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RTP, Jitter and audio quality in VoIP

by Smartvox on 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

by Smartvox on 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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VoIP QoS in practice: About Network Congestion

October 26, 2011

My previous two articles explored QoS tagging of voice data packets using ToS/DiffServ values and of Ethernet frames using CoS or Priority values. QoS is often advocated as an essential part of any self-respecting VoIP solution and there is no doubt it can make a big difference in the right circumstances. However, it would be [...]

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VoIP QoS Settings – part 2

July 29, 2011

In part 1, we examined the Layer 2 QoS settings available on most VoIP equipment. In this second part, I will explore the Layer 3 parameters and offer practical suggestions for the values that should be assigned to them. We will briefly look at the history and structure of the ToS and DSCP fields and [...]

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VoIP QoS Settings – part 1

June 17, 2011

The QoS settings on VoIP phones and related equipment can be perplexing. Here, I will attempt to explain what parameters like CoS, ToS, DiffServ and DSCP really mean and offer practical suggestions for the values that should be assigned to them. Part 1 of this article starts with a broad overview and then focuses on [...]

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Setting up shared voicemail on Asterisk – part 2

March 31, 2011

Part 1 laid the foundations for creating and accessing a shared voicemail box. In this, part 2, I explain how the lamp on the BLF key is switched on and off to show there are messages waiting in the shared box. Note that this is separate from any existing MWI lamp used for personal voicemail. [...]

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Setting up shared voicemail on Asterisk – part 1

March 23, 2011

It’s a requirement that people often seem to ask for - a single voicemail box, taking messages for a department, that can be easily monitored and accessed by several different users. A typical application would be to record out-of-hours messages which are then checked in the morning by any of a number of users, perhaps just depending who [...]

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Using Custom Device States to control BLF lamps

February 17, 2011

Do you want to know how to use a custom device state to control the lamp on a programmable key of an IP phone? In this article I explain how to set up the hints and make any number of IP phones subscribe to a custom device state and how to switch the custom status from within the Asterisk dial plan. [...]

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