Releases: AirenSoft/OvenMediaEngine
Releases · AirenSoft/OvenMediaEngine
v0.12.11
v0.12.10
v0.12.9
[Compatibility]
- Compatible with version 0.12.8 of Origin or Edge
- Compatible with version 0.12.8 of
Server.xml
andLogger.xml
.
[Added]
- WebRTC Provider sends RTCP PLI to peer
- In order to comply with regulations such as GDPR and CCPA, the
PrivacyProtection
option can be enabled to hide the client's IP information from logs and APIs. - Implemented SNI feature, every
VirtualHost
can each set their own certificate
[Improved]
- WebRTC Signaling, Thumbnail, HLS, DASH can use the same port
- WebRTC, RTSP Provider converts timestamp to PTS using RTCP SR for more accurate A/V sync
- Improved stability of WebSocket server
- Improved compatibility of RTMP Provider
- Improved compatibility of HTTPS client
- If there is no stream in Edge, pull stream from Origin and start RTMP Push (Thanks to @trapasso)
- Improved performance of SDP parser and URL parser
[Changed]
TCP_NODELAY
andTCP_QUICKACK
are enabled by default- Retry when RTMP push session is disconnected
- RTMP Push tries to reconnect when the connection is lost
[Fixed]
- Fixed the problem that
${PublicIP}
macro does not work inServer.xml
- Fixed a problem that
include
attribute doesn't work with relative path - Fixed an issue where TLS packets could not be parsed with a low probability
- Fixed an issue with incorrect bitstream conversion in RTMP Push Publisher
- Fixed a problem that could crash when handling tampered HTTP packets
- Fixed a problem that the WebRTC provider loads the publisher's bind settings (thanks to @trapasso) #580
- Fixed a problem that could crash when the log level of
SegmentStream
is set toDEBUG
(thanks to @trapasso) #584
v0.12.8
[Compatibility]
- Compatible with version 0.12.7 of Origin or Edge
- Compatible with version 0.12.7 of
Server.xml
andLogger.xml
.
[Added]
- If there is a b-frame in the input stream, a warning log is output.
- Added
<NoInputFailoverTimeout>
and<UnusedStreamDeletionTimeout>
properties toOriginMap
- Added Framemarking extension to RTP
- Added GOP statistics information to OvenRtcTester (correct values are output when testing OvenMediaEngine 0.12.8 or higher)
- Added
GET /v1/stats/current
to API server (Server's stat, sum of all VHosts)
[Improved]
- Improved compatibility of RTSP Pull
[Changed]
- Changed to use
<Server><Name>
for HLS service name - Change the default value of
<TcpForce>
to true - Change the default value of
<WebRTC><Rtx>
to false - Change the default value of
<WebRTC><Ulpfec>
to false - Changed the restart option of OvenMediaEngine service to always
[Fixed]
- Fixed a problem where the duration of AAC and OPUS frames could be calculated incorrectly
- Fixed an issue where
LastConfig.json
file was created in the wrong format - Fixed an issue that could cause a segfault in OVT Provider
v0.12.7
[Compatibility]
- Not compatible with Origin or Edge of 0.12.6
- Compatible with version 0.12.6 of
Server.xml
andLogger.xml
.
[Added]
- Added to automatically failover if multiple URLs are set in Edge's Origin Map (both
OVT
andRTSP
)- When the input stream is changed, it affects the publisher and has been tested in
WebRTC
andHLS
(DASH
does not work properly yet)
- When the input stream is changed, it affects the publisher and has been tested in
- Added presets of encoding quality and performance
- Added the ability to maintain original quality in Transcoding (Keep Origin)
[Improved]
- Improved performance of some depacketizers
[Fixed]
- Fixed an issue where multiple streams could not handle input in
SRT
- Fixed an issue where
OVT
andRTSPPull
could sometimes get deadlocked - Fixed
RTSP
status line being parsed incorrectly - Fixed a problem that did not work when empty space was entered in the
SignedPolicy
setting - Fixed an issue where options after the
-i
option were ignored when running OvenMediaEngine - Fixed an issue where default CORS was not applied
v0.12.6
[Compatibility]
- Compatible with version 0.12.5 of Origin or Edge.
- Compatible with version 0.12.5 of Server.xml and Logger.xml.
[Added]
- Added JitterBuffer for WebRTC Publisher (For Lip-sync) (see https://airensoft.gitbook.io/ovenmediaengine/streaming/webrtc-publishing#publisher)
[Improved]
- Improved A/V sync of WebRTC in all browsers
- Increased the limit of the number of files of OvenMediaEngine service
[Changed]
- Changed ice_servers to iceServers in the offer message of WebRTC signaling (ice_servers is not deleted yet)
- Changed the thread name AppWorker to AW-xxx (ex: AW-WebRTC)
- Reduced the number of chunk history from *5 to *2 in HLS
- Reduced the number of inbound/outbound worker threads
- Changed LastConfig.json, which saves settings applied by API, to LastConfig.xml, and fixed some errors
- Changed the OPUS encoder to the latest version
- Changed OPUS encoding option (10ms->20ms, s16->fltp, 10% loss->5% loss)
[Fixed]
- Fixed an issue where HTTP request header size greater than 1024 would cause an error
- Fixed API usage of deprecated Nvidia GPU
- Fixed an issue where WebSocket packets could not be parsed if fragmented (reproduced frequently in the latest iOS 15 or later)
v0.12.5
[Added]
- Support Amazon Linux
[Improved]
- Improve packaging performance of HLS and DASH
[Changed]
- Change the number of Inbound/Outbound Workers to half of
AppWorker
s - Reduce the size of HLS and DASH segment storage buffers
[Fixed]
- Fix the problem that
CloseWithState()
may not return normally
v0.12.4
v0.12.3
[Added]
- Added the
<TcpForce>
option to enforce WebRTC over TCP. - Added simulator for WebRTC performance measurement.
- Added segmentation rule setting option in file recording.
[Improved]
- AES GCM has been applied to SRTP.
- Improved Socket stability.
- Reduced the number of unnecessary threads in the transcoding filter.
[Changed]
- Changed
libsrtp
version to v2.4.0
[Fixed]
v0.12.2
[Changed]
- OvenMediaEngine's license changed from GPLv2 to GPLv3
- Upgraded OpenSSL to v3.0.0-beta1
- Changed the ffmpeg preset of the H.264 encoder from ultrafast to faster
[Added]
- Supports Hardware Encoding (Beta)
- Supports AdmissionWebhooks (Beta)
- Added Dockerfile,nv for Nvidia GPU
- Supports WebM container in recording module
[Improved]
- HTTPClient supports TLS
- Improved compatibility with some RTMP encoders
- Apply nasm to vpx, x264 library (Added back what was missing in 0.12.1)
- To improve server stability, forcibly terminate a session that is receiving very slowly for a long time
- Improved compatibility with some RTMP servers in the RTMP push publisher
[Fixed]
- Fix crash when using SignedPolicy in HLS
- Fix to parse id:password@ in RTSP URL
- Fix OOM issue in transcoder when timestamp increases abnormally
- Resolves the problem that macros do not work in the recording API
- Resolves a crash problem when accessing thumbnail publisher with wrong URL
- Fix to clean up the socket when the RTSP client is terminated
- Improved performance by fixing an issue with sockets being over-queued