Tag: transcoding
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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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RTP pass-through mode in Asterisk
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What is RTP pass-through mode in Asterisk? The speech for VoIP calls uses RTP (Real Time Protocol) to get from one end to the other and it is compressed using one of the many speech compression codecs available. Commonly used codecs include G.711 and G.729. When you call someone through an Asterisk PBX, there will be two “legs” to that call. The first leg is from your phone to Asterisk and the second is from Asterisk to…
