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Version Packages - #1896
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This PR was opened by the Changesets release GitHub action. When you're ready to do a release, you can merge this and the packages will be published to npm automatically. If you're not ready to do a release yet, that's fine, whenever you add more changesets to main, this PR will be updated.
Releases
@livekit/protocol@1.53.2
Patch Changes
Replace
NodeStats.forward_scheduling_delaywithNodeStats.forward_processing_per_write, the mean forwarding work per write to a subscriber - #1895 (@boks1971)webhook:
ResourceURLNotifier.Stop(false)waits for queued events to be sent, asURLNotifier.Stop(false)does - #1901 (@mnjalale)sdp: add CodecsFromSessionDescription and ParseFmtp helpers - #1899 (@anunaym14)
agent_simulation: dial a phone number over SIP instead of dispatching an agent - #1858 (@u9g)
Add
SIPDialtoneto select a regional dial tone on CreateSIPParticipant and TransferSIPParticipant; deprecateplay_dialtone- #1892 (@dennwc)sip: add
SIPMediaConfig.drop_audio_during_dtmfto stop sending audio RTP while DTMF is sent - #1907 (@alexlivekit)SIP authentication: compare IP masks against remote address rather than From header. - #1889 (@alexfish8)
github.com/livekit/protocol@1.53.2
Patch Changes
Replace
NodeStats.forward_scheduling_delaywithNodeStats.forward_processing_per_write, the mean forwarding work per write to a subscriber - #1895 (@boks1971)hwstats: ignore LIVEKIT_CPU_REQUEST when measuring whole-host CPU - #1904 (@boks1971)
hwstats: add WithHostCPU option to measure whole-host CPU from /proc/stat - #1903 (@boks1971)
webhook:
ResourceURLNotifier.Stop(false)waits for queued events to be sent, asURLNotifier.Stop(false)does - #1901 (@mnjalale)sdp: add CodecsFromSessionDescription and ParseFmtp helpers - #1899 (@anunaym14)
agent_simulation: dial a phone number over SIP instead of dispatching an agent - #1858 (@u9g)
Add
SIPDialtoneto select a regional dial tone on CreateSIPParticipant and TransferSIPParticipant; deprecateplay_dialtone- #1892 (@dennwc)sip: add
SIPMediaConfig.drop_audio_during_dtmfto stop sending audio RTP while DTMF is sent - #1907 (@alexlivekit)SIP authentication: compare IP masks against remote address rather than From header. - #1889 (@alexfish8)