Tag: SIP

  • SIP transport protocol transcoding in OpenSIPS

    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 […]

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  • Fixing SIP header addresses – Contact headers

    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 […]

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  • Fixing SIP header addresses – Via headers

    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 […]

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  • Contact and Record-Route headers explained

    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 […]

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  • Top reasons why VoIP calls drop

    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 1

    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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  • How to install Mediaproxy 2.5.2 on CentOS 6 64 bit

    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

    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?

    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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  • Using Custom Device States to control BLF lamps

    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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