WebRTC: Support WHEP for play. v5.0.182 (#3404)

RFC for WHIP: https://datatracker.ietf.org/doc/draft-ietf-wish-whip/

RFC for WHEP: https://datatracker.ietf.org/doc/draft-murillo-whep/

Please note that SRS 5.0 already had WHIP support. I didn't write a
document about WHIP, because WHIP is not a RFC right now, but there are
clues in
[srs-unity](https://github.com/ossrs/srs-unity#usage-publisher). SRS
WHIP url for publisher:
`http://localhost:1985/rtc/v1/whip/?app=live&stream=livestream`

This PR is for WHEP, the url for player is
`http://localhost:1985/rtc/v1/whep/?app=live&stream=livestream`

PS: There is a great PR for OBS to have WHIP support, see
obsproject/obs-studio#7926 and #3581

PS: WHIP for FFmpeg ossrs/ffmpeg-webrtc#1

See #3170

---------

Co-authored-by: Haibo Chen <495810242@qq.com>
Co-authored-by: john <hondaxiao@tencent.com>
Co-authored-by: ChenGH <chengh_math@126.com>

trunk/doc/CHANGELOG.md

@@ -7,6 +7,7 @@ The changelog for SRS.
 <a name="v5-changes"></a>
 
 ## SRS 5.0 Changelog
+* v5.0, 2023-09-21, Merge [#3404](https://github.com/ossrs/srs/pull/3404): WebRTC: Support WHEP for play. v5.0.182 (#3404)
 * v5.0, 2023-09-21, Merge [#3807](https://github.com/ossrs/srs/pull/3807): Prevent the output of srt logs in utest. v5.0.181 (#3807)
 * v5.0, 2023-09-21, Merge [#3696](https://github.com/ossrs/srs/pull/3696): SRT: modify log level from error to debug when no socket to accept. v5.0.180 (#3696)
 * v5.0, 2023-09-18, Merge [#3804](https://github.com/ossrs/srs/pull/3804): Support FFmpeg timecode, fix AMF0 parsing failed. v5.0.179 (#3804)

trunk/research/players/js/srs.page.js

@@ -179,5 +179,5 @@ function srs_init_whip(id, query) {
 }
 function srs_init_whep(id, query) {
     update_nav();
-    $(id).val(build_default_whip_whep_url(query, '/rtc/v1/whip-play/'));
+    $(id).val(build_default_whip_whep_url(query, '/rtc/v1/whep/'));
 }

trunk/research/players/js/srs.sdk.js

@@ -571,7 +571,7 @@ function SrsRtcWhipWhepAsync() {
 
     // See https://datatracker.ietf.org/doc/draft-ietf-wish-whip/
     // @url The WebRTC url to play with, for example:
-    //      http://localhost:1985/rtc/v1/whip-play/?app=live&stream=livestream
+    //      http://localhost:1985/rtc/v1/whep/?app=live&stream=livestream
     self.play = async function(url) {
         if (url.indexOf('/whip-play/') === -1 && url.indexOf('/whep/') === -1) throw new Error(`invalid WHEP url ${url}`);
 

trunk/src/app/srs_app_rtc_api.cpp

@@ -678,7 +678,8 @@ srs_error_t SrsGoApiRtcWhip::do_serve_http(ISrsHttpResponseWriter* w, ISrsHttpMe
     if (action.empty()) {
         action = "publish";
     }
-    if (srs_string_ends_with(r->path(), "/whip-play/")) {
+    // For whip-play or whep, parsed to https://datatracker.ietf.org/doc/draft-murillo-whep/
+    if (srs_string_ends_with(r->path(), "/whip-play/") || srs_string_ends_with(r->path(), "/whep/")) {
         action = "play";
     }
 

trunk/src/app/srs_app_rtc_server.cpp

@@ -471,13 +471,18 @@ srs_error_t SrsRtcServer::listen_api()
     }
 
     // Generally, WHIP is a publishing protocol, but it can be also used as playing.
+    // See https://datatracker.ietf.org/doc/draft-ietf-wish-whep/
     if ((err = http_api_mux->handle("/rtc/v1/whip/", new SrsGoApiRtcWhip(this))) != srs_success) {
         return srs_error_wrap(err, "handle whip");
     }
 
     // We create another mount, to support play with the same query string as publish.
+    // See https://datatracker.ietf.org/doc/draft-murillo-whep/
     if ((err = http_api_mux->handle("/rtc/v1/whip-play/", new SrsGoApiRtcWhip(this))) != srs_success) {
-        return srs_error_wrap(err, "handle whip play");
+        return srs_error_wrap(err, "handle whep play");
+    }
+    if ((err = http_api_mux->handle("/rtc/v1/whep/", new SrsGoApiRtcWhip(this))) != srs_success) {
+        return srs_error_wrap(err, "handle whep play");
     }
 
 #ifdef SRS_SIMULATOR

trunk/src/core/srs_core_version5.hpp

@@ -9,6 +9,6 @@
 
 #define VERSION_MAJOR       5
 #define VERSION_MINOR       0
-#define VERSION_REVISION    181
+#define VERSION_REVISION    182
 
 #endif