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RTC: Call on_play before create session, for it might be freed for timeout. #3455

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merged 3 commits into from
Mar 25, 2023
Merged

RTC: Call on_play before create session, for it might be freed for timeout. #3455

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c7eca95
RTC: fix crash on play
xiaozhihong Mar 7, 2023
95692e2
Merge branch 'develop' into bugfix/issue-3413
winlinvip Mar 25, 2023
db8c961
Update release to v5.0.149, v6.0.37
winlinvip Mar 25, 2023
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2 changes: 2 additions & 0 deletions trunk/doc/CHANGELOG.md
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Original file line number Diff line number Diff line change
Expand Up @@ -8,6 +8,7 @@ The changelog for SRS.

## SRS 6.0 Changelog

* v6.0, 2023-03-25, Merge [#3455](https://github.com/ossrs/srs/pull/3455): RTC: Call on_play before create session, for it might be freed for timeout. v6.0.37 (#3455)
* v6.0, 2023-03-22, Merge [#3427](https://github.com/ossrs/srs/pull/3427): WHIP: Support DELETE resource for Larix Broadcaster. v6.0.36 (#3427)
* v6.0, 2023-03-20, Merge [#3460](https://github.com/ossrs/srs/pull/3460): WebRTC: Support WHIP/WHEP players. v6.0.35 (#3460)
* v6.0, 2023-03-07, Merge [#3441](https://github.com/ossrs/srs/pull/3441): HEVC: webrtc support hevc on safari. v6.0.34 (#3441)
Expand Down Expand Up @@ -50,6 +51,7 @@ The changelog for SRS.

## SRS 5.0 Changelog

* v5.0, 2023-03-25, Merge [#3455](https://github.com/ossrs/srs/pull/3455): RTC: Call on_play before create session, for it might be freed for timeout. v5.0.149 (#3455)
* v5.0, 2023-03-22, Merge [#3427](https://github.com/ossrs/srs/pull/3427): WHIP: Support DELETE resource for Larix Broadcaster. v5.0.148 (#3427)
* v5.0, 2023-03-20, Merge [#3460](https://github.com/ossrs/srs/pull/3460): WebRTC: Support WHIP/WHEP players. v5.0.147 (#3460)
* v5.0, 2023-03-07, Merge [#3446](https://github.com/ossrs/srs/pull/3446): WebRTC: Warning if no ideal profile. v5.0.146 (#3446)
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10 changes: 4 additions & 6 deletions trunk/src/app/srs_app_rtc_api.cpp
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Expand Up @@ -220,18 +220,16 @@ srs_error_t SrsGoApiRtcPlay::serve_http(ISrsHttpResponseWriter* w, ISrsHttpMessa
}
}

if ((err = http_hooks_on_play(ruc->req_)) != srs_success) {
return srs_error_wrap(err, "RTC: http_hooks_on_play");
}

// TODO: FIXME: When server enabled, but vhost disabled, should report error.
// We must do stat the client before hooks, because hooks depends on it.
SrsRtcConnection* session = NULL;
if ((err = server_->create_session(ruc, local_sdp, &session)) != srs_success) {
return srs_error_wrap(err, "create session, dtls=%u, srtp=%u, eip=%s", ruc->dtls_, ruc->srtp_, ruc->eip_.c_str());
}

// We must do hook after stat, because depends on it.
if ((err = http_hooks_on_play(ruc->req_)) != srs_success) {
return srs_error_wrap(err, "RTC: http_hooks_on_play");
}

ostringstream os;
if ((err = local_sdp.encode(os)) != srs_success) {
return srs_error_wrap(err, "encode sdp");
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