Smartvox Labs

Tag: WebRTC

  • SIP transport protocol transcoding in OpenSIPS

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    John Quick

    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 – Introduction

    Fixing SIP header addresses – Introduction

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    John Quick

    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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    John Quick

    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 […]

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