SIP

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

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

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.

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

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

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

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

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

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

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