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Category: SIP

Installing WebRTC2SIP gateway – tutorial

System preparation

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) and libspeexdsp (audio processing and jitter buffer) add

Kamailio Call establishment permission rules

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, a company can allow the people to call outside, for example to PSTN. The rules can change over time as well as the numbering scheme itself.

Installing SIPp 3.2 on Debian Squeeze 6.0.5 32 bit

SIPp version 3.2 hasn't precompiled binary packages. There are some instructions provided at the sipp home page, however small comment about limits.h is missing and therefore the compilation will not be succesfull. Here we provide instructions how to compile Sipp with SSL, pcap play and distributed pauses features enabled.

Install prerequisities

Three pre-requisites are necessary to compile SIPp: