livekit
Support Opus on SIP Trunks
When trying to enable OPUS with "OPUS/48000/2": true in the config file, SIP invites still don't offer the Opus codec over the trunk.
According to the contents of sip/pkg/sip /media_codecs.go the only codecs supported on the SIP side of things are G711 and G722.
I looked at the pkg/media/opus/opus.go file and compared it to pkg/media/g722/g722.go and I see that some functions are similar, but the opus.go file is missing the init() function and doesn't register itself.
Many SIP trunks (particularly when connecting to a PBX of some sort) support the Opus codec, so it would be very useful to have it available for use on the SIP side.
It's definitely on our roadmap, especially because LiveKit uses OPUS internally.
This would be really awesome with the whatsapp calling api - any update on timings?
No specific date yet, but we are considering rising a priority of it. WA is definitely one of the important use cases for it.
We're also very interested in using the new WhatsApp Business Call API, but currently unable to proceed due to the Opus codec limitation in the SIP bridge. This is becoming a critical integration point for us, as WhatsApp is a major channel for our voice agent infrastructure.
It would be really helpful if you could share whether support for this is on the roadmap, and if so, when it might be available. If there's anything we can do to help prioritize or test this feature, we're happy to contribute.
@dennwc Do you have any updates regarding the WA issue? We are eagerly awaiting your response.
No specific date yet, but we are considering rising a priority of it. WA is definitely one of the important use cases for it.
+1 as another very interested in Whatsapp Calling and unable to proceed due to lack of Opus support
https://kb.livekit.io/articles/5987869072-whatsapp-integration-with-livekit
source: https://livekit-users.slack.com/archives/C07FRP66XGE/p1755638133569199?thread_ts=1752642962.140529&cid=C07FRP66XGE
+1 Opus SIP - very important for our use case both with Whatsapp calling and creating high-quality connections to our internal PBX for human+ai hybrid conversations
While we are waiting for this feature request, has anyone tried the SIP Proxy workaround mentioned here?
I'm trying to figure it out using FreeSwitch, would appreciate any insights into this.
@dennwc will this issue be resolved with https://github.com/livekit/agents/issues/3375#issuecomment-3556164066? Or OPUS support in Livekit SIP will be addressed separately? If so, is this issue still on the roadmap?