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Using TLS in OpenSIPS v2.2.x

by Smartvox on March 16, 2017

I’ve been looking at TLS connections to OpenSIPS recently. Support for TLS existed in version 1, but the configuration changed significantly in version 2. Using version 2.2.3, you will need to include a listen statement a bit like this: listen=tls: Load the following modules, in addition to the usual ones: loadmodule “” loadmodule “” …and […]

Top reasons why VoIP calls drop

by Smartvox on November 17, 2013

VoIP based phone systems bring many benefits, but they also bring some problems. Not least is the annoying tendency for some calls to drop mid-way through your conversation for no obvious reason. In this article I will identify the most common reasons why a VoIP call might suddenly drop mid-way through an established call and explain […]

Pika µFirewall SIP firewall review

October 7, 2013

A new way to secure your IP-PBX Recently introduced by the well-established Canadian telecoms manufacturer Pika Technologies, the Pika µFirewall offers a novel way to make your Asterisk (or any other SIP-based PBX) more secure. The best way to describe it is as a “SIP Firewall”, but unlike conventional network firewalls, installation is very simple […]

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SIP User Credentials in the ITSP environment – part 3

December 15, 2012

In this, the third part of the article, we will look at how the ITSP Proxy server handles outbound calls, paying particular attention to the authentication requirements and to the handling of Caller ID. You may also wish to refer back to part 1 where we reviewed the basic entities and concepts used to identify […]

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SIP User Credentials in the ITSP environment – part 2

September 13, 2012

In part 1 of this article we reviewed the basic entities and concepts used to identify a user, a handset or service. It discussed in detail how these entities are used to register devices, either simple single-line devices or SIP trunks. Now, in part 2, we will examine in more detail the handling of inbound […]

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SIP User Credentials in the ITSP environment – part 1

August 10, 2012

This three-part article reviews the SIP entities that identify a user, a handset or service and looks at how they are used to register devices, authenticate and identify users and route calls. The subtle differences in the use of these entities can be confusing, even to an experienced SIP technician. Handling of the entities in […]

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RTP, Jitter and audio quality in VoIP

April 24, 2012

In this article we will briefly look at what RTP is and how it is used to stream VoIP audio. The article then considers how certain network transmission characteristics may introduce jitter or packet loss and the measures that are used in VoIP equipment to mitigate the effects. Other phenomenon which have a bearing on […]

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How to install Mediaproxy 2.5.2 on CentOS 6 64 bit

March 16, 2012

Mediaproxy 2.5.2 is a Python application from AG-Projects which is available as a free download as well as being available as a commercial product from AG-Projects. It is used in combination with the Mediaproxy module of OpenSIPS. Mediaproxy 2 has several dependencies and can be quite tricky to install. The INSTALL instructions that come with […]

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OpenSIPS vs Asterisk

January 17, 2012

OpenSIPS and Asterisk are both open source projects and both are used for Voice over IP. However, they perform quite different roles, have different capabilities and different strengths and weaknesses. This article reviews how they are so different and considers what role each product can play in the infrastructure of an Internet Telephony Service Provider […]

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What is OpenSIPS?

December 31, 2011

There are a number of open source applications available that are used to build IP Telephony solutions. OpenSIPS may not be as well-known as Asterisk, but it is widely used by service providers as a core part of their infrastructure because of its robustness, speed and capacity. In this article I will review the history […]

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