v0.10.8

[Fixed]

• Fixed an issue that would occur if multiple Origins were set up on one Edge (#197). Special thanks to @deepsurvey
• Fixed a problem of SIGSEGV when disconnecting RTMP connection during Origin/Edge is working.
• Fixed a problem of increasing RAM usage if the PTS increases rapidly (#161).
• Fixed an issue that might cause SIGSEGV when deleting a stream.
• Modified not to use jemalloc (#171).

[Improved]

• InputStream can be created without track information.
• Modified TCP keep alive to work (#175).
• Modified to generate SPS/PPS information periodically (#197).
• Now, the connection will be lost if you send it to an app that is not set in Server.xml.
• Modified the warning log when receiving a host in mDNS from ICE candidate (#177).

v0.10.7

[Fixed]

• Fixed a problem that SIGSEGV occurred intermittently in WebRTC (#114, #163, #164). Special thanks to @stefanomazzoni, @nangavar and @alex-jhones for finding and analyzing the problem.
• Fixed an issue where the RTSP pull provider does not reconnect when the server disconnects. (#168)

[Improved]

• Improved to support the case of dynamically changing sample rates in AAC (#157). Special thanks to @VTFLAB for finding and analyzing the problem.

v0.10.6

[Fixed]

• Fixed a problem where the resource was not cleaned up when the client was disconnected
• Fixed a problem where the RtspcProvider timeout did not work properly (#150)

[Changed]

• Changed queue size (#155)

v0.10.5

[Fixed]

• Fixed a typo in the configuration module (#144, #147)
• Fixed to install NASM as source code in CentOS and Fedora (#146)
• Fixed MPEG-TS setting provided in sample Server.xml (#145)
• Fixed Origin-Relay-Edge issue (#148)

v0.10.4

[Added]

• Support MPEG-TS/UDP input (Beta)
• Support RTSP Pull input (Beta)
• Support Audio/Video only input stream
• Added an output stream with the same name as the input stream's name automatically generated
  • The output stream name in the default configuration file is now the same as the input stream name. (Remove _o from the existing <Input Stream Name>_o)
  • For details, see https://airensoft.gitbook.io/ovenmediaengine/v/0.10.4/getting-started#play

[Fixed]

• Fixed a problem that occur when the PTS does not increase monotonously in RTSP
• Fixed a problem of crash when playing WebRTC due to SDP update
• Fixed a problem caused by timing issues that update the SDP during WebRTC playback
• Fixed a problem with multiple input streams in RTMP

[Changed]

• Refactored Provider/Media Router codes

v0.10.3

[Fixed]

• Fixed the problem that the session log is repeated
• Fixed several memory leaks
• Fixed HLS not working in Edge clustering
• Fixed an issue that caused a forced termination when deleting a stream

[Changed]

• The performance of RtspcProvider and HlsPublisher has been improved

v0.10.2

NOTE: Some external libraries have been changed and added, so you have to run prerequisites.sh again.

[Added]

• Use jemalloc to improve memory allocation performance
• Supported audio for RtspcProvider

[Fixed]

• Fixed a problem that the P2P module didn't work
• Fixed jitter correction errors
• Fixed several memory leaks
• Fixed a crash issue by memory allocation
• Removed warning on compile

[Changed]

• The performance of RtspcProvider/OvtStream/Orchestrator/ServerSocket have been improved

v0.10.1

[Added]

• Unnecessary providers and publishers can be turned off
• Added a correction algorithm to correct LL-DASH timestamp
• Organized source codes related to VHost

[Fixed]

• Rewritten algorithm to detect SPS/PPS
• Fixed high CPU usage problem
• Fixed an issue that could cause crash when using OvtPacketizer

[Changed]

• Changed OpenSSL download URL in prerequisites.sh

v0.10.0

[Added]

• Streaming Server
  • Support virtual hosts
  • Support multiple applications
• Origin-Edge
  • OVT pull provider and OVT publisher for origin-edge structure
  • OriginMap with Origins configuration
  • Pulling a origin stream dynamically when an user requests
• Providers
  • RTSP pull provider (beta)
• Publishers
  • Low-Latency MPEG-DASH
  • Improved AV sync of WebRTC by adding RTCP SR
  • Implemented WebSocket ping/pong specification
• Transcoding
  • Added bypass profile
  • Optimized transcoder for performance
• Statistics
  • Added a Monitoring module to collect statistics metrics
  • WebRTC, HLS and DASH statistics log output (beta)
• Etc.
  • Support Docker

[Fixed]

• Improved RTMP stability
• Improved HTTP Server stability
• Fixed several transcoding related errors
• Fixed several memory leaks

[Changed]

• Changed configuration file structure
• Changed the encoder for H.264 from OPENH264 to x264
• Refactoring to increase code readability
• Moved all OvenMediaEngine related libraries to /opt/ovenmediaengine to avoid conflicts

v0.9.1

[0.9.1]

Fixed:

- Modify to output more detailed logs when an error related to OpenSSL occurs
- Changed default ICE candidate list
- Organize source code of bitstream converter
- Fixed a timestamp calculation problem
- Fixed a problem with input buffer deletion
- Fixed an issue with incorrect pts processing