SquashSRS4: Support RTC2RTMP.

ossrs · May 1, 2021 · 74bb47c · 74bb47c
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diff --git a/README.md b/README.md
@@ -22,13 +22,13 @@ docker run --rm -p 1935:1935 -p 1985:1985 -p 8080:8080 \
     ossrs/srs:v4.0.85
 ```
 
-> To enable WebRTC, user MUST set the env `CANDIDATE`, see [#307](https://github.com/ossrs/srs/issues/307#issue-76908382).
+> To enable WebRTC, user MUST set the env `CANDIDATE`, see [wiki](https://github.com/ossrs/srs/wiki/v4_CN_WebRTC#config-candidate).
 
 Open [http://localhost:8080/](http://localhost:8080/) to check it, then publish 
-[stream](https://github.com/ossrs/srs/blob/3.0release/trunk/doc/source.200kbps.768x320.flv) by:
+[stream](https://github.com/ossrs/srs/blob/3.0release/trunk/doc/source.flv) by:
 
 ```bash
-docker run --rm --network=host ossrs/srs:encoder ffmpeg -re -i ./doc/source.200kbps.768x320.flv \
+docker run --rm --network=host ossrs/srs:encoder ffmpeg -re -i ./doc/source.flv \
   -c copy -f flv -y rtmp://localhost/live/livestream
 ```
 > Note: If WebRTC enabled, you can publish by [H5](http://localhost:8080/players/rtc_publisher.html?autostart=true).
@@ -67,8 +67,7 @@ Fast index for Wikis:
 * How to deliver HTTP-FLV streaming?([CN][v4_CN_SampleHttpFlv], [EN][v4_EN_SampleHttpFlv])
 * How to deliver HLS streaming?([CN][v4_CN_SampleHLS], [EN][v4_EN_SampleHLS])
 * How to deliver low-latency streaming?([CN][v4_CN_SampleRealtime], [EN][v4_EN_SampleRealtime])
-* Usage: How to play WebRTC from SRS? [#307](https://github.com/ossrs/srs/issues/307)
-* Usage: How to publish WebRTC to SRS? [#307](https://github.com/ossrs/srs/issues/307)
+* How to use WebRTC? ([CN][v4_CN_WebRTC], [EN][v4_EN_WebRTC])
 
 Other important wiki:
 
@@ -127,6 +126,7 @@ Other important wiki:
 - [x] [Experimental] Support mux RTP/RTCP/DTLS/SRTP on one port for WebRTC, [#307][bug #307].
 - [x] [Experimental] Support client address changing for WebRTC, [#307][bug #307].
 - [x] [Experimental] Support transcode RTMP/AAC to WebRTC/Opus, [#307][bug #307].
+- [x] [Experimental] Support AV1 codec for WebRTC, [#2324][bug #2324].
 - [x] [Experimental] Enhance HTTP Stream Server for HTTP-FLV, HTTPS, HLS etc. [#1657][bug #1657].
 - [x] [Experimental] Support push stream by GB28181, [#1500][bug #1500].
 - [x] [Experimental] Support DVR in MP4 format, read [#738][bug #738].
@@ -163,7 +163,10 @@ Other important wiki:
 
 ## V4 changes
 
-* v4.0, 2021-04-29, RTC: Support av1 for Chrome M90. 4.0.91
+* v5.0, 2021-04-20, Support RTC2RTMP bridger and shared FastTimer. 4.0.95
+* v5.0, 2021-04-20, Refine transcoder to support aac2opus and opus2aac. 4.0.94
+* v4.0, 2021-05-01, Timer: Extract and apply shared FastTimer. 4.0.93
+* v4.0, 2021-04-29, RTC: Support AV1 for Chrome M90. 4.0.91
 * v4.0, 2021-04-24, Change push-RTSP as deprecated feature.
 * v4.0, 2021-04-24, Player: Change the default from RTMP to HTTP-FLV.
 * v4.0, 2021-04-24, Disable CherryPy by --cherrypy=off. 4.0.90
@@ -1263,8 +1266,8 @@ Maintainers of SRS project:
 * [Winlin](https://github.com/winlinvip): All areas of streaming server and documents.
 * [Wenjie](https://github.com/wenjiegit): The focus of his work is on the [HDS](https://github.com/simple-rtmp-server/srs/wiki/v4_CN_DeliveryHDS) module.
 * [Runner365](https://github.com/runner365): The focus of his work is on the [SRT](https://github.com/simple-rtmp-server/srs/wiki/v4_CN_SRTWiki) module.
-* [John](https://github.com/xiaozhihong): Focus on [WebRTC](https://github.com/simple-rtmp-server/srs/wiki/v4_CN_RTCWiki) module.
-* [B.P.Y(Bepartofyou)](https://github.com/Bepartofyou): Focus on [WebRTC](https://github.com/simple-rtmp-server/srs/wiki/v4_CN_RTCWiki) module.
+* [John](https://github.com/xiaozhihong): Focus on [WebRTC](https://github.com/simple-rtmp-server/srs/wiki/v4_CN_WebRTC) module.
+* [B.P.Y(Bepartofyou)](https://github.com/Bepartofyou): Focus on [WebRTC](https://github.com/simple-rtmp-server/srs/wiki/v4_CN_WebRTC) module.
 * [Lixin](https://github.com/xialixin): Focus on [GB28181](https://github.com/ossrs/srs/issues/1500) module.
 
 A big THANK YOU goes to:
@@ -1387,6 +1390,8 @@ Winlin
 [v4_EN_SampleForward]: https://github.com/ossrs/srs/wiki/v4_EN_SampleForward
 [v4_CN_SampleRealtime]: https://github.com/ossrs/srs/wiki/v4_CN_SampleRealtime
 [v4_EN_SampleRealtime]: https://github.com/ossrs/srs/wiki/v4_EN_SampleRealtime
+[v4_CN_WebRTC]: https://github.com/ossrs/srs/wiki/v4_CN_WebRTC
+[v4_EN_WebRTC]: https://github.com/ossrs/srs/wiki/v4_EN_WebRTC
 [v4_CN_SampleARM]: https://github.com/ossrs/srs/wiki/v4_CN_SampleARM
 [v4_EN_SampleARM]: https://github.com/ossrs/srs/wiki/v4_EN_SampleARM
 [v4_CN_SampleIngest]: https://github.com/ossrs/srs/wiki/v4_CN_SampleIngest
@@ -1817,7 +1822,6 @@ Winlin
 [bug #1543]: https://github.com/ossrs/srs/issues/1543
 [bug #1509]: https://github.com/ossrs/srs/issues/1509
 [bug #1575]: https://github.com/ossrs/srs/issues/1575
-[bug #307]: https://github.com/ossrs/srs/issues/307
 [bug #1070]: https://github.com/ossrs/srs/issues/1070
 [bug #1580]: https://github.com/ossrs/srs/issues/1580
 [bug #1547]: https://github.com/ossrs/srs/issues/1547
@@ -1872,6 +1876,8 @@ Winlin
 [bug #1998]: https://github.com/ossrs/srs/issues/1998
 [bug #2106]: https://github.com/ossrs/srs/issues/2106
 [bug #2011]: https://github.com/ossrs/srs/issues/2011
+[bug #2324]: https://github.com/ossrs/srs/issues/2324
+[bug #1500]: https://github.com/ossrs/srs/issues/1500
 [bug #zzzzzzzzzzzzz]: https://github.com/ossrs/srs/issues/zzzzzzzzzzzzz
 
 [exo #828]: https://github.com/google/ExoPlayer/pull/828

diff --git a/trunk/conf/full.conf b/trunk/conf/full.conf
@@ -446,52 +446,53 @@ srt_server {
 rtc_server {
     # Whether enable WebRTC server.
     # default: off
-    enabled         on;
+    enabled on;
     # The udp listen port, we will reuse it for connections.
     # default: 8000
-    listen          8000;
+    listen 8000;
     # The exposed candidate IPs, response in SDP candidate line. It can be:
     #       *           Retrieve server IP automatically, from all network interfaces.
     #       eth0        Retrieve server IP by specified network interface name. # TODO: Implements it.
-    #       $CANDIDATE  Read the IP from ENV variable, use * if not set, see https://github.com/ossrs/srs/issues/307#issuecomment-599028124
+    #       $CANDIDATE  Read the IP from ENV variable, use * if not set.
     #       x.x.x.x     A specified IP address or DNS name, which can be access by client such as Chrome.
     # You can specific more than one interface name:
     #       eth0 eth1   Use network interface eth0 and eth1. # TODO: Implements it.
     # Also by IP or DNS names:
     #       192.168.1.3 10.1.2.3 rtc.me # TODO: Implements it.
     # And by multiple ENV variables:
     #       $CANDIDATE $EIP # TODO: Implements it.
-    # @remark For Firefox, the candidate MUST be IP, MUST NOT be DNS name, see also https://github.com/ossrs/srs/pull/1998#issuecomment-726442999
+    # @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
     # default: *
-    candidate       *;
-    # The IP family filter for candidate, it can be:
+    candidate *;
+    # The IP family filter for auto discover candidate, it can be:
     #       ipv4        Filter IP v4 candidates.
     #       ipv6        Filter IP v6 candidates.
     #       all         Filter all IP v4 or v6 candidates.
     # For example, if set to ipv4, we only use the IPv4 address as candidate.
     # default: ipv4
-    ip_family        ipv4;
+    ip_family ipv4;
     # Whether use ECDSA certificate.
     # If not, use RSA certificate.
     # default: on
-    ecdsa           on;
+    ecdsa on;
     # Whether encrypt RTP packet by SRTP.
     # @remark Should always turn it on, or Chrome will fail.
     # default: on
-    encrypt         on;
+    encrypt on;
     # We listen multiple times at the same port, by REUSEPORT, to increase the UDP queue.
     # Note that you can set to 1 and increase the system UDP buffer size by net.core.rmem_max
     # and net.core.rmem_default or just increase this to get larger UDP recv and send buffer.
     # default: 1
-    reuseport       1;
+    reuseport 1;
     # Whether merge multiple NALUs into one.
     # @see https://github.com/ossrs/srs/issues/307#issuecomment-612806318
     # default: off
-    merge_nalus     off;
+    merge_nalus off;
     # Whether enable the perf stat at http://localhost:1985/api/v1/perf
     # TODO: FIXME: We should enable it when refined.
     # default: off
-    perf_stat       off;
+    perf_stat off;
     # For RTP packet and its payload cache.
     rtp_cache {
         # Whether enable the RTP packet cache.
@@ -533,61 +534,67 @@ vhost rtc.vhost.srs.com {
     rtc {
         # Whether enable WebRTC server.
         # default: off
-        enabled     on;
-        # The strategy for bframe.
-        #       keep        Keep bframe, which may make browser with playing problems.
-        #       discard     Discard bframe, maybe cause browser with little problems.
-        # default: keep
-        bframe      discard;
-        # The strategy for aac audio.
-        #       transcode   Transcode aac to opus.
-        #       discard     Discard aac audio packet.
-        # default: transcode
-        aac         transcode;
+        enabled on;
+        # Whether support NACK.
+        # default: on
+        nack on;
+        # Whether directly use the packet, avoid copy.
+        # default: on
+        nack_no_copy on;
+        # Whether support TWCC.
+        # default: on
+        twcc on;
         # The timeout in seconds for session timeout.
         # Client will send ping(STUN binding request) to server, we use it as heartbeat.
         # default: 30
-        stun_timeout    30;
-        # The strick check when process stun.
+        stun_timeout 30;
+        # The strict check when process stun.
         # default: off
         stun_strict_check on;
         # The role of dtls when peer is actpass: passive or active
         # default: passive
-        dtls_role  passive;
+        dtls_role passive;
         # The version of dtls, support dtls1.0, dtls1.2, and auto
         # default: auto
         dtls_version auto;
         # Drop the packet with the pt(payload type), 0 never drop.
         # 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;
+        ###############################################################
+        # For transmuxing RTC to RTMP.
+        # Whether trans-mux RTC to RTMP streaming.
+        # Default: off
+        rtc_to_rtmp off;
+        # The PLI interval in seconds, for RTC to RTMP.
+        # Note the available range is [0.5, 30]
+        # Default: 6.0
+        pli_for_rtmp 6.0;
     }
-    # whether enable min delay mode for vhost.
+    ###############################################################
+    # For transmuxing RTMP to RTC, it will impact the default values if RTC is on.
+    # Whether enable min delay mode for vhost.
     # default: on, for RTC.
-    min_latency     on;
+    min_latency on;
     play {
         # set the MW(merged-write) latency in ms.
         # @remark For WebRTC, we enable pass-timestamp mode, so we ignore this config.
         # default: 0 (For WebRTC)
-        mw_latency      0;
+        mw_latency 0;
         # Set the MW(merged-write) min messages.
-        # default: 0 (For Real-Time, min_latency on)
-        # default: 1 (For WebRTC, min_latency off)
-        mw_msgs         0;
-    }
-    # For NACK.
-    nack {
-        # Whether support NACK.
-        # default: on
-        enabled on;
-        # Whether directly use the packet, avoid copy.
-        # default: on
-        no_copy on;
-    }
-    # For TWCC.
-    twcc {
-        # Whether support TWCC.
-        # default: on
-        enabled on;
+        # default: 0 (For Real-Time, that is min_latency on)
+        # default: 1 (For WebRTC, that is min_latency off)
+        mw_msgs 0;
     }
 }
 

diff --git a/trunk/conf/rtc.conf b/trunk/conf/rtc.conf
@@ -18,21 +18,20 @@ stats {
     network         0;
 }
 rtc_server {
-    enabled         on;
+    enabled on;
     # Listen at udp://8000
-    listen          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;
+    # @see https://github.com/ossrs/srs/wiki/v4_CN_RTCWiki#config-candidate
+    candidate $CANDIDATE;
 }
 
 vhost __defaultVhost__ {
     rtc {
         enabled     on;
-        bframe      discard;
     }
     http_remux {
         enabled     on;

diff --git a/trunk/conf/rtc2rtmp.conf b/trunk/conf/rtc2rtmp.conf
@@ -0,0 +1,42 @@
+
+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 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
+    candidate $CANDIDATE;
+}
+
+vhost __defaultVhost__ {
+    rtc {
+        enabled     on;
+        rtc_to_rtmp on;
+    }
+    http_remux {
+        enabled     on;
+        mount       [vhost]/[app]/[stream].flv;
+    }
+}
+
diff --git a/trunk/conf/rtc_live.conf b/trunk/conf/rtc_live.conf
diff --git a/trunk/doc/source.flv b/trunk/doc/source.flv
@@ -0,0 +1 @@
+source.200kbps.768x320.flv