* Use mapped SDP answer when using single peer connection mode.
In single peer connection mode, the client does not get the
TR_ ids for tracks. That is because the transceiver is created with a
receiver when the offer is generated and a random id assigned. Once the
id is assigned, it is not updated. So, when client gets `onTrack`, the
id of the associated MediaStreamTrack is not the TR_ variety.
So, send the mapping of mid -> trackID in signalling when sending
answer. And used that on client side to find LK track ID for given mid.
Side note: Probably worth doing it for dual peer connection also when
subscriber sends offer so that we do not have to deduce it on client
side using some rules.
* generated protobuf
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Co-authored-by: github-actions <41898282+github-actions[bot]@users.noreply.github.com>
* Adding TrunkInfo to SipCallInfo as well as GetSIPTrunkAuthenticationResponse
* generated protobuf
* Directly using provider info instead of deprecated SipTrunkInfo
* generated protobuf
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Co-authored-by: github-actions <41898282+github-actions[bot]@users.noreply.github.com>
* add presigned post request for cloud-agents api
* generated protobuf
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Co-authored-by: github-actions <41898282+github-actions[bot]@users.noreply.github.com>
* Marking width and height as not deprecated.
Those fields reflect the capture dimensions.
* generated protobuf
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Co-authored-by: github-actions <41898282+github-actions[bot]@users.noreply.github.com>
* add server region to agent info and create for log handling
* generated protobuf
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Co-authored-by: github-actions <41898282+github-actions[bot]@users.noreply.github.com>
Extending models to support MP3 audio encoding in egress
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Co-authored-by: github-actions <41898282+github-actions[bot]@users.noreply.github.com>
* Suport for simulcast codec of audio.
Note that a disable should be a mute on client side and not a track
stop to allow re-enabling the track again.
* generated protobuf
* name change
* generated protobuf
* Create young-mayflies-wash.md
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Co-authored-by: github-actions <41898282+github-actions[bot]@users.noreply.github.com>
* Adding utilities for phonenumber functionality
1. DB prefixes
2. Helper utilities for area code and number type
* Adding changeset
* Adding enum for phone number type
* generated protobuf
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Co-authored-by: github-actions <41898282+github-actions[bot]@users.noreply.github.com>
* Adding an RPC to record call context using the callinfo
* generated protobuf
* Adding changeset
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Co-authored-by: github-actions <41898282+github-actions[bot]@users.noreply.github.com>
* Adding a call context field to CallInfo for dynamically setting info about the call
* generated protobuf
* Using a repeated type instead so we don't need to maintain key strings
* generated protobuf
* Adding changeset
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Co-authored-by: github-actions <41898282+github-actions[bot]@users.noreply.github.com>
* Adding get and update apis for livekit phone numbers
* generated protobuf
* Forcing a recompile
* Checking in changeset
* Using rule id instead of dispatchruleinfo struct for PurchasedPhoneNumber
* generated protobuf
* Removing unused import
* generated protobuf
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Co-authored-by: github-actions <41898282+github-actions[bot]@users.noreply.github.com>
* Public facing api for livekit phone numbers
* more changes
* Adding internal apis
* Adding some more changes to the proto
* generated protobuf
* generated protobuf
* Updating to remove internal apis
* Adding changeset
* Clean up and changes around common fields for Phone number messages and pricing for the same
* generated protobuf
* Using a more generic costing model based on resources
* generated protobuf
* Adding a reference to SipDispatchRule in the PurchasedPhoneNumber message
* generated protobuf
* Using google protobuf timestamp and adding a struct for TokenPagination
* generated protobuf
* Using an array of numbers for the release phone number api as well
* generated protobuf
* Making spam score a double
* generated protobuf
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Co-authored-by: github-actions <41898282+github-actions[bot]@users.noreply.github.com>
* Add webhook for aborted participant connection.
Will be sent when the participant session is closed without participant
becoming active.
* generated protobuf
* analytics event
* generated protobuf
* Create famous-rivers-repeat.md
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Co-authored-by: github-actions <41898282+github-actions[bot]@users.noreply.github.com>
Co-authored-by: David Zhao <dz@livekit.io>
* Store SDP cid in TrackInfo.
Only store it if different to avoid redundant info.
For example, FF uses a different stream id between AddTrack and SDP offer.
* generated protobuf
* Create cuddly-doors-relate.md
* spelling
* generated protobuf
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Co-authored-by: github-actions <41898282+github-actions[bot]@users.noreply.github.com>
* Allow clients to specify video layers mode.
The main issue is distinguishing between one RTP streaming carry
multiple spatial layers (example SVC codecs) vs one spatial layer
(codecs that are typically simulcast). The enum names are long, but
wanted them to be clear.
* generated protobuf
* Create unlucky-birds-drive.md
* comment
* generated protobuf
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Co-authored-by: github-actions <41898282+github-actions[bot]@users.noreply.github.com>
* 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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Co-authored-by: github-actions <41898282+github-actions[bot]@users.noreply.github.com>
* 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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Co-authored-by: github-actions <41898282+github-actions[bot]@users.noreply.github.com>
