This PR refactors the stream bridge architecture in SRS to improve code
organization, type safety, and maintainability by replacing the generic
ISrsStreamBridge interface with protocol-specific bridge classes and
target interfaces.
1. New Target Interface Architecture:
- Introduces ISrsFrameTarget for AV frame consumers (RTMP sources)
- Introduces ISrsRtpTarget for RTP packet consumers (RTC sources)
- Introduces ISrsSrtTarget for SRT packet consumers (SRT sources)
2. Protocol-Specific Bridge Classes:
- SrsRtmpBridge: Converts RTMP frames to RTC/RTSP protocols
- SrsSrtBridge: Converts SRT packets to RTMP/RTC protocols
- SrsRtcBridge: Converts RTC packets to RTMP protocol
3. Simplified Bridge Management:
- Removes the generic SrsCompositeBridge chain pattern
- Each source type now uses its appropriate bridge type directly
With this improvement, you are able to implement very complex bridge and
protocol converting, for example, you can bridge RTMP to RTC with opus
audio when you support enhanced RTMP with opus.
Another plan is to support bridging RTC to RTSP, directly without
converting RTP to media frame packet, but directly deliver RTP packet
from RTC source to RTSP source.
---------
Co-authored-by: OSSRS-AI <winlinam@gmail.com>
The issue occurred when srs_rtp_seq_distance(start, end) + 1 resulted in
values <= 0
due to sequence number wraparound (e.g., when end < start). This caused
assertion
failures and server crashes.
SrsRtcFrameBuilder::check_frame_complete(): Added validation to return
false
for invalid sequence ranges instead of asserting.
However, it maybe cause converting RTC to RTMP stream failure, because
this issue
should be caused by the problem of sequence number of RTP, which means
there potentially
be stream problem in RTC stream. Even so, changing assert to warning
logs is better,
because SRS should not crash when stream is corrupt.
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Co-authored-by: OSSRS-AI <winlinam@gmail.com>
try to fix#4428.
## Cause
rtmp do not support hevc, rtmp enhanced do.
## How to reproduce
1. start srs.
`./objs/srs -c conf/srt.conf`
2. publish hevc (h.265) stream to srs by srt.
`ffmpeg -re -i ./doc/source.flv -c:v libx265 -crf 28 -preset medium -c:a
copy -pes_payload_size 0 -f mpegts
'srt://127.0.0.1:10080?streamid=#!::r=live/livestream,m=publish'`
3. probe the rtmp stream
`ffprobe rtmp://localhost/live/livestream`
## About the Failed BlackBox test
The failed blackbox test: `TestSlow_SrtPublish_RtmpPlay_HEVC_Basic`
`TestSlow_SrtPublish_HttpFlvPlay_HEVC_Basic`
### Cause:
The ffmpeg 5 is used to record a piece of video (DRV), the ffmpeg will
transcode the enhanced flv format to TS format, but ffmpeg 5 don't
support enhanced rtmp (or flv) in this case.
The solution is to replace the ffmpeg to version 7 in those 2 test
cases.
### why not upgrade ffmpeg to version 7?
The black tests dependency on ffmpeg 5 will fail, and there are a few of
them are not easy to resolve in ffmpeg 7.
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Co-authored-by: winlin <winlinvip@gmail.com>
This PR significantly enhances the kernel module by adding comprehensive
unit test coverage and improving interface design for core buffer and
load balancer components.
- **ISrsDecoder**: New interface for decoding/deserialization operations
- **ISrsLbRoundRobin**: Extracted interface from concrete
SrsLbRoundRobin class for better abstraction
- **Enhanced Documentation**: Added comprehensive inline documentation
for ISrsEncoder, ISrsCodec, SrsBuffer, and SrsBitBuffer classes
---------
Co-authored-by: OSSRS-AI <winlinam@gmail.com>
This PR introduces anonymous coroutine macros for easier coroutine
creation and improves the State Threads (ST) mutex and condition
variable handling in SRS.
- **Added coroutine macros**: `SRS_COROUTINE_GO`, `SRS_COROUTINE_GO2`,
`SRS_COROUTINE_GO_CTX`, `SRS_COROUTINE_GO_CTX2`
- **Added `SrsCoroutineChan`**: Channel for sharing data between
coroutines with coroutine-safe operations
- **Simplified coroutine creation**: Go-like syntax for creating
anonymous coroutines with code blocks
---------
Co-authored-by: Jacob Su <suzp1984@gmail.com>
Co-authored-by: OSSRS-AI <winlinam@gmail.com>
Fixes a bug in WebRTC NACK packet recovery mechanism where recovered
packets were being discarded instead of processed.
In `SrsRtcRecvTrack::on_nack()`, when a retransmitted packet arrived
(found in NACK receiver), the method would:
1. ✅ Remove the packet from NACK receiver (correct)
2. ❌ Return early without adding the packet to RTP queue (BUG)
This caused recovered packets to be lost, defeating the purpose of the
NACK mechanism and potentially causing media quality issues.
Restructured the control flow in `on_nack()` to ensure both new and
recovered packets reach the packet insertion logic:
- **Before**: Early return for recovered packets → packets discarded
- **After**: Conditional NACK management + unified packet processing →
all packets queued
Closes#3820
---------
Co-authored-by: Haibo Chen <495810242@qq.com>
Co-authored-by: OSSRS-AI <winlinam@gmail.com>
This PR modernizes SRS's HTTP handling by upgrading from the legacy
http-parser library to the more performant and actively maintained
llhttp library.
* Replace http-parser with llhttp: Migrated from the deprecated
http-parser to llhttp for better performance and maintenance
* API compatibility: Updated all HTTP parsing logic to use llhttp APIs
while maintaining backward compatibility
* Simplified URL parsing: Replaced complex http-parser URL parsing with
custom simple parser implementation
Enhanced error handling: Improved error reporting with llhttp's better
error context and positioning
---------
Co-authored-by: OSSRS-AI <winlinam@gmail.com>
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 removes the embedded GB28181 SIP server implementation from SRS
and enforces the use of external SIP servers for production deployments.
The embedded SIP server depended on the deprecated `http-parser`
library. With the planned migration to `llhttp` (which doesn't support
SIP parsing), maintaining the embedded SIP server would require
significant additional work. Since external SIP servers are already the
recommended approach for production, removing the embedded
implementation simplifies the codebase and eliminates this dependency.
Eliminated `srs_gb28181_test` from CI workflow.
Removed SIP configuration validation tests.
---------
Co-authored-by: OSSRS-AI <winlinam@gmail.com>
Co-authored-by: haibo.chen <495810242@qq.com>
