rtc-v0.10.0

What's Changed

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

New Contributors

• @nbsp made their first contribution in #181
• @dennwc made their first contribution in #190

Full Changelog: rtc-v0.9.1...rtc-v0.10.0