RTC: Refine for writing doc. 4.0.92

Author: winlinvip

README.md

@@ -22,7 +22,7 @@ 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.flv) by:
@@ -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:
 
@@ -1258,8 +1257,8 @@ 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:
@@ -1382,6 +1381,8 @@ 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
@@ -1812,7 +1813,6 @@ 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

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 +534,58 @@ 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;
     }
-    # 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;
     }
 }
 

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;

trunk/src/app/srs_app_config.cpp

@@ -3789,8 +3789,7 @@ srs_error_t SrsConfig::check_normal_config()
                 && n != "play" && n != "publish" && n != "cluster"
                 && n != "security" && n != "http_remux" && n != "dash"
                 && n != "http_static" && n != "hds" && n != "exec"
-                && n != "in_ack_size" && n != "out_ack_size" && n != "rtc" && n != "nack"
-                && n != "twcc") {
+                && n != "in_ack_size" && n != "out_ack_size" && n != "rtc") {
                 return srs_error_new(ERROR_SYSTEM_CONFIG_INVALID, "illegal vhost.%s", n.c_str());
             }
             // for each sub directives of vhost.
@@ -5104,7 +5103,7 @@ bool SrsConfig::get_rtc_enabled(string vhost)
 
 bool SrsConfig::get_rtc_bframe_discard(string vhost)
 {
-    static bool DEFAULT = false;
+    static bool DEFAULT = true;
 
     SrsConfDirective* conf = get_rtc(vhost);
 
@@ -5117,7 +5116,7 @@ bool SrsConfig::get_rtc_bframe_discard(string vhost)
         return DEFAULT;
     }
 
-    return conf->arg0() == "discard";
+    return conf->arg0() != "keep";
 }
 
 bool SrsConfig::get_rtc_aac_discard(string vhost)
@@ -5214,7 +5213,7 @@ int SrsConfig::get_rtc_drop_for_pt(string vhost)
 {
     static int DEFAULT = 0;
 
-    SrsConfDirective* conf = get_vhost(vhost);
+    SrsConfDirective* conf = get_rtc(vhost);
     if (!conf) {
         return DEFAULT;
     }
@@ -5231,7 +5230,7 @@ bool SrsConfig::get_rtc_nack_enabled(string vhost)
 {
     static bool DEFAULT = true;
 
-    SrsConfDirective* conf = get_vhost(vhost);
+    SrsConfDirective* conf = get_rtc(vhost);
     if (!conf) {
         return DEFAULT;
     }
@@ -5241,29 +5240,19 @@ bool SrsConfig::get_rtc_nack_enabled(string vhost)
         return DEFAULT;
     }
 
-    conf = conf->get("enabled");
-    if (!conf || conf->arg0().empty()) {
-        return DEFAULT;
-    }
-
     return SRS_CONF_PERFER_TRUE(conf->arg0());
 }
 
 bool SrsConfig::get_rtc_nack_no_copy(string vhost)
 {
     static bool DEFAULT = true;
 
-    SrsConfDirective* conf = get_vhost(vhost);
-    if (!conf) {
-        return DEFAULT;
-    }
-
-    conf = conf->get("nack");
+    SrsConfDirective* conf = get_rtc(vhost);
     if (!conf) {
         return DEFAULT;
     }
 
-    conf = conf->get("no_copy");
+    conf = conf->get("nack_no_copy");
     if (!conf || conf->arg0().empty()) {
         return DEFAULT;
     }
@@ -5275,7 +5264,7 @@ bool SrsConfig::get_rtc_twcc_enabled(string vhost)
 {
     static bool DEFAULT = true;
 
-    SrsConfDirective* conf = get_vhost(vhost);
+    SrsConfDirective* conf = get_rtc(vhost);
     if (!conf) {
         return DEFAULT;
     }
@@ -5285,11 +5274,6 @@ bool SrsConfig::get_rtc_twcc_enabled(string vhost)
         return DEFAULT;
     }
 
-    conf = conf->get("enabled");
-    if (!conf || conf->arg0().empty()) {
-        return DEFAULT;
-    }
-
     return SRS_CONF_PERFER_TRUE(conf->arg0());
 }
 

trunk/src/app/srs_app_http_stream.hpp

@@ -179,6 +179,7 @@ class SrsBufferWriter : public SrsFileWriter
 };
 
 // HTTP Live Streaming, to transmux RTMP to HTTP FLV or other format.
+// TODO: FIXME: Rename to SrsHttpLive
 class SrsLiveStream : public ISrsHttpHandler
 {
 private:

trunk/src/app/srs_app_source.hpp

@@ -500,7 +500,8 @@ class ISrsSourceBridger
     virtual void on_unpublish() = 0;
 };
 
-// live streaming source.
+// The live streaming source.
+// TODO: FIXME: Rename to SrsLiveStream.
 class SrsSource : public ISrsReloadHandler
 {
     friend class SrsOriginHub;

trunk/src/core/srs_core_version4.hpp

@@ -26,6 +26,6 @@
 
 #define VERSION_MAJOR       4
 #define VERSION_MINOR       0
-#define VERSION_REVISION    91
+#define VERSION_REVISION    92
 
 #endif