rtc-v0.10.0

What's Changed

• AudioSource.capture_frame now buffer audio before asking the users to wait https://github.com/livekit/rust-sdks/pull/320
• use protoc's --pyi_out instead of mypy-protobuf package by @keepingitneil in https://github.com/livekit/python-sdks/pull/174
• fix livekit-protocol: generate_proto.sh by @nbsp in https://github.com/livekit/python-sdks/pull/181
• expose sample_rate & num_channels by @theomonnom in https://github.com/livekit/python-sdks/pull/183
• dix docs build logic by @keepingitneil in https://github.com/livekit/python-sdks/pull/186
• add SIP services. Support SIP DTMF in RTC. by @dennwc in https://github.com/livekit/python-sdks/pull/190 https://github.com/livekit/rust-sdks/pull/319
• export sip stuff by @theomonnom in https://github.com/livekit/python-sdks/pull/191
• debug logs must be explicitly enabled by @theomonnom in https://github.com/livekit/python-sdks/pull/192 https://github.com/livekit/rust-sdks/pull/322

New Contributors

• @nbsp made their first contribution in https://github.com/livekit/python-sdks/pull/181
• @dennwc made their first contribution in https://github.com/livekit/python-sdks/pull/190

Full Changelog: https://github.com/livekit/python-sdks/compare/rtc-v0.9.1...rtc-v0.10.0