Codecs supported by livekit when SIP
# ask-aim
miniature-nail-35141
07/29/2025, 6:34 AMCodecs supported by livekit when SIP
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tall-airline-16756
07/29/2025, 6:34 AMThanks for reaching out, Rajadurai. I'm an AI assistant and I'm here to help. Give me a moment while I research this topic.
tall-airline-16756
07/29/2025, 6:35 AMLiveKit supports the following audio codecs when using SIP:
• G.711 (PCMU/PCMA) - Traditional telephony codecs
• G.722 - Wideband audio codec for HD Voice
• Opus - WebRTC codec used for transcoding
• DTMF - For dual-tone multi-frequency signaling
LiveKit also supports wideband audio codecs like Opus and G.722 for HD Voice, enabling higher quality audio when used with compatible SIP trunks or endpoints. Additionally, the SIP service handles codec transcoding between telephony codecs (e.g., PCMU/PCMA) and WebRTC codecs (e.g., Opus), ensuring seamless integration between traditional phone systems and WebRTC-based LiveKit rooms.
Sources: res/README.md | SIP Integration | livekit/livekit_composite | DeepWiki | Configuring HD Voice for SIP | LiveKit Docs
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miniature-nail-35141
07/29/2025, 6:41 AMDoes DTMF RFC2833 support?
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tall-airline-16756
07/29/2025, 6:42 AMHey there! Let me break down the DTMF support in LiveKit for you. LiveKit's SIP implementation has got you covered with DTMF functionality. It supports RFC 4733 for digital DTMF signals, and the code shows it's also compatible with RFC 2833. So whether you're using digital signals or audio DTMF tones, you'll have a smooth experience. Pretty cool, right? We've made sure to cover all the bases to give you the most flexible communication options. Sources: dtmf/dtmf.go | SIP Integration | livekit/livekit_composite | DeepWiki | Handling DTMF | LiveKit Docs