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RTC: Call on_play before create session, for it might be freed for timeout. v5.0.149, v6.0.37 (#3455)

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Co-authored-by: winlin <winlin@vip.126.com>
Co-authored-by: ChenGH <chengh_math@126.com>
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john 2023-03-25 11:44:48 +08:00 • committed by GitHub
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commit d8755711c1
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2 changed files with 6 additions and 6 deletions
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2
trunk/doc/CHANGELOG.md
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@ -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)
@ -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)

10
trunk/src/app/srs_app_rtc_api.cpp
View file

@ -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");