Session Initiation Protocol (SIP)

SIP Presence

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 and willingness to communicate its … 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 .

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

SIP VoIP system Architecture

SIP solutioning and architectures  is a subsequent article after SIP introduction, which can be found here. A VOIP Solution is designed to accommodate the signalling and media both along with integration leads to various external endpoints such as various SIP phones ( desktop, softphones , webRTC ) ,  telecom carriers  , different voip network providers  … Continue reading SIP VoIP system Architecture

SIP in IMS

A diagrammatic layout of the nodes , interwokring among them and involvment of SIP in the different planes of  IMS architecture .

JSR 116 – SIP SERVLET 1.0

SIP Servlet 1.0 API •JSR 116 •Built into the Servlet container that also hosts  portlets and HTTP Servlets. •SIP Servlet API developed under the JCP (Java Community Process) as JSR 116 (Java Specification Request), as a set of neutral interfaces Servlet Container •Environment in which a servlet can exist •Loads and initializes a servlet •Invokes … Continue reading JSR 116 – SIP SERVLET 1.0

Service Creation Environment (SCE ) for SIP Applications

Hosted IP-PBX and its SBC

SIP Servlets

JAINSLEE

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

Features set JAINSLEE vs SIP/J2EE

Feature Set JAINSLEE vs SIP/J2EE Portability Portability of JAINSLEE is limited to number of available applications servers on the market. Complexity 1) SIP Servlet components handle directly SIP signaling, there is no abstraction layer so there is no loss in network features. 2) If a comparison between SIP Servlets and JAIN SLEE is made it … Continue reading Features set JAINSLEE vs SIP/J2EE

What is JAIN SLEE ?

PSTN/2G/3G/4G to IMS – Internet Telephony Converged Platform

SIP Security

Attacks on SIP Networks

Major standards bodies including 3GPP, ITU-T, and ETSI have all adopted SIP as the core signalling protocol for services such as LTE , VoIP, conferencing, Video on Demand (VoD), IPTV (Internet Television), presence, and Instant Messaging (IM) etc. With the continous evolution of SIP as the defacto VoIP protocol , we need to underatdn the … Continue reading Attacks on SIP Networks

Certificates, compliances and Security in VoIP

This article describes various Certificates and compliances, Bill and Acts on data privacy, Security and prevention of Robocalls as adopted by countries around the world pertaining to Interconnected VoIP providers, telecommunications services, wireless telephone companies , HIPPa , SOX , GDPR , COPPA , CPNI , CCPA , PDP,SPIT ,Traced ACT , CRTC , Fcc E911

CLI/NCLI, Robocalls and STIR/SHAKEN

To understand the need for implementing an identification verification technique in Internet protocol based network to network communication system , we need to evaluate the existing problem plaguing the VoIP setup . What is Call ID spoofing ?  Vulnerability of existing interconnection phone system which is used by robo-callers to mask their identity or to … Continue reading CLI/NCLI, Robocalls and STIR/SHAKEN

Secure Communication with SRTP and key managemnt protocols like SDES , ZRTP and DTLS

With advent of Voice over IP , the real time streaming of data/audio/video also became critically important to be protected from eavesdropping or modification over the open internet. While Secure Real-time Transport Protocol (SRTP) is a profile of the Real-time Transport Protocol (RTP), which can provide confidentiality, message authentication, and replay protection to the RTP … Continue reading Secure Communication with SRTP and key managemnt protocols like SDES , ZRTP and DTLS

SIP servers

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 . It is supported on jdk1.5 hence the system’s environment variables must match . Otherwise in later stages deploying applications throw class version error . 1. Install Bea Weblogic 2. Follow the Installation steps … Continue reading BEA Weblogic SIP server

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

sipP ( SIP testing tool )

Kamailio DNS and NAT

DNS sub-system in Kamailio To resolve hostname into ips it can do either of below use libresolv and a combination of the locally configured DNS server /etc/hosts and the local Network Information Service (NIS/YP a.s.o) or cache the query results and first look into internal cache DNS failover – if destination resolves to multiple addresses … Continue reading Kamailio DNS and NAT

Kamailio WebRTC SIP Server

The purpose of this article if to demo the process of using Kamailio + RTP Engine to enable SIP based WebRTC call to a traditional SIP UA like Xlite. Kamailio Will thus provide not only call routing but also NATing , TLS and websocket support for webrtc endpoints. For this bridging of SRTP from WebRTC … Continue reading Kamailio WebRTC SIP Server

Kamailio Transaction management

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 . Branches – A single SIP INVITE request may be forked to multiple destinations , all … Continue reading Kamailio Transaction management

Lua Scripts for kamailio Routing

Kamailio uses a native scripting laguage for its configuration file kamailio.cfg . This components of this file are : global parameters loading modules module parameters routing blocks like request_route {…}, reply_route {…}, branch_route {…} etc These parameters including initialization event routes , are interpeted and loaded at kamailio startup. We know that restart of sip … Continue reading Lua Scripts for kamailio Routing

RTP engine on kamailio SIP server

RTPengine is a proxy for RTP traffic and other UDP based media traffic over either IPv4 or IPv6. It can even bridge between diff IP networks and interfaces . It can do TOS/QoS field setting. It is Multi-threaded , can advertise different addresses for operation behind NAT.

Kamailio Security

Kamailio security modules , Sanity , permission , topos , ACL , Fireqall , anti flood,s ecfilter module

Kamailio as Inbound/Outbound proxy or Session Border Controller (SBC)

role of an SBC is to shield the core network from external entities such as user agent’s , carrier network while also providing security , auth and accounting services . In many cases SBC also provides NAT traversal and policy control features ( such as rate limiting , ACL etc ) . In advanced cases transcoding, topology concealment and load balancing is also achievable via a SBC such as Kamailio .

Kamailio Architecture , Core and Modules

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.

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 .

Kamailio Transaction Module

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

Freeswitch PBX system

This article talks about setting up an in-house hosted Enterprise PBX system for sure and private communication within enterprise communication. FreeSWITCH is free and open source communications software licensed under Mozilla Public License. It if often the core of voice core to provider call routing and media control . Its core library, libfreeswitch, is capable of being embedded into other projects, … Continue reading Freeswitch PBX system

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 following categories : Loggers XML Interfaces Event Handlers Application Language ASR/TTS Loggers mod_console mod_graylog2 mod_logfile mod_syslog mod_yaml Multi-Faceted mod_enum is a dialplan interface, … Continue reading Freeswitch Modules

FreeSwitch SIP and Media Server

FreeSWITCH is free and open source communications software licensed under Mozilla Public License. It if often the core of voice core to provider call routing and media control . Its core library, libfreeswitch, is capable of being embedded into other projects, as well as being used as a stand-alone application.  FreeSWITCH is designed to route and … Continue reading FreeSwitch SIP and Media Server

Opensips

Due to its very flexible and customisable routing engine it can be used in number of scenarios such as an SIP proxy or a router and due to its high throughput it is widely recommended as an enterprise grade inbound/outbound proxy server.

Asterisk – dialplans

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 – dialplans

WebRTC

JavaScript Session Establishment Protocol (JSEP) in WebRTC handshake

This article is aimed at explaning the intricacies and detailed offer answer flow in webrtc handshake and JSEP . You can read the following artciles on WebRTC as prereq before reading through this one WebRTC API – Peerconnection , getUserMedia , Datachannel , DataStaats JSEP (JavaScript Session Establishment Protocol) JSEP (JavaScript Session Establishment Protocol) is … Continue reading JavaScript Session Establishment Protocol (JSEP) in WebRTC handshake

SIP IMS and WebRTC

WebRTC SIP & IMS Solution

WebRTC call between browser and SIP softpphone

STUN and TURN

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.

WebRTC security

WebRTC Security

WebRTC Security
Identity Management ,
Browser Security ,
Authentication and
Media encryption.
Browser Threat Model
Best practices for the Webrtc comm agents
ICE TURN challenges
DTLS
SRTP

Performance of WebRTC sites and apps

As security is a broad topic touching on many sections of WebRTC this section is not meant to address all topics but instead to focus on specific “hot spots”, areas that require special attention due to the unique properties of the WebRTC service. —There are several security related topics that are of particular interest with respect to WebRTC. They can be grouped into the following areas: Identity Management
Browser Security
Authentication
Media encryption
Syntax checks using regex

WebRTC APIs

APPRTC , Talky.io , TokBox

Augmented Reality

Three.Js

WebGL , Three.js and WebRTC

WebRTC Software as a Service SaaSwebrtc_development_logo

WebRTC CPaaS ( Communication Platform as a Service )

CPasS ( communication platform as a service ) is cloud based communication platform that provides real time communication capabilities. This should be easily integrable with any given external environment or application of the customer, without him worrying about building backend infrastructure or interfaces .

Session Border controller for WebRTC

SBC became important part of comm systems developed over SIP and MGCP. SBC offer B2BUA ( Back to Back user agent) behavior to control both signalling and media traffic.

Setting up ubuntu ec2 t2 micro for webrtc and socketio

Setting up a ec2 instance on AWS for web real time communication platform over nodejs and socket.io using WebRTC . Primarily a Web Call , Chat and conference platform uses WebRTC for the media stream and socketio for the signalling . Additionally used technologies are nosql for session information storage , REST Apis for getting sessions details to third parties.

Steps for building and deploying WebRTC solution

Error in connectivity , errors in console , blank video are the problems that might appear . So well err things begin to get a bit complicated from here . To bypass network firewalls , corporate net policies , UDP blocks and filters we require a TURN server .

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

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TFX Widgets Development

TangoFX Sessions is a customizable solution where developers can create and add their own widget over the underlying WebRTC communication mechanism . It can support extensive set of user activity such as video chat , message , play games , collaborate on code , draw something together etc . It can go as wide as your imagination .

TFX WebRTC SaaS (Software as a Service )

TFX sessions is a part of TFX . It is a free Chrome extension WebRTC client that enables parties communicating and collaborating, to have an interactive and immersive experience. The 3 possible approaches for TFX Integration in increasing order of deployment time are :

WebSite’s widget on TFX chrome extension .
Launch TFX extension in an independent window from website
TFX call from embedded Window inside the website page

TFX platform

So I haven’t written anything worthy in a while , just published some posts that were lying around in my drafts . Here I write about the main thing . some thing awesome that I was trying to accomplish in the last quarter . << TFX is now live in chrome store , open and … Continue reading TFX platform

Setting up ubuntu ec2 t2 micro for webrtc and socketio

WebRTC Media Stack

WebRTC Audio/Video Codecs

Codecs signifies the media stream’s compession and decompression. For peers to have suceesfull excchange of media, they need a common set of codecs to agree upon for the session . The list codecs are sent  between each other as part of offeer and answer or SDP in SIP. As WebRTC provides containerless bare mediastreamgtrackobjects. Codecs … Continue reading WebRTC Audio/Video Codecs

Janus MCU

Janus MCU for WebRTC broadcasting and multi party conference

WebRTC service’s

WebRTC communication over Web Services

Call Continuity from Mobile GSM network to WebRTC

E-Learning

Internet Of Things ( IOT ) iotlogo

 BLE

BLE app for cypress BLE

raspberry-pi-logo

Remote machine control via Rpi

IOT Surveillance with Arduino + Rpi +WebRTC

MQTT

Paho Android MQTT lib

Moska Mqtt Server

LoRA

Wifi and WiMax

Wifi 6

RFID

RFID- Radio Frequency Identification

Automatic identification method, relying on storing and remotely retrieving data using devices called RFID tags or transponders. An RFID tag is a small object that can be attached to or incorporated into a product, animal, or person. RFID tags contain silicon chips and antennas to enable them to receive and respond to radio-frequency queries from an RFID transceiver. Passive tags require no internal power source, whereas active tags require a power source.

Robotics

Building a  Robot with IOT principles

Remote machine control via Raspberry pi

Copy of Ramudroid Logo White (1).jpg

RamuDroid

Bot to clean roads and outdoors for a better and cleaner India. It lifts up small objects like plastic cups,wrappers,leaves etc. The droid also provides real-time camera stream and detects obstruction to re-route itself. It can communicates over GSM ,wifi and BLE . It can also be remote navigated via browsers or android. Working : … Continue reading RamuDroid

Media Server LogoMedia Archietcture

Live streaming & Broadcasting

Audio and Acoustic Signal Processing

Audio signals are electronic representations of sound waves—longitudinal waves which travel through air, consisting of compressions and rarefactions and Audio Signal Processing focuses on the computational methods for intentionally altering auditory signals or sounds, in order to achieve a particular goal. Application of audio Signal processing in general storage data compression music information retrieval speech processing … Continue reading Audio and Acoustic Signal Processing

SIP conferencing and Media Bridges

SIP is the most popular signalling protocol in VOIP ecosystem. It is most suited to a caller-callee scenario , yet however supporting scalable conferences on VOIP is a market demand. It is desired that SIP must for multimedia stream but also provide conference control for building communication and collaboration apps for new and customisable solutions. … Continue reading SIP conferencing and Media Bridges

crtmpserver + ffmpeg

This post will show the process of installing , running and using crtmpserver on ubuntu 64 bit machine with gstreamer . gcc and cmake We shall build gstreamer directly from sources . For this we first need to determine if gcc is installed on the machine . If not installed then  run the following command GNU Compiler Collection … Continue reading crtmpserver + ffmpeg

Wowza RTMP Authentication with Third party Token provider over Tiny Encryption Algorithm (TEA)

this article is focused on  Wowza RTMP Authentication with  Third party Token provider over Tiny Encryption Algorithm (TEA)  and  is a continuation of the previous post about setting up a basic RTMP Authentication module on Wowza Engine above version 4. The task is divided into 3 parts . RTMP Encoder Application Wowza RTMP Auth module … Continue reading Wowza RTMP Authentication with Third party Token provider over Tiny Encryption Algorithm (TEA)

Wowza RTMP Authenticate Module

To purpose of the article is the use the RTMP Authentication Module in wowza Engine .  This will enable us to intercept a connect request with username and password to be checked from any outside source like – database , password file , third party token provider , third party oauth etc.  Once the password … Continue reading Wowza RTMP Authenticate Module

WebRTC Live Stream Broadcast

WebRTC has the potential to drive the Live Streaming broadcasting area with its powerful no plugin , no installation , open standard  policy . However the only roadblock is the VP8 codec which differs from the traditional H264 codec that is used by almost all the media servers , media control units , etc . … Continue reading WebRTC Live Stream Broadcast

Live 555 media Server

Wowza Media Server

Wowza REST APIs and HTTP Providers

This article show the different ways to make calls to Wowza Media Engine from external applications and environments for various purposes  such as getting server status , listeners , connections , applications and its streams etc . HTTP Providers HTTP Providers are Java classes that are configured on a per-virtual host basis. Some pre packaged … Continue reading Wowza REST APIs and HTTP Providers

Wowza Secure URL params Authentication for streams in an application

To secure the publishers for a common application through username -password specific for streamnames , this post is useful . It uses Module Core Security to prompt back the user for supplying credentials.

check the rtmp query-string for parameters and performs the checks – is user is allowed to connect and is user allowed to stream on given streamname is given below .

WebRTC Live Stream Broadcast

Wowza RTMP Authentication Module

Wowza RTMP Authentication with Third party Token provider over Tiny Encryption Algorithm (TEA)

Natural Language Processing

NLP ( Natural Language Processing ) in VoIP

NLP has great potential in cognitive and artificial intelligence , but also with increasing human to machine interaction and enhancement in Machine learning ,NLP is set to revolutionize the Voice over IP space.

Codecs

Standards and Protocols

Rich Communication Services ( RCS)

Protocol libs for signalling

PJSIP

SIP stack written in C. Available under GPL PJSip user agent Attributes: local_info+tag, local_contact, call_id Operations: pj_status_t pjsip_ua_init(endpt, param); pj_status_t pjsip_ua_destroy(void); pjsip_module* pjsip_ua_instance(void); pjsip_endpoint* pjsip_ua_get_endpt(ua); PJSip dialog Attributes: state, session_counter, initial_cseq, local_cseq, remote_cseq, route_set, local_info+tag, local_contact, remote_info+tag, remote_contact, next_set Operations: pj_status_t pjsip_dlg_create_uac(ua, local_uri, contact, …); pj_status_t pjsip_dlg_create_uas(ua, rdata, contact, &dlg); pj_status_t pjsip_dlg_fork(old_dlg,rdata,&dlg); pj_status_t pjsip_dlg_set_route_set(dlg, route_set); … Continue reading PJSIP

H 323

International standard for multimedia communication ( real-time voice, video, and data communication) over packet-switched networks includes the standards H.323, H.225.0, H.245, the H.450-series documents, and the H.460-series. T.120 for data collaboration and file transfer. Ability to utilize network services for address resolution, including ENUM, LDAP, and DNS an provide various supplementary services Call transfer , … Continue reading H 323

Auxiliary Technologies in VoIP

VOIP Call Metric Monitoring and MOS ( Mean Opinion Score)

Metrics for monitoring a VOIP call can be obtained from any node in media path of the call flow . Essentially used for analysis via calculation and aggregation , and sometimes used for realtime performance tracking and rectification too .

Rating Factor (R-Factor) and Mean Opinion Score (MOS) are two commonly-used measurements of overall VoIP call quality.

R-Factor: A value derived from metrics such as latency, jitter, and packet loss per ITU‑T Recommendation G.107. It assess the quality-of-experience for VoIP calls on your network. Typical scores range from 50 (bad) to 90 (excellent).

BlockChain programming

Market analysts and industry specialist have said that block-chain is a revolutionizing technology which will create a decentralized network for not just currency exchange but also many other aspects such as double spent problem , universal identities , document management etc . Example : Bitcoin protocol , which contains a full record of every transaction ever executed with the currency at any time in past. It is also a solution to problem like black – money , double spending , tax evasions etc. Other areas include:

Extensible Messaging and Presence Protocol ( XMPP) 

XMPP Client Server Setup and Programming

XMPP is a open XML technology for real-time communication. Applications are instant messaging, presence, media negotiation, whiteboarding, collaboration, lightweight middleware, content syndication, and generalized XML routing according to XMPP standards Foundation (XSF) . installation steps for openfire on Ubuntu version 15 64 bit system

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Telecom Architectures

IP Multimedia Subsystem

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

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

IP Multimedia Subsystem ( IMS )

 IMS is a an architectural framework for IP based multimedia rich communications. It was standardized by a group called 3GPP formed in 1999. It started as an enabler for 3rd generation mobile networks in European market and later spread to wirelne networks too . IMS became the key to  Fixed Mobile Convergence (FMC). Based on IETF … Continue reading IP Multimedia Subsystem ( IMS )

IMS in EPC ( Evolved Packet Core )

Packet Switched and/or Circuit swicthed Communication The earlier models were distributed between legacy circuit switched networks and evolving packet switched networks With the massive improvents in quality of network srevices packet switched comunication protocls became more resilent and replaced the circuit swicthed protcols for realtime communication. LTE ( Long Term Evolution ) LTE evolved its … Continue reading IMS in EPC ( Evolved Packet Core )

Unified Service Delivery Stack

The Unified Communication Solution leads to Network Agnostic, Agile, Cost Effective  & Customer Experience Centric Services Platform.   The Way from Copper -> Fiber -> 2nd Generation -> 3rd Generation -> LTE , depicts evolution of Telecommunications over the decades , in the Network layer Infrastructure area The Sevice Layer Infrastructure is built  using techniques … Continue reading Unified Service Delivery Stack

IMS , the revolution ahead

vision : To make a model that separates the services offered by fixed-line (traditional telcos),                mobile (traditional cellular),            and            converged service providers (cable companies and others who provide triple-play — voice,  video, and data — services) from the access networks used to receive those services. … Continue reading IMS , the revolution ahead

Telecommunications convergence

The signalling protocols migration like from signalling system 7 (SS7) to session initial protocol (SIP) have been taking place in Telco-Industry. Similarly nodes of legacy network like signal transfer point (STP) of legacy network are being migrated to call session control function (CSCF) of IMS that allows the rapid development and deployment of enhanced, revenue-generating multimedia services for fixed, mobile and cable operators.

IMS architecture enables operators to seamlessly run a plethora of next-generation converged services over their fixed, mobile and cable networks, achieve a faster time-to-market for new services and have fewer performance bottlenecks.

VoIP system DevOPS, operations and Infrastructure management Automation

Legacy telecom

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 system . Often my clients are surprised to hear what they have in current operation is actually fitted in … Continue reading Legacy Telecom Networks

VPN

VPN ( Virtual Public Network ) over SIP

VOIP across an SSL-based VPN is achieved in good quality by encapsulating the UDP VOIP packets ( SIP and RTP ) in TCP/IP .

Data used for defining a VPN like its Groups, its Members and the associated profiles is organized hierarchically.It includes information like who is the operator, subscriber of VPN, group ID and member ID.

Telecom Info

WebRTC APIs

Contents Media Capture and streams Peer to peer Connection- RTCPeerConnection, RTCConfiguration, ICE, Offer/Answer, states RTP Media API RTCRtpSender RTCRtpReceiver RTCRtpTransreciver RTCDtlsTransport RTCIceCandidate ICE gathering RTCIceTransport Interface Peer-to-peer Data API Peer-to-peer DTMF Statistics Peer-to-peer connections creates p2p communication channel RTCConfiguration Dictionary RTCIceCredentialType Enum supports OAuth 2.0 based authentication. The application, acting as the OAuth Client, is … Continue reading WebRTC APIs

Hosted IP-PBX and its SBC

SBC ( Session Borde Controllers ) are basically gateways that provide interconnectivity between the hosted IP-PBX of the enterprise to the outside world endpoints such as telco service provider, PSTN/ TDM , SIP trunking providers or even third party OTT provider apps like skype for business etc.

sipP ( SIP testing tool )

SIPp is an opensource (GNU GPL license) performance testing tool for the SIP protocol and is widely used for Quality assurabce of callflows in voip applications for UAC / UASs cenarios. It can emulate functioing of a sip phone such as REGISTER , establishes and releases multiple calls with the INVITE and BYE methods , … Continue reading sipP ( SIP testing tool )

Gstreamer

GStreamer ( LGPL )ia a media handling library written in C for applicatioan such as streaming , recording, playback , mixing and editing attributes etc. Even enhnaced applicaiosn such as tsrancoding , media ormat conversion , streaming servers for embeeded devices ( read more about Gstreamer in RPi in my srticle here). It encompases various … Continue reading Gstreamer

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 .

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  

VoIP/ OTT / Telecom Solution startup’s strategy for Building a scalable flexible SIP platform

I have been contemplating points that make for a successful developer  to develop solutions and services for a  Telecom Application Server.  The trend has shown many variations from pure IN programs like VPN , Prepaid billing logic to SIP servlets for call parking , call completion. From SIP servlets to JAISNLEE open standard based communication. … Continue reading VoIP/ OTT / Telecom Solution startup’s strategy for Building a scalable flexible SIP platform

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Signals

sound waves

Sound is converted into electricity by a telephone and then transmitted as an analog signal. These waves have 3 fundamental characteristics: Amplitude, meaning the height (intensity) of the wave Frequency, which is the number of waves that pass in a single second and is measured in Hertz (cycles/second) (wavelength, the length of the wave from … Continue reading sound waves

Wave Modulation – digital

Information can be sent from A to B as an electromagnetic signal, in either an analog or digital form. The difference between the two is that : Analog is  continuous signal with intensity varying over time. Digital is discrete signal, switching between two different states over time. We shall cover digital wave modulation here . … Continue reading Wave Modulation – digital

Wave Modulation – analog

Modulating a wave means changing one or more of its fundamental characteristics to encode information. The general function for a sinusoid is or f(t) = A sin(omega t + phi) we can see that this has 3 parameters that can be altered . A – called the magnitude, or amplitude of the sinusoid. omega – … Continue reading Wave Modulation – analog

Access and Physical Layer Technologies

5G

In the course of evolution of RAN ( Radio Access layer) technologies 5G outsmarts 4G-2010 which comes in succession after 3G-2000 , 2.5G, 2G -1990 and 1G / PSTN -1980 respectively . Among the mosy striking features of 5G – entirely IP based ability to connect 100x more devices ( IOT favourable ) speed upto … Continue reading 5G

MIMO ( multiple-input and multiple-output )

Multiplying the capacity of a radio link using multiple transmission and receiving antennas to exploit multipath propagation.Key technology for achieving a vast increase of wireless communication capacity over a finite electromagnetic spectrum. Antenna configuration – implies antenna spatial diversity by useing arrays of multiple antennas on one or both ends of a wireless communication link … Continue reading MIMO ( multiple-input and multiple-output )

2nd and 3rd generation of telecommunication


6 thoughts on “

  1. All the topics shared are really amazing to read and completely clarify the title of this website, love to read, as no doubt here are huge things that related to the telecom and upcoming years new technology updates too are shared here.

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