Merge branch '4.0release' into merge/develop

trunk/conf/clion-ingest.conf

@@ -25,7 +25,7 @@ rtc_server {
     # The $CANDIDATE means fetch from env, if not configed, use * as default.
     #
     # The * means retrieving server IP automatically, from all network interfaces,
-    # @see https://github.com/ossrs/srs/wiki/v4_CN_RTCWiki#config-candidate
+    # @see https://github.com/ossrs/srs/wiki/v4_CN_WebRTC#config-candidate
     candidate $CANDIDATE;
 }
 

trunk/conf/clion.conf

@@ -25,13 +25,17 @@ rtc_server {
     # The $CANDIDATE means fetch from env, if not configed, use * as default.
     #
     # The * means retrieving server IP automatically, from all network interfaces,
-    # @see https://github.com/ossrs/srs/wiki/v4_CN_RTCWiki#config-candidate
+    # @see https://github.com/ossrs/srs/wiki/v4_CN_WebRTC#config-candidate
     candidate $CANDIDATE;
 }
 
 vhost __defaultVhost__ {
     rtc {
         enabled     on;
+        # @see https://github.com/ossrs/srs/wiki/v4_CN_WebRTC#rtmp-to-rtc
+        rtmp_to_rtc on;
+        # @see https://github.com/ossrs/srs/wiki/v4_CN_WebRTC#rtc-to-rtmp
+        rtc_to_rtmp on;
     }
     http_remux {
         enabled     on;

trunk/conf/docker.conf

@@ -17,6 +17,7 @@ http_server {
 rtc_server {
     enabled on;
     listen 8000;
+    # @see https://github.com/ossrs/srs/wiki/v4_CN_WebRTC#config-candidate
     candidate $CANDIDATE;
 }
 vhost __defaultVhost__ {
@@ -29,5 +30,9 @@ vhost __defaultVhost__ {
     }
     rtc {
         enabled     on;
+        # @see https://github.com/ossrs/srs/wiki/v4_CN_WebRTC#rtmp-to-rtc
+        rtmp_to_rtc off;
+        # @see https://github.com/ossrs/srs/wiki/v4_CN_WebRTC#rtc-to-rtmp
+        rtc_to_rtmp off;
     }
 }

trunk/conf/full.conf

@@ -388,7 +388,7 @@ rtc_server {
     # And by multiple ENV variables:
     #       $CANDIDATE $EIP # TODO: Implements it.
     # @remark For Firefox, the candidate MUST be IP, MUST NOT be DNS name.
-    # @see https://github.com/ossrs/srs/wiki/v4_CN_RTCWiki#config-candidate
+    # @see https://github.com/ossrs/srs/wiki/v4_CN_WebRTC#config-candidate
     # default: *
     candidate *;
     # The IP family filter for auto discover candidate, it can be:
@@ -459,19 +459,16 @@ vhost rtc.vhost.srs.com {
         # default: 0
         drop_for_pt 0;
         ###############################################################
-        # For transmuxing RTMP to RTC, the strategy for bframe.
-        #       keep        Keep bframe, which may make browser with playing problems.
-        #       discard     Discard bframe, maybe cause browser with little problems.
-        # default: discard
-        bframe discard;
-        # For transmuxing RTMP to RTC, the strategy for aac audio.
-        #       transcode   Transcode aac to opus.
-        #       discard     Discard aac audio packet.
-        # default: transcode
-        aac transcode;
+        # Whether enable transmuxing RTMP to RTC.
+        # If enabled, transcode aac to opus.
+        # default: off
+        rtmp_to_rtc off;
+        # Whether keep B-frame, which is normal feature in live streaming,
+        # but usually disabled in RTC.
+        # default: off
+        keep_bframe off;
         ###############################################################
-        # For transmuxing RTC to RTMP.
-        # Whether trans-mux RTC to RTMP streaming.
+        # Whether enable transmuxing RTC to RTMP.
         # Default: off
         rtc_to_rtmp off;
         # The PLI interval in seconds, for RTC to RTMP.
@@ -889,6 +886,7 @@ vhost publish.srs.com {
         # whether parse the sps when publish stream.
         # we can got the resolution of video for stat api.
         # but we may failed to cause publish failed.
+        # @remark If disabled, HLS might never update the sps/pps, it depends on this.
         # default: on
         parse_sps   on;
     }

trunk/conf/https.rtc.conf

@@ -44,7 +44,7 @@ rtc_server {
 vhost __defaultVhost__ {
     rtc {
         enabled     on;
-        bframe      discard;
+        keep_bframe off;
     }
 }
 

trunk/conf/regression-test.conf

@@ -20,20 +20,15 @@ stats {
 }
 rtc_server {
     enabled         on;
-    # Listen at udp://8000
     listen          8000;
-    #
-    # The $CANDIDATE means fetch from env, if not configed, use * as default.
-    #
-    # The * means retrieving server IP automatically, from all network interfaces,
-    # @see https://github.com/ossrs/srs/issues/307#issuecomment-599028124
     candidate       $CANDIDATE;
 }
 
 vhost __defaultVhost__ {
     rtc {
         enabled     on;
-        bframe      discard;
+        rtmp_to_rtc on;
+        keep_bframe off;
         rtc_to_rtmp on;
     }
     play {

trunk/conf/rtc.conf

@@ -21,19 +21,18 @@ stats {
 }
 rtc_server {
     enabled on;
-    # Listen at udp://8000
     listen 8000;
-    #
-    # The $CANDIDATE means fetch from env, if not configed, use * as default.
-    #
-    # The * means retrieving server IP automatically, from all network interfaces,
-    # @see https://github.com/ossrs/srs/wiki/v4_CN_RTCWiki#config-candidate
+    # @see https://github.com/ossrs/srs/wiki/v4_CN_WebRTC#config-candidate
     candidate $CANDIDATE;
 }
 
 vhost __defaultVhost__ {
     rtc {
         enabled     on;
+        # @see https://github.com/ossrs/srs/wiki/v4_CN_WebRTC#rtmp-to-rtc
+        rtmp_to_rtc off;
+        # @see https://github.com/ossrs/srs/wiki/v4_CN_WebRTC#rtc-to-rtmp
+        rtc_to_rtmp off;
     }
     http_remux {
         enabled     on;

trunk/conf/rtc2rtmp.conf

@@ -19,19 +19,17 @@ stats {
 }
 rtc_server {
     enabled on;
-    # Listen at udp://8000
     listen 8000;
-    #
-    # The $CANDIDATE means fetch from env, if not configed, use * as default.
-    #
-    # The * means retrieving server IP automatically, from all network interfaces,
-    # @see https://github.com/ossrs/srs/wiki/v4_CN_RTCWiki#config-candidate
+    # @see https://github.com/ossrs/srs/wiki/v4_CN_WebRTC#config-candidate
     candidate $CANDIDATE;
 }
 
 vhost __defaultVhost__ {
     rtc {
         enabled     on;
+        # @see https://github.com/ossrs/srs/wiki/v4_CN_WebRTC#rtmp-to-rtc
+        rtmp_to_rtc on;
+        # @see https://github.com/ossrs/srs/wiki/v4_CN_WebRTC#rtc-to-rtmp
         rtc_to_rtmp on;
     }
     http_remux {

trunk/conf/rtmp2rtc.conf

@@ -0,0 +1,40 @@
+
+listen              1935;
+max_connections     1000;
+daemon              off;
+srs_log_tank        console;
+
+http_server {
+    enabled         on;
+    listen          8080;
+    dir             ./objs/nginx/html;
+}
+
+http_api {
+    enabled         on;
+    listen          1985;
+}
+stats {
+    network         0;
+}
+rtc_server {
+    enabled on;
+    listen 8000;
+    # @see https://github.com/ossrs/srs/wiki/v4_CN_WebRTC#config-candidate
+    candidate $CANDIDATE;
+}
+
+vhost __defaultVhost__ {
+    rtc {
+        enabled     on;
+        # @see https://github.com/ossrs/srs/wiki/v4_CN_WebRTC#rtmp-to-rtc
+        rtmp_to_rtc on;
+        # @see https://github.com/ossrs/srs/wiki/v4_CN_WebRTC#rtc-to-rtmp
+        rtc_to_rtmp on;
+    }
+    http_remux {
+        enabled     on;
+        mount       [vhost]/[app]/[stream].flv;
+    }
+}
+

trunk/conf/srs.conf

@@ -18,6 +18,7 @@ http_server {
 rtc_server {
     enabled on;
     listen 8000;
+    # @see https://github.com/ossrs/srs/wiki/v4_CN_WebRTC#config-candidate
     candidate $CANDIDATE;
 }
 vhost __defaultVhost__ {
@@ -30,5 +31,9 @@ vhost __defaultVhost__ {
     }
     rtc {
         enabled     on;
+        # @see https://github.com/ossrs/srs/wiki/v4_CN_WebRTC#rtmp-to-rtc
+        rtmp_to_rtc off;
+        # @see https://github.com/ossrs/srs/wiki/v4_CN_WebRTC#rtc-to-rtmp
+        rtc_to_rtmp off;
     }
 }

trunk/conf/srt2rtc.conf

@@ -32,7 +32,7 @@ rtc_server {
     # The $CANDIDATE means fetch from env, if not configed, use * as default.
     #
     # The * means retrieving server IP automatically, from all network interfaces,
-    # @see https://github.com/ossrs/srs/wiki/v4_CN_RTCWiki#config-candidate
+    # @see https://github.com/ossrs/srs/wiki/v4_CN_WebRTC#config-candidate
     candidate $CANDIDATE;
 }
 
@@ -40,6 +40,10 @@ rtc_server {
 vhost __defaultVhost__ {
     rtc {
         enabled     on;
+        # @see https://github.com/ossrs/srs/wiki/v4_CN_WebRTC#rtmp-to-rtc
+        rtmp_to_rtc off;
+        # @see https://github.com/ossrs/srs/wiki/v4_CN_WebRTC#rtc-to-rtmp
+        rtc_to_rtmp off;
     }
     http_remux {
         enabled     on;