Releases
ams-v2.12.0
Ant Media Server Community 2.12.0
Latest
mekya
released this
12 Dec 07:45
Highlights
Update FFmpeg to 7.1
Support WebP, JPEG previews
Support ID3 in HLS HEVC streams
Support USB stick licensing
Optimize MongoDB utilization
Ant-Media-Server
6862 Refactor Subscriber & Add setting to write events to datastore #6860 #6861
6857 Fix big app installation in clusters - #6777
6852 Support WebP Previews - #6859
6849 Add user publish id constant - ant-media/conference-call-application#426
6844 Handle port number suffixes in IP filters - #6838 @jantekb
6842 Add JPG support for previews - #6841
6833 Add subFolder AppSettings with mainTrackId and streamId variable support - #6821
6829 Add Debian support to enable_ssl.sh - #6792
6828 Reconnect faster and shutdown faster - #6786
6822 Cache subscriber to decrease query count - #6783
6816 Add get subtrack count commands - ant-media/conference-call-application#403
6814 Add Spring Packages used in applications -
6811 Update FFmpeg to 7.1 - #6804
6784 Fix typo in getTOTP REST method output -
6769 Fix HLS Modifier Returning Wrong Content in Case of Https - #6308
6767 Fix allowable values in Broadcast definitions - thank you to @Himpoke - #6740
6764 Add stream name to the stream status webhook -
6763 Return error in getToken REST Method When token type is not valid - #6744
6752 Resolve CVE 202 13956 - #6750
6739 Add allowableValues expects an array to map to enum -
6735 Make encoding Q size configurable - #6705
6734 Fix Wrong Memory Display If AMS is running in a Container - #6626
6726 Upgrade FFmpeg to 7.1 - #6551
6721 Correct JWT function name in enable_ssl.sh -
6718 Call webhook for stream status & Make streamId unique index in MongoDB for Broadcasts - #3847
6717 Call webhook for stream status - #6711
6709 Split RTMP StreamId by slash(/) to parse Token and TOTP - #6704
6701 add getRemoteIpAddress into IWebRTCClient - #6638
6688 Add SRT latency configuration in application settings - #6665
6687 Add correct IP address in Docker cluster environments - #6707
6686 Fix enable_ssl.sh script warnings in Docker environment - #6706
6683 Add latitude, longitude, altitude, metadata and description fields to VoD - #6662
6676 Fix frame size format in ID3 header
6661 Fix unexpected null pointer exception for streamId in logs -
6656 Add custom text support in stream endpoints for VoD & HLS - #6650
6644 Support mp3 file upload - #6643
6630 Add getter for dbId field in AppSettings - #6629
6620 Improve rtmp overflow check because it causes a test fail in enterprise - #6427
6613 Improve VoD Upload API - #6612
6611 Get REST API JWT From Authorization Bearer - #6516
6610 Add Settings Field to Stop WebRTC Ingest if no audio/video packet is received for the timeout duration - #6605
6609 Reverted - Ignore this PR: Stop WebRTC Ingest if no audio packet is received for the timeout duration - #6605
6604 Relay RTMP metadata to Muxers writeMetadata method - #6550
6602 Allow RTSPS urls as stream source -
6598 Remove date extension from the access log filename to rotate the log files
6559 Fix RTMP timestamp overflow after 24 days of non-stop streaming - #6486
6457 Fix logging statement
6411 Support for role based streaming - host, speaker, attendee - ant-media/conference-call-application#272
6352 Refactor Installing SSL in Marketplace Instances -
Ant-Media-Enterprise
Fix big app installation in clusters- #6777
Fix intermittently unplayable tracks in video conferencing- ant-media/conference-call-application#421
Support WebP and atomic move in preview files-
Pass user publish id in play request- ant-media/conference-call-application#426
Add JPG support for previews- #6841
Make tokenService use AppSettings' secretCode first- #6783
Support dynamic broadcast subfolder- #6821
Refactor S3 upload integration-
Shutdown faster in cluster mode and check if instance alive- #6832
Add get subtrack count- ant-media/conference-call-application#403
Update FFmpeg(7.1) and SRT(1.5.3)- #6804
Add integration test code for HLSManifestModifier- #6697
Create broadcast during join room if necessary- ant-media/conference-call-application#369
Add automated release upload to S3-
Fix: Data channel messages cannot send by play only users in cluster mode- #6758
Fix WebRTC Playback when OBS publishes with NVENC H.264 Encoder- #4643
Make SRT inactivity timeout increase further to 10-20secs- #6736
Check usb serial for licence validation-
Fix connecting to signaling server- #6747
Make encoding Q size configurable- #6705
Upgrade to FFmpeg 7.1-1.5.11-SNAPSHOT- #6551
Handle WHIP codec mismatch- #6723
Make handleMainTrack synchronized and use unique index in mongodb-
Add logs for video track count increase-
Add Integration test for spliting RTMP StreamId by slash(/) to parse Token and TOTP- #6704
Get remote IP address from WebRTC Stats- #6638
Configure SRT Latency through Application Settings- #6665
Don't make http request to origin for orphan streams- #6383
Mp4 recording with restore test- #6651
Add woocommerce automation: Update downloadable file in products-
Put audio packets in queue, update audio/video timing after restore- #5601
Fix Wrong Overflow Trigger in Audio & Video- #6632
Delete old AppSettings's index if exists to not have IndexConflict in MongoDB 4.4- #6629
Stop connection(not immediately) when ICE is disconnected because it already waits 10 secs to let the client reconnect- #6427
Add Settings Field to Stop WebRTC Ingest if no audio/video packet is received for the timeout duration- #6605
Add Settings Field to Stop WebRTC Ingest if no audio packet is received for the timeout duration- #6605
Draft: Add metadata to hls-
Add reserved status info in the video track assignments- ant-media/conference-call-application#262
Support for role based streaming - host, speaker, attendee- ant-media/conference-call-application/issues/272 ant-media/conference-call-application/issues/311
StreamApp
522 Extends fields for join/leave commands - ant-media/conference-call-application#369
521 Pass user publish streamId in play request - ant-media/conference-call-application#426
519 Reconnect faster when connection is failed/closed - #6786
517 Add mechanism to get subtrack count - ant-media/conference-call-application#403
516 Refactor dependency version to use variable -
512 fix configuration - Commentout keycloak configuration by default.
508 Key cloak integration - #6770
507 Add option to debug signaling endpoint - #6747
506 Add playback stats for WebRTC and add test codes -
505 Support role in conference sample -
499 Update update-embedded-player.yml - update embedded player version auto
498 Update update-embedded-player.yml - automaticly create PR on new web player
497 Update web player version to 2.11.2 -
496 Update @antmedia/web_player version to 2.11.1 -
494 Update web player version - #6643
492 Create update-embedded-player.yml - update embedded player version
491 Update web player version - #6682
488 Increase the wait time of resolution change process - #6424
487 Video track cannot stop properly in case of the virtual backgorund change - ant-media/conference-call-application#336
485 show the assignment in conference sample page - Show the video track assignment in conference sample page
484 Reset websocket connections flags to re-connect faster - #6427
Ant-Media-Server-Parent
148 Fix Kafka vulnerability CVE-2024-31141 -
147 Update spring-security version - #6815
146 Add Spring dependencies -
145 Update FFmpeg(7.1) and SRT(1.5.3) - #6804
144 Fix CVE-2024-52316 : Update Tomcat to 10.1.30 - #6808
143 Update spring core version to 6.2.0 -
142 CVE-38821/22234/22257/CVE-2023-52428 - CVE-2024-22234 CVE-2024-22257 CVE-2023-52428 #6794
141 Fix CVE-2023-6378 - #6793
140 Upgrade FFmpeg to the latest - #6551
139 Update versions for vulnerabilities CVE-2024-7254 , CVE-2024-8391 -
Ant-Media-Management-Console
267 Enable SSL through Ant Media Subdomain for Marketplace Builds - #6807
265 Add JWT to VoD URL During Playlist Item Selection - #6727
264 Update webplayer version 2.11.2 -
263 Update @antmedia/web_player version to 2.11.1 -
262 Update web player for mp3 playback - #6643
261 Support mp3 files to upload - #6643
260 Allow RTSPS urls while creating stream source - Accept rtsps URLs as Stream Source
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