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