mirror of
https://github.com/livekit/livekit.git
synced 2026-05-10 17:16:09 +00:00
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This project adheres to [Semantic Versioning](https://semver.org/spec/v2.0.0.html).
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## [1.8.1] - 2025-01-03
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### Added
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- Allow requesting a dialtone during call transfer (#3122)
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- Handle room configuration that's set in the grant itself (#3120)
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- Update ICE to pick up accepting use-candidate unconditionally for ICE lite agents (#3150)
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- auto create rooms during create agent dispatch api request (#3158)
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- Annotate SIP errors with Twirp codes. (#3161)
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- TWCC based congestion control (#3165 #3234 #3235 #3244 #3245 #3250 #3251 #3253 #3254 #3256 #3262 #3282)
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- Loss based congestion signal detector. (#3168 #3169)
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- Fix header size calculation in stats. (#3171)
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- add per message deflate to signal ws (#3174)
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- Add ResyncDownTracks API that can be used to resync all down tracks on (#3185)
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- One shot signalling mode (#3188 #3192 #3194 #3223)
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- Server side metrics (#3198)
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- Add datastream packet type handling (#3210)
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- Support SIP list filters. (#3240)
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- Add RTX to downstream (#3247)
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- Handle REMB on RTX RTCP (#3257)
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- Thottle the publisher data channel sending when subscriber is slow (#3255 #3265 #3281)
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### Fixed
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- avoids NaN (#3119)
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- reduce retransmit by seeding duplicate packets and bytes. (#3124)
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- don't return video/rtx to client (#3142)
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- ignore unexported fields in yaml lint (#3145)
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- Fix incorrect computation of SecondsSinceNodeStatsUpdate (#3172)
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- Attempt to fix missing participant left webhook. (#3173)
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- Set down track connected flag in one-shot-signalling mode. (#3191)
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- Don't SetCodecPreferences for video transceiver (#3249)
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- Disable av1 for safari (#3284)
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- fix completed job status updates causing workers to reconnect (#3294)
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### Changed
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- Display both pairs on selected candidate pair change (#3133)
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- Maintain RTT marker for calculations. (#3139)
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- Consolidate operations on LocalNode. (#3140)
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- Use int64 nanoseconds and reduce conversion in a few places (#3159)
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- De-centralize some configs to where they are used. (#3162)
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- Split out audio level config. (#3163)
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- Use int64 nanoseconds and reduce conversion in a few places (#3159)
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- Reduce lock scope. (#3167)
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- Clean up forwardRTP function a bit. (#3177)
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- StreamAllocator (congestion controller) refactor (#3180)
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- convert psprc error to http code in rtc service failure response (#3187)
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- skip http request logging when the client aborts the request (#3195)
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- Do not treat data publisher as publisher. (#3204)
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- Publish data and signal bytes once every 30 seconds. (#3212)
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- upgrade to pion/webrtc v4 (#3213)
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- Don't wait rtp packet to fire track (#3246)
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- Keep negotiated codec parameters in Downtrack.Bind (#3271)
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- Structured logging of ParticipantInit (#3279)
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- Start stream allocator after creating peer connection. (#3283)
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- Reduce memory allocation in WritePaddingRTP / WriteProbePackets (#3288)
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- add room/participant to logger context for SIP APIs (#3290)
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## [1.8.0] - 2024-10-18
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### Added
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+1
-1
@@ -14,4 +14,4 @@
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package version
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const Version = "1.8.0"
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const Version = "1.8.1"
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