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Home / v1.8.3
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checksums.txt 2025-01-07 591 Bytes
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livekit_1.8.3_linux_amd64.tar.gz 2025-01-07 12.0 MB
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livekit_1.8.3_linux_armv7.tar.gz 2025-01-07 11.2 MB
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livekit_1.8.3_windows_amd64.zip 2025-01-07 12.1 MB
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livekit_1.8.3_windows_arm64.zip 2025-01-07 11.0 MB
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README.md 2025-01-07 3.0 kB
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v1.8.3 source code.tar.gz 2025-01-07 775.0 kB
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v1.8.3 source code.zip 2025-01-07 1.0 MB
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Totals: 10 Items   70.5 MB 0

[1.8.3] - 2025-01-07

Added

  • Allow requesting a dialtone during call transfer (#3122)
  • Handle room configuration that's set in the grant itself (#3120)
  • Update ICE to pick up accepting use-candidate unconditionally for ICE lite agents (#3150)
  • auto create rooms during create agent dispatch api request (#3158)
  • Annotate SIP errors with Twirp codes. (#3161)
  • TWCC based congestion control (#3165 [#3234] [#3235] [#3244] [#3245] [#3250] [#3251] [#3253] [#3254] [#3256] [#3262] [#3282])
  • Loss based congestion signal detector. (#3168 [#3169])
  • Fix header size calculation in stats. (#3171)
  • add per message deflate to signal ws (#3174)
  • Add ResyncDownTracks API that can be used to resync all down tracks on (#3185)
  • One shot signalling mode (#3188 [#3192] [#3194] [#3223])
  • Server side metrics (#3198)
  • Add datastream packet type handling (#3210)
  • Support SIP list filters. (#3240)
  • Add RTX to downstream (#3247)
  • Handle REMB on RTX RTCP (#3257)
  • Thottle the publisher data channel sending when subscriber is slow (#3255 [#3265] [#3281])

Fixed

  • avoids NaN (#3119)
  • reduce retransmit by seeding duplicate packets and bytes. (#3124)
  • don't return video/rtx to client (#3142)
  • ignore unexported fields in yaml lint (#3145)
  • Fix incorrect computation of SecondsSinceNodeStatsUpdate (#3172)
  • Attempt to fix missing participant left webhook. (#3173)
  • Set down track connected flag in one-shot-signalling mode. (#3191)
  • Don't SetCodecPreferences for video transceiver (#3249)
  • Disable av1 for safari (#3284)
  • fix completed job status updates causing workers to reconnect (#3294)

Changed

  • Display both pairs on selected candidate pair change (#3133)
  • Maintain RTT marker for calculations. (#3139)
  • Consolidate operations on LocalNode. (#3140)
  • Use int64 nanoseconds and reduce conversion in a few places (#3159)
  • De-centralize some configs to where they are used. (#3162)
  • Split out audio level config. (#3163)
  • Use int64 nanoseconds and reduce conversion in a few places (#3159)
  • Reduce lock scope. (#3167)
  • Clean up forwardRTP function a bit. (#3177)
  • StreamAllocator (congestion controller) refactor (#3180)
  • convert psprc error to http code in rtc service failure response (#3187)
  • skip http request logging when the client aborts the request (#3195)
  • Do not treat data publisher as publisher. (#3204)
  • Publish data and signal bytes once every 30 seconds. (#3212)
  • upgrade to pion/webrtc v4 (#3213)
  • Don't wait rtp packet to fire track (#3246)
  • Keep negotiated codec parameters in Downtrack.Bind (#3271)
  • Structured logging of ParticipantInit (#3279)
  • Start stream allocator after creating peer connection. (#3283)
  • Reduce memory allocation in WritePaddingRTP / WriteProbePackets (#3288)
  • add room/participant to logger context for SIP APIs (#3290)
  • vp8 temporal layer selection with dependency descriptor (#3302)
  • Use contiguous groups to determine queuing region. (#3308)
Source: README.md, updated 2025-01-07