For #3369 to support an external powerful SIP server, do not use the
embedded SIP server of SRS.
For more information, detailed steps, system architecture, and
background explanation, please see
https://ossrs.net/lts/zh-cn/docs/v6/doc/gb28181#external-sip
---------
Co-authored-by: Jacob Su <suzp1984@gmail.com>
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>
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>
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. 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>
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>
## 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>
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>
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
* FLV: Support set default has_av and disable guessing. v5.0.110
1. Support config default has_audio and has_video.
2. Support disable guessing has_audio or has_video.
* FLV: Reset to false if start to guess has_av.
* FLV: Add regression test for FLV header av metadata.
1. Ignore audo or video packets if FLV header disable it.
2. Run: Add regression test config and run for IDEA.
3. Test: Refine regression test to allow no audio/video for FLV
4. Config: Whether drop packet if not match header.
1. Support Java/log4j log level text.
2. Support configuring by `--log-new-level=on` which is enabled by default.
3. Support `--log-new-level=off` to use SRS 4.0 log level for compatibility.
01. Support GB config as StreamCaster.
02. Support disable GB by --gb28181=off.
03. Add utests for SIP examples.
04. Wireshark plugin to decode TCP/9000 as rtp.rfc4571
05. Support MPEGPS program stream codec.
06. Add utest for PS stream codec.
07. Decode MPEGPS packet stream.
08. Carry RTP and PS packet as helper in PS message.
09. Support recover from error mode.
10. Support process by a pack of PS/TS messages.
11. Add statistic for recovered and msgs dropped.
12. Recover from err position fastly.
13. Define state machine for GB session.
14. Bind context to GB session.
15. Re-invite when media disconnected.
16. Update GitHub actions with GB28181.
17. Support parse CANDIDATE by env or pip.
18. Support mux GB28181 to RTMP.
19. Support regression test by srs-bench.
Haibo Chen
Jacob Su
Winlin
Mr. Li
chundonglinlin
MarkCao
Winlin
mapengfei53
stone
john