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>
