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.
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Co-authored-by: OSSRS-AI <winlinam@gmail.com>
Co-authored-by: haibo.chen <495810242@qq.com>
This PR makes WebRTC a core feature of SRS and enforces C++98
compatibility by:
1. Always Enable WebRTC Support
- Remove `--rtc=on|off` configuration option - WebRTC is now always
enabled
- Eliminate all `#ifdef SRS_RTC` conditional compilation blocks
- Include WebRTC-related modules (RTC, SRTP, DTLS) in all builds
- Update build scripts to always link WebRTC dependencies
2. Enforce C++98 Compatibility
- Remove `--cxx11=on|off` and `--cxx14=on|off` configuration options
- Force `SRS_CXX11=NO` and `SRS_CXX14=NO` in build system
- Move these options to deprecated section with warnings
- Ensure codebase maintains C++98 standard compatibility
3. Remove Windows/Cygwin Support
- Remove all Windows and Cygwin64 conditional compilation blocks (#ifdef
_WIN32, #ifdef CYGWIN64)
- Delete Cygwin64 build configurations from build scripts (
auto/options.sh, auto/depends.sh, configure)
- Remove Cygwin64 assembly files and State Threads platform support (
md_cygwin64.S)
- Eliminate Windows-specific GitHub Actions workflows and CI/CD jobs
- Remove NSIS packaging files and Windows installer generation
- Delete Windows documentation and update feature lists to mark support
as removed in v7.0
- Simplify OS detection to only support Unix-like systems (Linux, macOS)
4. Code Cleanup
- Remove conditional WebRTC code blocks throughout the codebase
- Simplify build configuration by removing WebRTC-related conditionals
- Update constructor delegation patterns to be C++98 compatible
- Fix vector initialization to use C++98 syntax
- Eliminate Windows-specific implementations for file operations, time
handling, and networking
- Unified platform handling with consistent POSIX API usage
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Co-authored-by: OSSRS-AI <winlinam@gmail.com>
In the scenario of converting WebRTC to RTMP, this conversion will not
proceed until an SenderReport is received; for reference, see:
https://github.com/ossrs/srs/pull/2470.
Thus, if HTTP-FLV streaming is attempted before the SR is received, the
FLV Header will contain only audio, devoid of video content.
This error can be resolved by disabling `guess_has_av` in the
configuration file, since we can guarantee that both audio and video are
present in the test cases.
However, in the original regression tests, the
`TestRtcPublish_HttpFlvPlay` test case contains a bug:
5a404c089b/trunk/3rdparty/srs-bench/srs/rtc_test.go (L2421-L2424)
The test would pass when `hasAudio` is true and `hasVideo` is false,
which is actually incorrect. Therefore, it has been revised so that the
test now only passes if both values are true.
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Co-authored-by: Copilot <175728472+Copilot@users.noreply.github.com>
Co-authored-by: winlin <winlinvip@gmail.com>
HLS typically has a delay of around 30 seconds, roughly comprising three
segments, each lasting 10 seconds. We can reduce the delay to about 5
seconds by lowering the segment duration to 2 seconds and starting
playback from the last segment, achieving a stable delay.
Of course, this requires setting the OBS's GOP to 1 second, and the
profile to baseline, preset to fast, and tune to zerolatency.
Additionally, updating a few configurations in the hls.js player is
necessary, such as setting it to start playback from the last segment,
setting the maximum buffer, and initiating accelerated playback to
reduce latency.
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Co-authored-by: chundonglinlin <chundonglinlin@163.com>
Co-authored-by: john <hondaxiao@tencent.com>
The `ffmpeg-opus` tool allows you to control the delay using the
`opus_delay` option. The minimum delay can be set to 2.5ms. However, in
practice, you cannot set it this low. You need to set at least 10 frames
to allow the audio encoder to lookahead. Otherwise, the sound will be
distorted.
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Co-authored-by: chundonglinlin <chundonglinlin@163.com>
1. After enabling FFmpeg opus, the transcoding time for each opus packet
is around 4ms.
2. To speed up case execution, our test publisher sends 400 opus packets
at intervals of 1ms.
3. After the publisher starts, wait for 30ms, then the player starts.
4. Due to the lengthy processing time for each opus packet, SRS
continuously receives packets from the publisher, so it doesn't switch
coroutines and can't accept the player's connection.
5. Only after all opus packets are processed will it accept the player
connection. Therefore, the player doesn't receive any data, leading to
the failure of the case.
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Co-authored-by: winlin <winlinvip@gmail.com>
### Description
When converting between AAC and Opus formats (aac2opus or opus2aac), the
`av_frame_get_buffer` API is frequently called.
### Objective
The goal is to optimize the code logic and reduce the frequent
allocation and deallocation of memory.
In the case of aac2opus, av_frame_get_buffer is still frequently called.
In the case of opus2aac, the goal is to avoid calling
av_frame_get_buffer and reduce memory allocations.
### Additional Note
Before calling the `av_audio_fifo_read` API, use
`av_frame_make_writable` to check if the frame is writable. If it is not
writable, create a new frame.
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Co-authored-by: john <hondaxiao@tencent.com>
For the DJI M30, there is a bug where empty NALU packets with a size of
zero are causing issues with HLS streaming. This bug leads to random
unpublish events due to the SRS disconnecting the connection for the HLS
module when it fails to handle empty NALU packets.
To address this bug, we have patched the system to ignore any empty NALU
packets with a size of zero. Additionally, we have created a tool in the
srs-bench to replay pcapng files captured by tcpdump or Wireshark. We
have also added utest using mprotect and asan to detect any memory
corruption.
It is important to note that this bug has been fixed in versions 4.0.271
6477f31004 and 5.0.170
939f6b484b. This patch specifically
addresses the issue in SRS 6.0.
Please be aware that there is another commit related to this bug that
partially fixes the issue but still leaves a small problem for asan to
detect memory corruption. This commit,
577cd299e1, only ignores empty NALU
packets but still reads beyond the memory.
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Co-authored-by: chundonglinlin <chundonglinlin@163.com>
1. Introduce a novel Docker tag in the x.y.z format, akin to the HELM
format, such as ossrs/srs:5.0.155.
2. Incorporate the SRS_PLATFORM flag for containers initiated through
HELM.
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`TRANS_BY_GPT3`
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Co-authored-by: chundonglinlin <chundonglinlin@163.com>
1. Enable blackbox test for each PR and push.
2. Refine Makefile and README for srs-bench.
3. Add blackbox using FFmpeg and ffprobe.
4. Add blackbox basic test for RTMP stream.
5. Add blackbox basic test for HTTP-FLV stream.
6. Fix utest rand seed issue.
PICK 2141d220b4
1. Build regression test tool for gb28181.
2. Run regression test for gb28181.
3. Format go code and eliminate logs.
4. Change base docker to ubuntu20.
PICK 7750bdae10
