RTC: For #1657, support http hooks on_play/stop/publish/unpublish

[duiniuluantanqin]

No description provided.

[duiniuluantanqin]

Due to the special nature of WebRTC, signaling and data are two separate channels.
Currently, the request to play/stream is done through the signaling channel, while the handling of stopping is done through the data channel.
Therefore, on_play/publish is placed in the signaling interaction process, while on_stop/unpublish is placed in the data processing flow.

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[codecov-commenter]

Codecov Report

Merging #2509 (6a99ba5) into develop (06f10b1) will decrease coverage by 0.06%.
The diff coverage is 3.38%.

@@             Coverage Diff             @@
##           develop    #2509      +/-   ##
===========================================
- Coverage    42.70%   42.64%   -0.07%     
===========================================
  Files          102      102              
  Lines        36058    36117      +59     
===========================================
+ Hits         15400    15401       +1     
- Misses       20658    20716      +58     

| Impacted Files | Coverage Δ | |'

Translated to English while maintaining the markdown structure:

'| Impacted Files | Coverage Δ | |
|---|---|---|
| trunk/src/app/srs_app_rtc_api.cpp | 0.00% <0.00%> (ø) | |'

Translated to English while maintaining the markdown structure:

| trunk/src/app/srs_app_rtc_api.cpp | 0.00% <0.00%> (ø) | |
| trunk/src/app/srs_app_rtc_conn.cpp | 9.74% <7.40%> (-0.04%) | ⬇️ |
| trunk/src/protocol/srs_service_utility.cpp | 72.43% <0.00%> (-0.55%) | ⬇️ |

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| Δ = absolute <relative> (impact), ø = not affected, ? = missing data |

Translated to English while maintaining the markdown structure:

| Δ = absolute <relative> (impact), ø = not affected, ? = missing data |

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[xiaozhihong]

Record an idea, usually there is a "start streaming" and a "stop streaming", but it seems that WebRTC doesn't pay much attention to this point. The player provided by SRS also does not provide an entry for "stop streaming". In the future, it may be considered to add a status button on the SRS playback page. In the WebRTC scenario, after "start streaming", it will check the streaming status. If the streaming is normal, the button will change to "stop streaming". Users can actively stop the WebRTC streaming through the http API (signaling).

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[winlinvip]

This is the issue that Guanghua mentioned before, regarding the problem of calling HTTP hooks in the destructor function.

publish' and 'play' must be called synchronously because they have authentication functionality.

unpublish' and 'stop' definitely do not require authentication, so it is possible to call them asynchronously.

Refer to ISrsAsyncCallTask.

Calling HTTP in the destructor can lead to coroutine switching, which carries a high risk.

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