diff --git a/.changeset/decodebudget-decoded-bytes.md b/.changeset/decodebudget-decoded-bytes.md deleted file mode 100644 index 2d6a7e192..000000000 --- a/.changeset/decodebudget-decoded-bytes.md +++ /dev/null @@ -1,5 +0,0 @@ ---- -"github.com/livekit/protocol": patch ---- - -decodebudget: move the protobuf/JSON decoding budgets to `utils/decodebudget` and add a calibrated decoded-allocation estimate (`MaxDecodedBytes`, `Estimate`, `ValidateProtoJSON`) diff --git a/.changeset/decodebudget-merged-messages.md b/.changeset/decodebudget-merged-messages.md deleted file mode 100644 index 5524c48e7..000000000 --- a/.changeset/decodebudget-merged-messages.md +++ /dev/null @@ -1,5 +0,0 @@ ---- -"github.com/livekit/protocol": patch ---- - -decodebudget: charge repeated occurrences of a singular message field against the merged message, closing a quadratic undercount diff --git a/.changeset/forward-processing-per-subscriber.md b/.changeset/forward-processing-per-subscriber.md deleted file mode 100644 index bc9fc2cc1..000000000 --- a/.changeset/forward-processing-per-subscriber.md +++ /dev/null @@ -1,6 +0,0 @@ ---- -"github.com/livekit/protocol": patch -"@livekit/protocol": patch ---- - -Replace `NodeStats.forward_scheduling_delay` with `NodeStats.forward_processing_per_write`, the mean forwarding work per write to a subscriber diff --git a/.changeset/hwstats-host-cpu-ignore-request.md b/.changeset/hwstats-host-cpu-ignore-request.md deleted file mode 100644 index b78deab6f..000000000 --- a/.changeset/hwstats-host-cpu-ignore-request.md +++ /dev/null @@ -1,5 +0,0 @@ ---- -"github.com/livekit/protocol": patch ---- - -hwstats: ignore LIVEKIT_CPU_REQUEST when measuring whole-host CPU diff --git a/.changeset/hwstats-host-cpu.md b/.changeset/hwstats-host-cpu.md deleted file mode 100644 index 8d8dc76b0..000000000 --- a/.changeset/hwstats-host-cpu.md +++ /dev/null @@ -1,5 +0,0 @@ ---- -"github.com/livekit/protocol": patch ---- - -hwstats: add WithHostCPU option to measure whole-host CPU from /proc/stat diff --git a/.changeset/resource-notifier-drain-wait.md b/.changeset/resource-notifier-drain-wait.md deleted file mode 100644 index b52c2502c..000000000 --- a/.changeset/resource-notifier-drain-wait.md +++ /dev/null @@ -1,6 +0,0 @@ ---- -"github.com/livekit/protocol": patch -"@livekit/protocol": patch ---- - -webhook: `ResourceURLNotifier.Stop(false)` waits for queued events to be sent, as `URLNotifier.Stop(false)` does diff --git a/.changeset/sdp-codec-helpers.md b/.changeset/sdp-codec-helpers.md deleted file mode 100644 index 3c3ce0f77..000000000 --- a/.changeset/sdp-codec-helpers.md +++ /dev/null @@ -1,6 +0,0 @@ ---- -"@livekit/protocol": patch -"github.com/livekit/protocol": patch ---- - -sdp: add CodecsFromSessionDescription and ParseFmtp helpers diff --git a/.changeset/sim-sip-call-to.md b/.changeset/sim-sip-call-to.md deleted file mode 100644 index fe8a1bfda..000000000 --- a/.changeset/sim-sip-call-to.md +++ /dev/null @@ -1,6 +0,0 @@ ---- -"@livekit/protocol": patch -"github.com/livekit/protocol": patch ---- - -agent_simulation: dial a phone number over SIP instead of dispatching an agent diff --git a/.changeset/sip-dialtone-preset.md b/.changeset/sip-dialtone-preset.md deleted file mode 100644 index 5ffe3dc6b..000000000 --- a/.changeset/sip-dialtone-preset.md +++ /dev/null @@ -1,6 +0,0 @@ ---- -"github.com/livekit/protocol": patch -"@livekit/protocol": patch ---- - -Add `SIPDialtone` to select a regional dial tone on CreateSIPParticipant and TransferSIPParticipant; deprecate `play_dialtone` diff --git a/.changeset/sip-drop-audio-during-dtmf.md b/.changeset/sip-drop-audio-during-dtmf.md deleted file mode 100644 index e6b855713..000000000 --- a/.changeset/sip-drop-audio-during-dtmf.md +++ /dev/null @@ -1,6 +0,0 @@ ---- -"@livekit/protocol": patch -"github.com/livekit/protocol": patch ---- - -sip: add `SIPMediaConfig.drop_audio_during_dtmf` to stop sending audio RTP while DTMF is sent diff --git a/.changeset/soft-webs-rush.md b/.changeset/soft-webs-rush.md deleted file mode 100644 index efc08c840..000000000 --- a/.changeset/soft-webs-rush.md +++ /dev/null @@ -1,6 +0,0 @@ ---- -"github.com/livekit/protocol": patch -"@livekit/protocol": patch ---- - -SIP authentication: compare IP masks against remote address rather than From header. diff --git a/CHANGELOG.md b/CHANGELOG.md index b5d5f5650..f70252c94 100644 --- a/CHANGELOG.md +++ b/CHANGELOG.md @@ -1,5 +1,31 @@ # github.com/livekit/protocol +## 1.53.2 + +### Patch Changes + +- decodebudget: move the protobuf/JSON decoding budgets to `utils/decodebudget` and add a calibrated decoded-allocation estimate (`MaxDecodedBytes`, `Estimate`, `ValidateProtoJSON`) - [#1920](https://github.com/livekit/protocol/pull/1920) ([@paulwe](https://github.com/paulwe)) + +- decodebudget: charge repeated occurrences of a singular message field against the merged message, closing a quadratic undercount - [#1921](https://github.com/livekit/protocol/pull/1921) ([@paulwe](https://github.com/paulwe)) + +- Replace `NodeStats.forward_scheduling_delay` with `NodeStats.forward_processing_per_write`, the mean forwarding work per write to a subscriber - [#1895](https://github.com/livekit/protocol/pull/1895) ([@boks1971](https://github.com/boks1971)) + +- hwstats: ignore LIVEKIT_CPU_REQUEST when measuring whole-host CPU - [#1904](https://github.com/livekit/protocol/pull/1904) ([@boks1971](https://github.com/boks1971)) + +- hwstats: add WithHostCPU option to measure whole-host CPU from /proc/stat - [#1903](https://github.com/livekit/protocol/pull/1903) ([@boks1971](https://github.com/boks1971)) + +- webhook: `ResourceURLNotifier.Stop(false)` waits for queued events to be sent, as `URLNotifier.Stop(false)` does - [#1901](https://github.com/livekit/protocol/pull/1901) ([@mnjalale](https://github.com/mnjalale)) + +- sdp: add CodecsFromSessionDescription and ParseFmtp helpers - [#1899](https://github.com/livekit/protocol/pull/1899) ([@anunaym14](https://github.com/anunaym14)) + +- agent_simulation: dial a phone number over SIP instead of dispatching an agent - [#1858](https://github.com/livekit/protocol/pull/1858) ([@u9g](https://github.com/u9g)) + +- Add `SIPDialtone` to select a regional dial tone on CreateSIPParticipant and TransferSIPParticipant; deprecate `play_dialtone` - [#1892](https://github.com/livekit/protocol/pull/1892) ([@dennwc](https://github.com/dennwc)) + +- sip: add `SIPMediaConfig.drop_audio_during_dtmf` to stop sending audio RTP while DTMF is sent - [#1907](https://github.com/livekit/protocol/pull/1907) ([@alexlivekit](https://github.com/alexlivekit)) + +- SIP authentication: compare IP masks against remote address rather than From header. - [#1889](https://github.com/livekit/protocol/pull/1889) ([@alexfish8](https://github.com/alexfish8)) + ## 1.53.1 ### Patch Changes diff --git a/package.json b/package.json index bc1ad3942..911345b7c 100644 --- a/package.json +++ b/package.json @@ -1,7 +1,7 @@ { "name": "github.com/livekit/protocol", "private": true, - "version": "1.53.1", + "version": "1.53.2", "scripts": { "changeset": "changeset", "ci:publish": "pnpm --filter @livekit/protocol run build && changeset publish" diff --git a/packages/javascript/CHANGELOG.md b/packages/javascript/CHANGELOG.md index 61a3bc599..f97fe70dd 100644 --- a/packages/javascript/CHANGELOG.md +++ b/packages/javascript/CHANGELOG.md @@ -1,5 +1,23 @@ # @livekit/protocol +## 1.53.2 + +### Patch Changes + +- Replace `NodeStats.forward_scheduling_delay` with `NodeStats.forward_processing_per_write`, the mean forwarding work per write to a subscriber - [#1895](https://github.com/livekit/protocol/pull/1895) ([@boks1971](https://github.com/boks1971)) + +- webhook: `ResourceURLNotifier.Stop(false)` waits for queued events to be sent, as `URLNotifier.Stop(false)` does - [#1901](https://github.com/livekit/protocol/pull/1901) ([@mnjalale](https://github.com/mnjalale)) + +- sdp: add CodecsFromSessionDescription and ParseFmtp helpers - [#1899](https://github.com/livekit/protocol/pull/1899) ([@anunaym14](https://github.com/anunaym14)) + +- agent_simulation: dial a phone number over SIP instead of dispatching an agent - [#1858](https://github.com/livekit/protocol/pull/1858) ([@u9g](https://github.com/u9g)) + +- Add `SIPDialtone` to select a regional dial tone on CreateSIPParticipant and TransferSIPParticipant; deprecate `play_dialtone` - [#1892](https://github.com/livekit/protocol/pull/1892) ([@dennwc](https://github.com/dennwc)) + +- sip: add `SIPMediaConfig.drop_audio_during_dtmf` to stop sending audio RTP while DTMF is sent - [#1907](https://github.com/livekit/protocol/pull/1907) ([@alexlivekit](https://github.com/alexlivekit)) + +- SIP authentication: compare IP masks against remote address rather than From header. - [#1889](https://github.com/livekit/protocol/pull/1889) ([@alexfish8](https://github.com/alexfish8)) + ## 1.53.1 ### Patch Changes diff --git a/packages/javascript/package.json b/packages/javascript/package.json index f229bd6b0..282e4c331 100644 --- a/packages/javascript/package.json +++ b/packages/javascript/package.json @@ -1,6 +1,6 @@ { "name": "@livekit/protocol", - "version": "1.53.1", + "version": "1.53.2", "description": "", "type": "module", "require": "dist/index.cjs",