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/")
```
* 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>
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. SRT: Fix srt to rtmp crash when sps or pps empty. v5.0.112 (#3323)
2. GB28181: Fix memory overlap for small packets. v5.0.111 (#3315)
3. FLV: Support set default has_av and disable guessing. v5.0.110 (#3311)
4. FLV: Drop packet if header flag is not matched. v5.0.109 (#3306)
5. FLV: Reset has_audio or has_video if only sequence header. (#3310)
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.
* Add utc time utility
* Fix calculate duration in fmp4
* Refine dash code, use segment template timeline
* Shrink m4s file and cleanup
* Support play by dash.js
* Use SegmentTemplate timeline mode with $Number$
Co-authored-by: winlin <winlin@vip.126.com>
* Add utc time utility
* Fix calculate duration in fmp4
* Refine dash code, use segment template timeline
* Shrink m4s file and cleanup
* Support play by dash.js
* Use SegmentTemplate timeline mode with $Number$
Co-authored-by: winlin <winlin@vip.126.com>
1. Parse video codec from PSM packet.
2. Return error and logging if HEVC packet.
3. Ignore invalid AVC NALUs, drop AVC AUD and SEI.
4. Disconnect TCP connection if HEVC.
1. Parse video codec from PSM packet.
2. Return error and logging if HEVC packet.
3. Ignore invalid AVC NALUs, drop AVC AUD and SEI.
4. Disconnect TCP connection if HEVC.
1. Update TS video codec to HEVC during streaming.
2. Return error when HEVC is disabled.
3. Parse HEVC NALU type by SrsHevcNaluTypeParse.
4. Show message when codec change for TS.
Co-authored-by: runner365 <shi.weibd@hotmail.com>
1. Support configure with --h265=on.
2. Parse HEVC(H.265) from FLV or RTMP packet.
3. Support HEVC over RTMP or HTTP-FLV.
Co-authored-by: runner365 <shi.weibd@hotmail.com>
1. Support cygwin by '--cygwin64=on'
2. Detect cygwin automatically.
3. Disalbe sanitizer, srt and srtp with openssl.
4. Disable multiple threads, use single threads.
5. Support utest for cygwin64.
6. Query features for windows by API.
7. Disable stat APIs for cygwin.
8. Use ST select event driver.
Co-authored-by: wenjie.zhao <740936897@qq.com>
* MP4: Fix memory leak when error.
* Kernel: Support free global objects for utest.
* HTTP: Fix memory leak when error.
* MP4: Support more sample rate for audio.
* RTMP: Support free field for utest.
* UTest: Support address sanitizer.
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.
