This PR adds separate audio and video frame counting to the HTTP API
(`/api/v1/streams/`) for better stream observability. The API now
reports three frame fields:
- `frames` - Total frames (video + audio)
- `video_frames` - Video frames/packets only
- `audio_frames` - Audio frames/packets only
This enhancement provides better visibility into stream composition and
helps detect issues with CBR/VBR streams, audio/video sync problems, and
codec-specific behavior.
**Before:**
```json
{
"streams": [
{
"frames": 0, // video frames.
}
]
}
```
**After:**
```json
{
"streams": [
{
"frames": 6912, // video frames.
"audio_frames": 5678, // audio frames.
"video_frames": 1234, // video frames.
}
]
}
```
Frame Counting Strategy
- All protocols report frames every N frames to balance accuracy and
performance
- Frames are counted at the protocol-specific message/packet level:
- RTMP: Counts RTMP messages (video/audio)
- WebRTC: Counts RTP packets (video/audio)
- SRT: Counts MPEG-TS messages (H.264/HEVC/AAC)
SRS forward feature only supports plain RTMP protocol, not RTMPS (RTMP over SSL/TLS). This is by design - SRS SSL is server-side only (accepting connections), not client-side (initiating connections). The forward feature uses SrsSimpleRtmpClient which has no SSL handshake or encryption capabilities for outgoing connections.
Changes:
1. Add RTMPS URL detection in SrsForwarder::initialize()
2. Return ERROR_NOT_SUPPORTED error when RTMPS destination is detected
3. Add unit test to verify RTMPS URLs are properly rejected
4. Add FAQ section to .augment-guidelines explaining the limitation
For users who need to forward to RTMPS destinations (e.g., AWS IVS), the recommended solution is to use FFmpeg with SRS HTTP Hooks:
- on_publish event: Automatically start FFmpeg to relay stream to RTMPS destination
- on_unpublish event: Automatically stop FFmpeg process when stream ends
This provides a fully automated, production-ready RTMPS relay solution without adding complexity to SRS core.
Related: #4536
---------
Co-authored-by: OSSRS-AI <winlinam@gmail.com>
Regression since 20f6cd595c
The early code might meet bridge is empty when
there is no bridge(e.x. rtc to rtc). Then srs_freep will free the brige.
Remove this code that seems redundant.
---------
Co-authored-by: Jacob Su <suzp1984@gmail.com>
Signed-off-by: Jack Lau <jacklau1222@qq.com>
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.
---------
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.
---------
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>
