Tag: WebRTC
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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 – Introduction
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The main theme I explore in these articles is when and how a SIP Proxy should alter (or “fix”) embedded sender address information – IP and port – in a SIP request that it has received. The headers that are most relevant here are Via, Contact and Record-Route
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WebRTC using OpenSIPS and RTPEngine
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In this article you will find tips, pointers and code snippets to help you get started with WebRTC using OpenSIPS and RTPEngine. At the end I have provided some notes and URL links that may be useful to anyone wishing to learn more about the media handling. In the initialisation section of opensips.cfg A listen statement is required to make opensips accept websocket connections. The usual port is 443, but you can use a different port…
