WebRTC RTC publish streams use timer callbacks (`SrsRtcPublishRtcpTimer`
and `SrsRtcPublishTwccTimer`) that can cause race conditions in SRS's
coroutine-based architecture. The timer callbacks are heavy functions
that may trigger coroutine switches, during which the timer object can
be freed by another coroutine, leading to use-after-free crashes.
The race condition occurs because:
1. Timer callbacks (`on_timer`) perform heavy operations that can yield
control
2. During coroutine switches, other coroutines may destroy the timer
object
3. When control returns, the callback continues executing on a freed
object
Fixes potential crashes in WebRTC RTC publish streams under high
concurrency.
This PR introduces a major refactoring to replace `SrsSharedPtrMessage`
with `SrsMediaPacket` throughout the SRS codebase, providing a more
unified and cleaner approach to media packet handling.
---------
Co-authored-by: OSSRS-AI <winlinam@gmail.com>
Move global xpps statistics variables from `srs_app_server.cpp` to
`srs_kernel_kbps.cpp`.
Extract global shared timers from `SrsServer` into new `SrsSharedTimer`
class.
Extract WebRTC session management logic from `SrsServer` into dedicated
`SrsRtcSessionManager` class.
Extract PID file handling into dedicated `SrsPidFileLocker` class.
---------
Co-authored-by: OSSRS-AI <winlinam@gmail.com>
This PR consolidates the SRT and RTC server functionality into the main
SrsServer class, eliminating the separate `SrsSrtServer` and
`SrsRtcServer` classes and their corresponding adapter classes. This
architectural change simplifies the codebase by removing the hybrid
server pattern and integrating all protocol handling directly into
`SrsServer`.
As unified connection manager (`_srs_conn_manager`) for all protocol
connections, all incoming connections are checked against the same
connection limit in `on_before_connection()`. This enables consistent
connection limits: `max_connections` now protects against resource
exhaustion from any protocol, not just RTMP.
Remove modules because it's not used now, so only keep the server
application module and main entry point. Remove the wait group to run
server, instead, directly run server and invoke the cycle method.
After this PR, the startup workflow and servers architecture should be
much easier to maintain.
---------
Co-authored-by: OSSRS-AI <winlinam@gmail.com>
This PR fixes a critical race condition in SRS source managers where
multiple coroutines could create duplicate sources for the same stream.
- **Atomic source creation**: Source lookup, creation, and pool
insertion now happen atomically within lock scope
- **Consistent interface**: Standardize on `ISrsRequest*` interface
throughout codebase
- **Handler simplification**: Remove `ISrsLiveSourceHandler*` parameter,
obtain from global server instance
---------
Co-authored-by: OSSRS-AI <winlinam@gmail.com>
**Introduce**
This pull request builds upon the foundation laid in
https://github.com/ossrs/srs/pull/4289 . While the previous work solely
implemented unidirectional HEVC support from RTMP to RTC, this
submission further enhances it by introducing support for the RTC to
RTMP direction.
**Usage**
Launch SRS with `rtc2rtmp.conf`
```bash
./objs/srs -c conf/rtc2rtmp.conf
```
**Push with WebRTC**
Upgrade browser to Chrome(136+) or Safari(18+), then open [WHIP
encoder](http://localhost:8080/players/whip.html?schema=http&&codec=hevc),
push stream with URL that enables HEVC by query string `codec=hevc`:
```bash
http://localhost:1985/rtc/v1/whip/?app=live&stream=livestream&codec=hevc
```
This query string `codec=hevc` is used to select the video codec, and
generate lines in the answer SDP.
```
m=video 9 UDP/TLS/RTP/SAVPF 49 123
a=rtpmap:49 H265/90000
```
The encoder log also show the codec:
```
Audio: opus, 48000HZ, channels: 2, pt: 111
Video: H265, 90000HZ, pt: 49
```
**Play with RTMP**
Play HEVC stream via RTMP.
```bash
ffplay -i rtmp://localhost/live/livestream
```
You will see the codec in logs:
```
Stream #0:0: Audio: aac (LC), 48000 Hz, stereo, fltp
Stream #0:1: Video: hevc (Main), yuv420p(tv, bt709), 320x240, 30 fps, 30 tbr, 1k tbn
```
You can also use [WHEP
player](http://localhost:8080/players/whep.html?schema=http&&codec=hevc)
to play the stream.
Important refactor with AI:
* [AI: Refactor packet cache for RTC frame
builder.](b8ffa1630e)
* [AI: Refactor the packet copy and free for
SrsRtcFrameBuilder](f3487b45d7)
* [AI: Refactor the frame detector for
SrsRtcFrameBuilder](4ffc1526b9)
* [AI: Refactor the packet_video_rtmp for
SrsRtcFrameBuilder](81f6aef4ed)
* [AI: Add utests for
SrsCodecPayload.codec](61eb1c0bfc)
* [AI: Add utests for VideoPacketCache in
SrsRtcFrameBuilder.](fd25480dfa)
* [AI: Add utests for VideoFrameDetector in
SrsRtcFrameBuilder.](b4aa977bbd)
* [AI: Add regression test for RTC2RTMP with
HEVC.](5259a2aac3)
---------
Co-authored-by: Jacob Su <suzp1984@gmail.com>
Co-authored-by: winlin <winlinvip@gmail.com>
* Replaced all occurrences of sprintf with snprintf to address deprecation warnings
* Ensured proper buffer size is passed to snprintf to prevent potential buffer overflows
* Ran tests to confirm that the changes do not introduce any new issues or regressions
---------
Co-authored-by: ChenGH <chengh_math@126.com>
* fix annotation spell failed
* RTC to RTMP using SenderReport to sync av timestamp
* update pion/webrtc versio from v3.0.4 -> v3.0.13, auto config sender/receiver report
* Add rtc push flv play regression test
* Add unit test of ntp and av sync time
* Take flag CXX to makefile of utest
* Add annotation about rtc unit test
* Fix compiler error in C++98
* Add FFmpeg log callback funciton.
