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Does Livekit support SIP INVITES and RTP audio processing?

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Yes, LiveKit supports SIP INVITES for call handling and RTP audio processing for media streams. This includes features like bidirectional audio bridging and decoding/encoding RTP packets for audio streams.
 Sources: <https://github.com/livekit/sip/blob/71966230cf270d5615ecb9dfe939308efb8a9dff/pkg/sip/media_port.go#L490-L562|pkg/sip/media_port.go> | <https://deepwiki.com/livekit/livekit_composite/4-livekit-server-components|LiveKit Server Components | livekit/livekit_composite | DeepWiki> | <https://github.com/livekit/livekit/blob/5e483e7554e5afbf254acf84e3ec0aa6e108e758/CHANGELOG.md#L1-L45|CHANGELOG.md> | <https://github.com/livekit/protocol/blob/f44a41e4fd30e487f3e4dd94044104db9f48a7d3/CHANGELOG.md#L1-L48|CHANGELOG.md>