Category: VoIP
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Using the Clusterer Module for contact replication
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Summary In this, the second part of a three-part article about the Clusterer Module, I explain how I got on when testing a pair of OpenSIPS Registrar Proxies configured as a highly available cluster. The design, which uses Pacemaker to assign a floating IP to the currently active server, is described in some detail in […]
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HA Scenarios and the OpenSIPS Clusterer Module
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Summary System reliability and avoidance of down-time are themes that recur time and again with Smartvox customers. In this article I will review some of options available to a system designer who wants to build a highly available OpenSIPS solution, paying particular attention to the new Clusterer module which is at the heart of a set […]
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Using TLS in OpenSIPS v2.2.x
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Using TLS with OpenSIPS: Why do we need it and how is it configured? While support for TLS existed in version 1, the configuration changed significantly in version 2. This article briefly covers the new v2 setup. The role of TLS in VoIP calls Today, it is common practice to use encrypted communication over the […]
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Top reasons why VoIP calls drop
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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 […]
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SIP User Credentials in the ITSP environment – part 3
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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
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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
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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
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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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OpenSIPS vs Asterisk
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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?
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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 […]
