## Cause
dash auto dispose is configured by seconds, but the code compare by
usecond, 1 second = 1,000,000 useconds.
releated to #4097
Bug introduced after #4097 supported Dash auto dispose after a timeout
without media data.
## How to reproduce
1. `./objs/srs -c conf/dash.conf`
2. publish a rtmp stream.
3. play dash stream. -> no dash stream, always 404 error.
---------
Co-authored-by: winlin <winlinvip@gmail.com>
To manage an object:
```cpp
// Before
MyClass* ptr = new MyClass();
SrsAutoFree(MyClass, ptr);
ptr->do_something();
// Now
SrsUniquePtr<MyClass> ptr(new MyClass());
ptr->do_something();
```
To manage an array of objects:
```cpp
// Before
char* ptr = new char[10];
SrsAutoFreeA(char, ptr);
ptr[0] = 0xf;
// Now
SrsUniquePtr<char[]> ptr(new char[10]);
ptr[0] = 0xf;
```
In fact, SrsUniquePtr is a limited subset of SrsAutoFree, mainly
managing pointers and arrays. SrsUniquePtr is better than SrsAutoFree
because it has the same API to standard unique ptr.
```cpp
SrsUniquePtr<MyClass> ptr(new MyClass());
ptr->do_something();
MyClass* p = ptr.get();
```
SrsAutoFree actually uses a pointer to a pointer, so it can be set to
NULL, allowing the pointer's value to be changed later (this usage is
different from SrsUniquePtr).
```cpp
// OK to free ptr correctly.
MyClass* ptr;
SrsAutoFree(MyClass, ptr);
ptr = new MyClass();
// Crash because ptr is an invalid pointer.
MyClass* ptr;
SrsUniquePtr<MyClass> ptr(ptr);
ptr = new MyClass();
```
Additionally, SrsAutoFreeH can use specific release functions, which
SrsUniquePtr does not support.
---------
Co-authored-by: Jacob Su <suzp1984@gmail.com>
1. fix redundant null check, there is no potential risks by the way,
just redundant null check.
2. Potential use pointer after free, that's not true. So we can ignore
this one, or find a way to make stupid security tool happy.
---------
Co-authored-by: winlin <winlinvip@gmail.com>
1. Add live benchmark support in srs-bench, which only connects and
disconnects without any media transport, to test source creation and
disposal and verify source memory leaks.
2. SmartPtr: Support cleanup of HTTP-FLV stream. Unregister the HTTP-FLV
handler for the pattern and clean up the objects and resources.
3. Support benchmarking RTMP/SRT with srs-bench by integrating the gosrt
and oryx RTMP libraries.
4. Refine SRT and RTC sources by using a timer to clean up the sources,
following the same strategy as the Live source.
---------
Co-authored-by: Haibo Chen <495810242@qq.com>
Co-authored-by: Jacob Su <suzp1984@gmail.com>
Try to fix two blackbox test:
1. TestSlow_SrtPublish_HttpTsPlay_HEVC_Basic: fixed by enlarge the wait
time from 3 seconds to 4 before run ffprobe task, which will record the
stream by ffmpeg first.
2 TestSlow_SrtPublish_HlsPlay_HEVC_Basic: fixed by wait 16 seconds
before run ffprobe task.
About the 2 case: it seems ridiculous to add 16 seconds delay before run
ffprobe task.
> So, I start #4088 to check the github action workflow process, I start
this branch from a very earlier version `6.0.113
(srs/core/srs_core_version6.hpp)`, what I found it that, inside `SRS
blackbox-test`, the srs version `6.0.128`, the latest version, was
printed. That's really wired.
I confirmed this is not the SRS source code's problem, check
https://github.com/suzp1984/srs/actions/runs/9511600525/job/26218025559
the srs code 6.0.113 was checkout and running actions based on them,
still met same problem.
---------
Co-authored-by: winlin <winlinvip@gmail.com>
The object relations:
Session manages SIP and Media object using shared resource or shared
ptr. Note that I actually use SrsExecutorCoroutine to delete the object
when each coroutine is done, because there is always a dedicate
coroutine for each object.
For SIP and Media object, they directly use the session by raw pointer,
it's safe because session always live longer than session and media
object.
---
Co-authored-by: Jacob Su <suzp1984@gmail.com>
Fix#4037 SRS should not send the publish start message
`onStatus(NetStream.Publish.Start)` if hooks fail, which causes OBS to
repeatedly reconnect.
Note that this fix does not send an RTMP error message when publishing
fails, because neither OBS nor FFmpeg process this specific error
message; they only display a general error.
Apart from the order of messages, nothing else has been changed.
Previously, we sent the publish start message
`onStatus(NetStream.Publish.Start)` before the HTTP hook `on_publish`;
now, we have modified it to send this message after the HTTP hook.
Fix#3967 There is an API `SSL_use_certificate_chain_file`, which can load the
certification chain and also single certificate.
---------
Co-authored-by: winlin <winlinvip@gmail.com>
1. `trunk/research/st/exceptions.cpp` About exceptions with ST, works
well on linux and mac, not work on cygwin.
2. `trunk/research/st/pthreads.cpp` About pthreads with ST, works well
on all platforms.
3. `trunk/research/st/hello.cpp` Hello world, without ST, works well on
all platforms.
4. `trunk/research/st/hello-world.cpp` Hello world, with ST, works well
on all platforms.
5. `trunk/research/st/hello-st.cpp` A very simple version for hello
world with ST, works well on all platforms.
When debugging the RTMP protocol, we can capture packets using tcpdump
and then replay the pcap file. For example:
```bash
cd ~/git/srs/trunk/3rdparty/srs-bench/pcap
tcpdump -i any -w t.pcap tcp port 1935
go run . -f ./t.pcap -s 127.0.0.1:1935
```
However, sometimes due to poor network conditions between the server and
the client, there may be many retransmitted packets. In such cases,
setting up a transparent TCP proxy that listens on port 1935 and
forwards to port 19350 can be a solution:
```bash
./objs/srs -c conf/origin.conf
cd 3rdparty/srs-bench/tcpproxy/ && go run main.go
tcpdump -i any -w t.pcap tcp port 19350
```
This approach allows for the implementation of packet dumping,
multipoint replication, or the provision of detailed timestamps and byte
information at the proxy. It enables the collection of debugging
information without the need to modify the server.
---------
`TRANS_BY_GPT4`
---------
Co-authored-by: john <hondaxiao@tencent.com>
1. When converting RTC to RTMP, it is necessary to synchronize the audio
and video timestamps. When the synchronization status changes, whether
it is unsynchronized or synchronized, print logs to facilitate
troubleshooting of such issues.
2. Chrome uses the STAP-A packet, which means a single RTP packet
contains SPS/PPS information. OBS WHIP, on the other hand, sends SPS and
PPS in separate RTP packets. Therefore, SPS and PPS are in two
independent RTP packets, and SRS needs to cache these two packets.
---------
Co-authored-by: john <hondaxiao@tencent.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.
---------
Co-authored-by: chundonglinlin <chundonglinlin@163.com>
Co-authored-by: john <hondaxiao@tencent.com>
For WebRTC:
when player before publisher, it will happen track pt didn't change.
- At source change step, change track pt
---------
Co-authored-by: mingche.tsai <w41203208.work@gmail.com>
Co-authored-by: john <hondaxiao@tencent.com>
Accommodate certain complex parameters that include the "=" character,
for example.
`configure --extra-flags="-O2 -D_FORTIFY_SOURCE=2"`
---------
Co-authored-by: john <hondaxiao@tencent.com>
Description
A crash occurs when a forward relay connection has not been established
and an unpublish event is triggered simultaneously. For instance, if DVR
and forward are configured with a specified DVR path that already
exists, initiating a stream will trigger a crash.
Objective
Fix the crash caused by the forward mechanism.
Additional Information
For detailed reproduction steps, please refer to issue #3901.
---------
Co-authored-by: john <hondaxiao@tencent.com>
In an SDK that supports RTC Opus stereo, the parameter "stereo=1" may
appear. SRS (Spatial Reference System) needs to handle this correctly
and return an answer to enable WebRTC stereo support.
---------
`TRANS_BY_GPT4`
Security is the built-in IP whitelist feature of SRS, which allows and
denies certain IP and IP range users. Previously, it only supported
RTMP, but this PR now supports HTTP-FLV, HLS, WebRTC, SRT, and other
protocols.
See https://ossrs.io/lts/en-us/docs/v6/doc/security as example.
---------
Co-authored-by: john <hondaxiao@tencent.com>
The `-` character, when placed in the middle of a regular expression, is
interpreted as a range. It must be placed at the beginning or end to be
interpreted as a literal character.
---------
`TRANS_BY_GPT4`
---------
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.
---------
Co-authored-by: chundonglinlin <chundonglinlin@163.com>
In pure audio mode, there are no keyframes. Therefore, we can only rely
on the length of the slice to determine whether it should be output.
`hls_aof_ratio` is the coefficient that, once reached, will generate a
new slice.
In scenarios with video, if the `hls_aof_ratio` is too small, for
example 1.2, and the GOP (Group of Pictures) is 10 seconds, then a slice
will definitely be generated at 12 seconds. At this point, if there are
no keyframes, it will cause the next slice to start with a non-keyframe.
A safer coefficient is twice the GOP (Group of Pictures). This way, it
won't trigger incorrectly and prevent the individual transcoding of a ts
segment file.
---------
Co-authored-by: Haibo Chen <495810242@qq.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.
---------
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.
---------
Co-authored-by: john <hondaxiao@tencent.com>
By default, caching is enabled during compilation, which means that data
is cached in Docker. This helps to avoid compiling third-party
dependency libraries. However, sometimes when updating third-party
libraries, it's necessary to disable caching to temporarily verify if
the pipeline can succeed. Therefore, a configure option should be added.
When this option is enabled, the compilation cache will not be used, and
all third-party libraries will be compiled from scratch.
---------
Co-authored-by: winlin <winlinvip@gmail.com>
Follow the example in FFmpeg's doc, before calling the API
`avcodec_send_frame`, always use `av_frame_alloc` to create a new frame.
---------
Co-authored-by: Haibo Chen <495810242@qq.com>
SRS supports TCP WebRTC by reading 2 bytes of length, like `read(buf,
2)`. However, in some cases, it might receive 1 byte, causing subsequent
data to be incorrect and making it unable to push or play streams.
---------
Co-authored-by: john <hondaxiao@tencent.com>
Checking the HTTPS API or UDP connectivity for WHIP tests can be
difficult. For example, if the UDP port isn't available but the API is
fine, OBS only says it can't connect to the server. It's hard to see the
HTTPS API response or check if the UDP port is available.
This feature lets you set the ice username and password in SRS. You can
then send a STUN request using nc and see the response, making it easier
to check UDP port connectivity.
1. Use curl to test the WHIP API, including ice-frag and ice-pwd
queries.
2. Use nc to send a STUN binding request to test UDP connectivity.
3. If both the API and UDP are working, you should get a STUN response.
---------
Co-authored-by: john <hondaxiao@tencent.com>
When using Docker, logs are usually printed to console (stdout and
stderr). However, since Docker detection occurs late, after log
initialization, the default log output may be incorrect. In Docker, logs
may still be written to a file instead of the console as expected.
Additionally, the Dockerfile has been improved with a new environment
variable `SRS_IN_DOCKER=on` to clearly indicate a Docker environment. If
automatic Docker detection fails, the configuration will be read, and
this variable will correctly inform SRS that it's in a Docker
environment.
Lastly, the default configuration values have been improved for Docker
environments. By default, `SRS_LOG_TANK=console` and daemon mode is
disabled.
---------
Co-authored-by: john <hondaxiao@tencent.com>
The fix is for the DH_set_length error. As shown in lines 2-5, OpenSSL
3.0 added a check for length, which allowed this issue to be exposed.
```
1 if (dh->params.q == NULL) {
2 /* secret exponent length, must satisfy 2^(l-1) <= p */
3 if (dh->length != 0
4 && dh->length >= BN_num_bits(dh->params.p))
5 goto err;
6 l = dh->length ? dh->length : BN_num_bits(dh->params.p) - 1;
7 if (!BN_priv_rand_ex(priv_key, l, BN_RAND_TOP_ONE,
8 BN_RAND_BOTTOM_ANY, 0, ctx))
9 goto err;
... ...
}
```
---------
Co-authored-by: john <hondaxiao@tencent.com>
1. The comment on the ratio configuration says it can affect the slice
duration, but there is no effect after configuring it.
2. The default hls_td_ratio is 1.5, and after setting it to 1, the
duration is still slightly more than 10 seconds.
3. Even if the GOP is an integer, like 1 second, the slice is still a
non-integer, like 0.998 seconds, which seems a bit unreliable.
4. In the duration of the TS in the m3u8 file, it is one frame less than
the duration of the slice.
5. Set hls_dispose to 120s to dispose HLS files when no stream.
6. Use docker.conf for docker.
Before this patch:
```
#EXTINF:10.983, no desc
livestream-0.ts?hls_ctx=3p095hq0
```
After this patch:
```
#EXTINF:10.000, no desc
livestream-0.ts?hls_ctx=3p095hq0
```
Note: If the fragment is set to 10 seconds, but the GOP size cannot be
divided by 10, such as not 1, 2, 5, or 10, then the duration of ts will
still be more than 10 seconds.
---------
Co-authored-by: john <hondaxiao@tencent.com>
When I compile on OpenHarmony, I encounter an error at the
pthread_setname_np function:
```
./src/app/srs_app_threads.cpp:53:10: error: functions that differ only in their return type cannot be overloaded
void pthread_setname_np(pthread_t trd, const char* name) {
/data/local/ohos-sdk/linux/native/llvm/bin/../../sysroot/usr/include/pthread.h:379:5: note: previous declaration is here
int pthread_setname_np(pthread_t, const char *);
```
Our libc is using musl-libc and has no defined __GLIBC__, so we wanted
to add a judgment that __GLIBC__ already defined.
Please see https://github.com/ossrs/srs/issues/3803 for detail:
1. When using FFmpeg with the `-map 0` option, there may be a 4-byte
timecode in the AMF0 Data.
2. SRS should be able to handle this packet without causing a parsing
error, as it's generally expected to be an AMF0 string, not a 4-byte
timecode.
3. Disregard the timecode since SRS doesn't utilize it.
See [Error submitting a packet to the muxer: Broken pipe, Error muxing a
packet](https://trac.ffmpeg.org/ticket/10565)
---------
Co-authored-by: john <hondaxiao@tencent.com>
When the srs have multiple pps in hevc.the srs can't parse for this.
problem fixed this #3604
---------
Co-authored-by: chundonglinlin <chundonglinlin@163.com>
Co-authored-by: john <hondaxiao@tencent.com>
ISO_IEC_14496-10-AVC-2012.pdf, page 65
7.4.1.1 Encapsulation of an SODB within an RBSP (informative)
... 00 00 03 xx, the 03 byte should be drop where xx represents any 2
bit pattern: 00, 01, 10, or 11.
---------
Co-authored-by: john <hondaxiao@tencent.com>
Co-authored-by: chundonglinlin <chundonglinlin@163.com>
Co-authored-by: winlin <winlin@vip.126.com>
When accessing the SRS Stack, you should log in and use a token for each
request, or utilize the HTTP API with a secret Bearer token included in
every request. The SRS Stack HTTP API proxies both /api/v1 and /rtc/v1
to the SRS HTTP API while ensuring secure authentication. Additionally,
there is a console in the SRS Stack that requires the same token to
request the SRS Stack HTTP API, which is then proxied to the SRS HTTP
API.
The SRS Stack runs SRS with the HTTP API listening at 127.0.0.1:1985 on
the local loopback interface, allowing only the SRS Stack to access it
without authentication. All other users must login and access the SRS
Stack through its interface, rather than directly accessing the SRS HTTP
API within the SRS Stack.
---------
Co-authored-by: panda <542638787@qq.com>
remove unreachable links by python scripts:
```
def is_delete_issue(link):
try:
response = requests.get(link)
except RequestException as e:
print(f"An error occurred while trying to get the link: {e}")
return False
return "This issue has been deleted." in response.text
def remove_unreachable_links(dir):
string_to_search = re.compile(r'// @see https://github\.com/ossrs/srs/issues/.*')
file_list = util.find_files_with_extension(dir, ".cpp", True)
for file in file_list:
lines = []
with open(file, "r", encoding="utf-8") as f:
lines = f.readlines()
with open(file, "w", encoding="utf-8", newline="\n") as f:
for line in lines:
if string_to_search.search(line):
result = re.search(r'https://github\.com/ossrs/srs/issues/\d+', line)
if result:
link = result.group()
if is_delete_issue(link):
print("is_delete_issue link: file: %s, line: %s" % (file, line))
continue
f.write(line)
if __name__ == "__main__":
remove_unreachable_links("srs/trunk/src/")
```
I got an error when cross-compiling the aarch64 platform, the log is as
follows:
`./libavutil/libm.h:54:32: error: static declaration of 'cbrt' follows
non-static declaration`
I see that there are such compilation errors in the
trunk/auto/depends.sh file that have been resolved for the ARM and MIPSE
platforms, and it is recommended to add the ARCH64 platform
## Reload Error Ignore
During a Reload, several stages will be passed through:
1. Parsing new configurations: Parse.
2. Transforming configurations: Transform.
3. Applying configurations: Apply.
Previously, any error at any stage would result in a direct exit, making
the system completely dependent on configuration checks:
```bash
./objs/srs -c conf/srs.conf -t
echo $?
#0
```
Optimized to: If an error occurs before applying the configuration, it
can be ignored. If an error occurs during the application of the
configuration, some of the configuration may have already taken effect,
leading to unpredictable behavior, so SRS will exit directly.
## Reload Fetch API
Added a new HTTP API to query the result of the reload.
```nginx
http_api {
enabled on;
raw_api {
enabled on;
allow_reload on;
}
}
```
```bash
curl http://localhost:1985/api/v1/raw?rpc=reload-fetch
```
```json
{
"code": 0,
"data": {
"err": 0,
"msg": "Success",
"state": 0,
"rid": "0s6y0n9"
}
}
{
"code": 0,
"data": {
"err": 1023,
"msg": "code=1023(ConfigInvalid) : parse file : parse buffer containers/conf/srs.release-local.conf : root parse : parse dir : parse include buffer containers/data/config/srs.vhost.conf : read token, line=0, state=0 : line 3: unexpected end of file, expecting ; or \"}\"",
"state": 1,
"rid": "0g4z471"
}
}
```
This way, you can know if the last reload of the system was successful.
---------
Co-authored-by: Haibo Chen <495810242@qq.com>
SrsContextId object creation can be improved on `srs_protocol_log.cpp`,
No need to create one, then assign it back. It seems a common mistake
for Cpp programmers.
---------
Co-authored-by: john <hondaxiao@tencent.com>
on_hls and hls_dispose are two coroutines, with potential race
conditions. That is, during on_hls, if the API Server being accessed is
slower, it will switch to the hls_dispose coroutine to start cleaning
up. However, when the API Server is processing the slice, a situation
may occur where the slice does not exist, resulting in the following
log:
```
[2023-08-22 12:03:20.309][WARN][40][x5l48q7b][11] ignore task failed code=4005(HttpStatus)(Invalid HTTP status code) : callback on_hls http://localhost:2024/terraform/v1/hooks/srs/hls : http: post http://localhost:2024/terraform/v1/hooks/srs/hls with {"server_id":"vid-5d7dxn8","service_id":"cu153o7g","action":"on_hls","client_id":"x5l48q7b","ip":"172.17.0.1","vhost":"__defaultVhost__","app":"live","tcUrl":"srt://172.17.0.2/live","stream":"stream-44572-2739617660809856576","param":"secret=1ed8e0ffbc53439c8fc8da30ab8c19f0","duration":4.57,"cwd":"/usr/local/srs-stack/platform","file":"./objs/nginx/html/live/stream-44572-2739617660809856576-1.ts","url":"live/stream-44572-2739617660809856576-1.ts","m3u8":"./objs/nginx/html/live/stream-44572-2739617660809856576.m3u8","m3u8_url":"live/stream-44572-2739617660809856576.m3u8","seq_no":1,"stream_url":"/live/stream-44572-2739617660809856576","stream_id":"vid-0n9zoz3"}, status=500, res=invalid ts file ./objs/nginx/html/live/stream-44572-2739617660809856576-1.ts: stat ./objs/nginx/html/live/stream-44572-2739617660809856576-1.ts: no such file or directory
thread [40][x5l48q7b]: call() [./src/app/srs_app_hls.cpp:122][errno=11]
thread [40][x5l48q7b]: on_hls() [./src/app/srs_app_http_hooks.cpp:401][errno=11]
thread [40][x5l48q7b]: do_post() [./src/app/srs_app_http_hooks.cpp:638][errno=11]
[error] 2023/08/22 12:03:20.076984 [52][1001] Serve /terraform/v1/hooks/srs/hls failed, err is stat ./objs/nginx/html/live/stream-44572-2739617660809856576-1.ts: no such file or directory
invalid ts file ./objs/nginx/html/live/stream-44572-2739617660809856576-1.ts
main.handleOnHls.func1.1
/g/platform/srs-hooks.go:684
main.handleOnHls.func1
/g/platform/srs-hooks.go:720
net/http.HandlerFunc.ServeHTTP
/usr/local/go/src/net/http/server.go:2084
net/http.(*ServeMux).ServeHTTP
/usr/local/go/src/net/http/server.go:2462
net/http.serverHandler.ServeHTTP
/usr/local/go/src/net/http/server.go:2916
net/http.(*conn).serve
/usr/local/go/src/net/http/server.go:1966
runtime.goexit
/usr/local/go/src/runtime/asm_amd64.s:1571
```
Similarly, when stopping the stream, on_hls will also be called to
handle the last slice. If the API Server is slower at this time, it will
enter hls_dispose and call unpublish repeatedly. Since the previous
unpublish is still blocked in on_hls, the following interference log
will appear:
```
[2023-08-22 12:03:18.748][INFO][40][6498088c] hls cycle to dispose hls /live/stream-44572-2739617660809856576, timeout=10000000ms
[2023-08-22 12:03:18.752][WARN][40][6498088c][115] flush audio ignored, for segment is not open.
[2023-08-22 12:03:18.752][WARN][40][6498088c][115] ignore the segment close, for segment is not open.
```
Although this log will not cause problems, it can interfere with
judgment.
The solution is to add an 'unpublishing' status. If it is in the
'unpublishing' status, then do not clean up the slices.
---------
Co-authored-by: Haibo Chen <495810242@qq.com>
SRS supports including another configuration in the include package.
When generating configurations, we can only generate the changed
configurations, while the unchanged configurations are in the fixed
files, for example:
```nginx
listen 1935;
include server.conf;
```
In `server.conf`, we can manage the changing configurations with the
program:
```nginx
http_api { enabled on; }
```
However, during system initialization, we often create an empty
`server.conf`, and the content is generated only after the program
starts, so `server.conf` might be an empty file. This also makes it
convenient to use a script to confirm the existence of this file:
```bash
touch server.conf
```
Currently, SRS does not support empty configurations and will report an
error. This PR is to solve this problem, making it more convenient to
use include.
`TRANS_BY_GPT4`
---------
Co-authored-by: Haibo Chen <495810242@qq.com>
When reloading HLS, it directly operates unpublish and publish. At this
time, if HLS is pushed, an exception may occur.
The reason is that these two coroutines operated on the HLS object at
the same time, causing a null pointer.
Solution: Use asynchronous reload. During reload, only set variables and
let the message processing coroutine implement the reload.
---------
Co-authored-by: Haibo Chen <495810242@qq.com>
Co-authored-by: chundonglinlin <chundonglinlin@163.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.
---------
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.
---------
`TRANS_BY_GPT3`
---------
Co-authored-by: chundonglinlin <chundonglinlin@163.com>
Correct the word errors by changing "ShowCoroutines" to "ShowCoroutines".
---------
Co-authored-by: chundonglinlin <chundonglinlin@163.com>
Co-authored-by: john <hondaxiao@tencent.com>
It is advised to eliminate any instances of _srs_rtc_manager that occur
multiple times.
---------
Co-authored-by: john <hondaxiao@tencent.com>
Co-authored-by: chundonglinlin <chundonglinlin@163.com>
1. Ignore SDP GROUP LS.
2. Support ice in global session info.
3. Support audio codec "OPUS" or "opus".
---------
Co-authored-by: Johnny <hellojinqiang@gmail.com>
1. The MTU is effective, with the certificate being split into two DTLS records to comply with the limit.
2. The issue occurs when using BIO_get_mem_data, which retrieves all DTLS packets in a single call, even though each is smaller than the MTU.
3. An alternative callback is available for using BIO_new with BIO_s_mem.
4. Improvements to the MTU setting were made, including adding the DTLS_set_link_mtu function and removing the SSL_set_max_send_fragment function.
5. The handshake process was refined, calling SSL_do_handshake only after ICE completion, and using SSL_read to handle handshake messages.
6. The session close code was improved to enable immediate closure upon receiving an SSL CloseNotify or fatal message.
------
Co-authored-by: chundonglinlin <chundonglinlin@163.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>
* Improve README with AI and add new features
1. Update README file with AI to make it more informative and user-friendly
2. Add a detailed table of contents (TOC) with an introduction for easy navigation
3. Introduce an auto-detecting Automake feature that displays the correct installation command
4. Add support for SRT to HTTP-TS config file
5. Refine the WHIP delete location URL
6. Add support for disabling encryption for WHIP or WHEP
This pull request aims to enhance the quality of the project by introducing innovative features and making the necessary updates. These updates will help users navigate the project more efficiently while also improving the overall project's quality.
---------
Co-authored-by: ChenGH <chengh_math@126.com>
Co-authored-by: john <hondaxiao@tencent.com>
This PR improves the functionality of the HTTP DELETE method used by WHIP to notify the server when the client stops publishing. The URL is parsed from the location header returned by SRS, and the URL is refined with the addition of the action=delete parameter to ensure more accurate identification of the DELETE request.
Furthermore, SRS will disconnect and close the session, enabling the client to publish the stream again quickly and easily. This update eliminates the approximately 30-second waiting period previously required for republishing the stream after an unpublish event.
Overall, this update provides a more effective and efficient method for notifying the server about unpublish events and will enhance the workflow experience for users of the WHIP platform.
-------
Co-authored-by: Haibo Chen <495810242@qq.com>
Co-authored-by: ChenGH <chengh_math@126.com>
* RTMP: Support enhanced RTMP specification for HEVC, v6.0.42.
* Player: Upgrade mpegts.js to support it.
Enhanced RTMP specification: https://github.com/veovera/enhanced-rtmp
First, start SRS `v6.0.42+` with HTTP-TS support:
```bash
./objs/srs -c conf/http.ts.live.conf
```
Then, you can use [OBS 29.1+](https://github.com/obsproject/obs-studio/releases) to push HEVC via RTMP.
Start OBS with the following settings in the `Settings > Stream` tab:
* Server: `rtmp://localhost/live`
* Stream Key: `livestream`
* Encoder: Please select the HEVC hardware encoder.
Finally, open the player http://localhost:8080/players/srs_player.html?stream=livestream.ts
Or use VLS or ffplay to play `http://localhost:8080/live/livestream.ts`
---------
Co-authored-by: chundonglinlin <chundonglinlin@163.com>
In dockerfile, we can set the default RTC candidate to env:
```
ENV SRS_RTC_SERVER_CANDIDATE=\$CANDIDATE
CMD ["./objs/srs", "-e"]
```
When starts a docker container, user can setup the candidate by env:
```
docker run --rm -it --env CANDIDATE=1.2.3.4 ossrs/srs:5
```
We should parse the content of SRS_RTC_SERVER_CANDIDATE as env variable name and parse it again.
---------
Co-authored-by: chundonglinlin <chundonglinlin@163.com>
Co-authored-by: pengfei.ma <pengfei.ma@ctechm.com>
For WebRTC, SRS expect the h.264 codec is:
```
a=rtpmap:106 H264/90000
a=fmtp:106 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42e01f
```
But sometimes, the device does not support the profile, for example only bellow:
```
a=fmtp:123 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42e033
a=fmtp:122 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=420033
a=fmtp:121 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=640033
a=fmtp:120 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=4d0033
```
So we should warning user about the profile missmatch, because it might not work.
----------
Co-authored-by: chundonglinlin <chundonglinlin@163.com>
Co-authored-by: LiPeng <lipeng19811218@gmail.com>
If your OS is not CentOS, Ubuntu, macOS, cygwin64, run of configure will fail with:
```
Your OS Linux is not supported.
```
For other linux systems, we should support an option:
```
./configure --generic-linux=on
```
Please note that you might still fail for other issues while configuring or building.
-------
Co-authored-by: chundonglinlin <chundonglinlin@163.com>
Co-authored-by: ChenGH <chengh_math@126.com>
For some use scenario, the publisher is invited when player want to view the stream:
1. Publisher connect to system, but does not publish any stream to SRS yet.
2. Player connect to system and start to request the stream.
3. System notifies publisher to publish stream to SRS.
4. Player play the stream from SRS.
Please notice that `system` means your business system, not SRS.
This is what we called `on-demand-live-streaming`, so when the last player stop to view the stream, what happends?
1. System needs to notify publisher to stop publish.
2. Or, SRS disconnect the publisher when idle(the last player stops playing).
This PR is for the solution 2, so that the cleanup is very simple, your system does not need to notify publisher to stop publish, because SRS has already disconnected the publihser.
---------
Co-authored-by: winlin <winlin@vip.126.com>
Co-authored-by: chundonglinlin <chundonglinlin@163.com>
For compatibility, transform
rtmp://ip/app...vhost...VHOST/stream
to typical format:
rtmp://ip/app/stream?vhost=VHOST
This is used for some legacy devices, which does not
support standard HTTP url query string.
---------
Co-authored-by: chundonglinlin <chundonglinlin@163.com>
Co-authored-by: john <hondaxiao@tencent.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
* Asan: Disable asan for CentOS and use statically link if possible. v5.0.127 (#3347)
1. Disable asan for all CentOS by default, however user could enable it.
2. Link asan statically if possible.
* Update version to v5.0.127
Co-authored-by: winlin <winlin@vip.126.com>
PICK dd0f398296
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
1. Docker use srs.conf and env variables.
2. Show help if run SRS without any options.
3. Do not guess config file, use whatever from user.
PICK 07a9a005d5
