Tag: SIP
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SIP transport protocol transcoding in OpenSIPS
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Introduction This is the final article in my series about fixing SIP header addresses. It broadly overlaps with the theme of the earlier articles. As a VoIP solutions designer, you may want your proxy server to deliberately transcode between different transport protocols. For example, WebRTC to TCP or TLS to UDP. This is possible with OpenSIPS, but getting every detail correct across a wide range of use-cases is quite difficult. It requires attention to detail in…
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Fixing SIP header addresses – Contact headers
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Part 3 of this series of articles focusses on the Contact header. In particular, I examine the use-cases where it is necessary to “fix” (or alter) a received Contact header. Contact headers work in close combination with Record-Route and Route headers in a mechanism known as loose routing. To get the most from this article some prior knowledge is required about loose routing, so if you are not already well acquainted with this subject please head…
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Fixing SIP header addresses – Via headers
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In part 2, Via headers are put under the microscope. I examine how the address in the Via header is set by each node in the path; how and why it may differ from the source address. I will look at the functions available in OpenSIPS to detect and handle situations where the address in the received Via differs from the source address of the request. I touch on the options available within OpenSIPS to select…
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Contact and Record-Route headers explained
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Diagnosing some problems in the world of VoIP requires close inspection of the SIP messages being exchanged, but there are many occasions where a good understanding of loose routing will be invaluable. The headers that underpin loose routing are Contact, Record-Route and Route. In this post, I explain how they work and provide some insight into the way they interact. Some Acronyms and terminology UAC UAS SIP Proxy URI * Sequential Request User Agent Client (for…
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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 how you can diagnose the cause. At the end are some pointers to the solutions for these problems. This article is…
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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 a live ITSP environment, especially when dealing with a mix of simple single-line devices alongside multi-channel SIP trunks, is quite challenging.…
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How to install Mediaproxy 2.5.2 on CentOS 6 64 bit
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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 the package are very helpful, but unfortunately they are aimed primarily at installers who either have Debian or Ubuntu Linux distributions.…
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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 solution. Fundamental differences A succinct, but slightly technical, distinction between OpenSIPS and Asterisk is that OpenSIPS is essentially a SIP Proxy Server…
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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 of OpenSIPS, explain what it is, what features it offers and its core operational roles. This will include a summary of…
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Using Custom Device States to control BLF lamps
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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. In later articles I plan to show how this can be put to a practical use. I will explain how to…
