TCPdump is a powerful command-line packet analyzer, which may be used for a SIP message sniffing/analyzing, and thus for the troubleshooting of a SIP system. TCPdump is preinstalled on many Linux distributions, or may be installed directly from the Debian…

Sometimes there is a need for simple and quick analysis or the troubleshooting of a SIP server and its call functions. Of course, we should use the well-known tcpdump, mentioned in the article Using tcpdump for SIP diagnostics. However, for…

Table provides the overview of security features of nine analysed open-source SIP clients (some sources call them the RTC communicator). Source: P. Segeč, M. Moravčík, J. Hrabovský, J. Papán and J. Uramová, „Securing SIP infrastructures with PKI — The analysis,“…

We had observed a problem, where a SIP phone is registering, but the AOR record indicates, that as a Contact IP address the incorrect and strange private IP address is used. As is shown on following listing: voip*CLI> pjsip show…

First, add repository to system vim /etc/apt/sources.list add line deb http://files.freeswitch.org/repo/deb-master/debian/ wheezy main Then add GPG key to repository gpg –keyserver pool.sks-keyservers.net –recv-key D76EDC7725E010CF gpg -a –export D76EDC7725E010CF | apt-key add – Now, update repository apt-get update FreeSWITCH is ready…

Go to your desired folder (for example /var/www) and download Siremis from site http://siremis.asipto.com/pub/downloads/siremis/. cd /var/www/ wget http://siremis.asipto.com/pub/downloads/siremis/siremis-4.1.0.tgz Then extract Siremis folder from archive tar -xvf siremis-4.1.0.tgz Folder siremis-4.1.0 appears here.  

apt-get update apt-get upgrade As the first step we need to install packages necessary to build the main webrtc2sip gateway: apt-get install build essential libtool automake subversion git pkg-config screen libxml2-dev / libssl-dev libsrtp0-dev to support for libspeex (audio codec)…

Author: Patrik Formanek 2014 This tutorial instruct how to add the WebSocket support for your kamailio SIP server. As the prerequisities we need to have successfully installed and working kamailio server (described within several tutorials in this site, for example…

In many setups Kamailio is used as a PROXY server that takes care of routing calls to servers providing voice services, e.g. voicemail, IVR or conference calls. There are a few things an administrator must keep in mind.

This article talks about deploying permission control mechanism for call establishment in Kamailio SIP Proxy. In many VoIP solutions, it is crutial to deploy numbering scheme and write down rules where users are/aren't allowed to call. On top of that,…