Hi everyone,
I noticed that LiveKit SIP trunks currently support only the G.722 and G.711 codecs. Is there any way or workaround to initiate workers using the Opus codec instead?
Also, are there any plans to add Opus codec support for LiveKit SIP trunks in the future?
Thanks in advance!
@Robbe, Boss, no way to flip it on today. The codec-list config landed in PR #672 (merged 2026-05-06), which added OnlyListedCodecs in SIPMediaConfig, but the PR description explicitly says “For now there are no new supported codecs. The first planned additions are OPUS and AMR-WB which will follow shortly in separate PRs” [ livekit/sip # 672 ]. So the plumbing is in place but Opus and AMR-WB aren’t wired up yet. The PR queue at [ Pull requests · livekit/sip · GitHub ] is where the enablement PR will appear.
For PSTN destinations the codec choice matters less than it looks anyway. Twilio, Telnyx, and Plivo terminate to G.711 µ-law at their edge regardless of what you negotiate on the LK side, so Opus over SIP only meaningfully helps for IP-to-IP SIP (internal trunks, SIP-over-WebRTC peers). For PSTN, G.711/G.722 is the ceiling.
What about a FreeSwitch SIP proxy workaround? Also: @Muhammad_Usman_Bashir please check your WhatsApp, I think you overlooked my message.
This is the PR linked to from our internal backlog:
Yes, Boss.
My bad, getting back to you today for sure.
Yes, you can use freeswitch to transcode. But also LiveKit WhatsApp conenector is now open beta.