* feat(rpc/room): add `ListParticipants` and `GetParticipant`
Add `ListParticipants` and `GetParticipant` RPCs to the `Room` PSRPC service,
following the same topic-based routing pattern as:
- `DeleteRoom`,
- `SendData`,
- and `UpdateRoomMetadata`.
These are needed so the livekit server can route participant read operations to the authoritative
RTC node instead of reading from eventually-consistent Redis.
No breaking changes for existing `UnimplementedRoomServer` embedders
(new methods have default stubs).
* Add ListParticipants and GetParticipant features
* normalize protobuf "id" log field names
* Update changeset for '@livekit/protocol' patch
Updated the changeset to include a patch for '@livekit/protocol' and normalize protobuf 'id' log field names.
* generated protobuf
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* Adding retry cnt to support egress auto retry feature
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- Add projectID explicitly to GetIngressInfoResponse as we cannot send it in headers on a response
- move ingressobs.Reporter to protocol
- Remove the unused KillIngressSession RPC
- Random fmt fixes by the protobuf linter included in my editor...
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Makes sure RPC systems that use the livekit logger wrappers do not log these fields.
This also makes indenting and spaces more consistent by using passing the file through clang-format. Happy to revert this if we want to avoid large cosmetic PRs
This also changes the go_package location for logger/options.proto to github.com/livekit/protocol/livekit/logger" since github.com/livekit/protocol/logger already exists and has some non generated go code inside.
Also add a unit test for the egress redaction. The unit test lives in github.com/livekit/protocol/livekit because protocol/livekit/sip.go imports github.com/livekit/protocol/logger, so putting the test in github.com/livekit/protocol/logger would create an import loop.
* Add room service token to connector room join info
To update participant visible when twilio participant connected
* generated protobuf
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* Adding TrunkInfo to SipCallInfo as well as GetSIPTrunkAuthenticationResponse
* generated protobuf
* Directly using provider info instead of deprecated SipTrunkInfo
* generated protobuf
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* Adding an RPC to record call context using the callinfo
* generated protobuf
* Adding changeset
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* HTTP DELETE to delete a participant session - relay signal message for
that.
Also, doing a basic signal cache and building server message and client
message cache on top of it.
* generated protobuf
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* More v2 stuff
- Adding `Trickle` to client -> server message. There should not be any
tricklw from server -> client in v2. So, not adding it. Will add if
needed later.
- Some utils to parse envelopes as we are moving in the direction of
having an envelope of messages in the initial connect request to keep
it modular. May change again later, but today's state is this.
- change relay signal to pass wire message back and forth.
* generated protobuf
* add utils
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* Adding a new filed in the SipCall message to store the SIP protocol generated call id.
This can be used in the sip code base whenever the sip call id needs to be fetched at a later stage
* generated protobuf
* adding changeset
* Ensuring that the generated protobufs from replay folder are also added to the CI commit
* generated protobuf
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* WHIP internal signalling to be able to support WHIP in OSS.
* generated protobuf
* Create wicked-numbers-warn.md
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* feat: MoveParticipant API
This new API would move a participant from one cloud room to another.
It will support the following use cases:
- warm transfer in voice AI
- fast switching between different live streams
- prejoin room in meetings
* generated protobuf
* add RTC message
* repeat field number
* generated protobuf
* rpc
* generated protobuf
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Co-authored-by: cnderrauber <zengjie9004@gmail.com>
