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I have a setup to make sip-webrtc audio call and I have a scenario, where my browser uses only opus codec and my asterisk using only ilbc codec. So how can we make an audio to work in this scenrio? As I read from google we can do this by codec trans-coding(codec conversion) but I am not sure where and how I can implement this.

Can someone help me out on this please?

Regards,

Aravind

Linux Teki
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1 Answers1

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The WebRTC spec supports ilbc can you get your browser to use it?

If you can't then the fallback option is always to fallback to the G711 audio codec (ulaw or alaw) which has pretty much universal support including by WebRTC, Asterisk and every SIP phone I've ever come across.

sipsorcery
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  • the webrtc spec mandates opus and g711. ilbc is supported in webrtc.org but unlike ISAC is not enabled by default in Chrome even. – Philipp Hancke Oct 23 '19 at 04:24
  • Yeah I can use g711 however, it is consuming more bandwidth as compare to others so I am looking to have a codec transcoder from opus to ilbc but I am not sure how to implement this. can you help me out on this please. @PhilippHancke – Linux Teki Dec 16 '19 at 07:35