDynamic FPS control based on actual hardware encodingStream OrientationAudio StreamsAudio LevelGAIN calculationAcoustic Echo Cancellation (AEC)SDP signaling and negotiation for media planeMedia SourcePeer-to-Peer Media StreamFramesPacketsBytesHeadersPeer-to-Peer Data TransferBitratePacket LossJitter Round Trip Time Media Stream Tracks in WebRTC The MediaStreamTrack interface … Continue reading WebRTC Media Streams and Quality metrics
Regulatory/Legal Considerations and CALEA with WebRTC development
This post is deals with some less known real world implication of developing and integrating WebRTC with telecom service providers network and bring the solution in action . The regulatory and legal constrains are bought to light after the product is in action and are mostly result of short sightedness . The following is a … Continue reading Regulatory/Legal Considerations and CALEA with WebRTC development
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
Legacy Telecom Networks
Characteristics of Legacy Systems PSTN (Public Switched Network) TDM ISDN (Integrated Service Digital Network) Services of Legacy Telecom Networks I use the term legacy telecom system many a times , but have not really described what a legacy system actually is . In my conferences too I am asked to just exactly define a legacy … Continue reading Legacy Telecom Networks
SIP/VOIP transformation towards IMS (Total IP)
Upgrading a softswitrch solutions to IMS Intelligent Networks( IN) Fixed/mobile convergence(FMC) with IMS Legacy to IP transformation WebRTC based Unified Communication platform Challenges in Migration to IMS (Total IP ) The telecommunications industry has been going through a significant transformation over the past few years. At the outset incumbent operators used to focus on mainly basic … Continue reading SIP/VOIP transformation towards IMS (Total IP)
Evolution of voice Communication
This post describes the evolution of voice communication in access ,transport and session layers respectively.
Service Broker Architecture for IN and IMS
Service BrokerAdvantages of using Service Broker vs Total Migration from In to IMSProvisioning via fixed/mobile brands «service profile» in Service BrokerArchitecture of SDP / Service Broker Service Broker We know that a Service broker is a service abstraction layer between the network and application layer in a telecom environment. SB( Service Broker ) can enable … Continue reading Service Broker Architecture for IN and IMS
WebRTC compatible android client
This post describes the requirement of creating a SIP phone application on android over the same codecs as WebRTC ( PCMA , PCMU , VP8) . In my project concerning the demonstration of WebRTC inter operability ( presence , audio / video call , message ) with a native android client , I had to … Continue reading WebRTC compatible android client
2G to 3G – generation of telecom
Where 2G is referred to as the GSM era , 2.5 G as the GPRS with GSM era. As compared to its predecessor 1G which used FDMA ( Frequency Division Multiplexing ) for channelization , 2G used used TDMA and CDMA for dividing the channels .
Difference between WebRTC and plugin based communication
A lot of service providers ie telecom operators had deduced their own ways to provide Web based communication even before WebRTC was born . With time , as WebRTC has become stronger , more secure , resilient to failure they have come around to migrate their existing system from previous closed box native APIs to … Continue reading Difference between WebRTC and plugin based communication
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 .
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 .
