WebRTC: Support transport over TCP directly, not TURN.

trunk/conf/full.conf

@@ -116,6 +116,7 @@ auto_reload_for_docker on;
 # the rtmp listen ports, split by space, each listen entry is <[ip:]port>
 # for example, 192.168.1.100:1935 10.10.10.100:1935
 # where the ip is optional, default to 0.0.0.0, that is 1935 equals to 0.0.0.0:1935
+# Overwrite by env SRS_LISTEN
 listen              1935;
 # the default chunk size is 128, max is 65536,
 # some client does not support chunk size change,
@@ -135,12 +136,14 @@ chunk_size          60000;
 # where the cli can only be used in shell/terminate.
 http_api {
     # whether http api is enabled.
+    # Overwrite by env SRS_HTTP_API_ENABLED
     # default: off
     enabled         on;
     # The http api listen entry is <[ip:]port>, For example, 192.168.1.100:8080, where the ip is optional, default to
     # 0.0.0.0, that is 8080 equals to 0.0.0.0:8080.
     # Note that you're able to use a dedicated port for HTTP API, such as 1985, to be different with HTTP server. In
     # this situation, you you must also set another HTTPS API port.
+    # Overwrite by env SRS_HTTP_API_LISTEN
     # Default: 1985
     listen          8080;
     # whether enable crossdomain request.
@@ -164,11 +167,13 @@ http_api {
     # For https_api or HTTPS API.
     https {
         # Whether enable HTTPS API.
+        # Overwrite by env SRS_HTTP_API_HTTPS_ENABLED
         # default: off
         enabled on;
         # The listen endpoint for HTTPS API.
         # Note that you're able to use a dedicated port for HTTPS API, such as 1990, and the HTTP API should not be
         # the same of HTTP server(8080) neither.
+        # Overwrite by env SRS_HTTP_API_HTTPS_LISTEN
         # Default: 1990
         listen 8088;
         # The SSL private key file, generated by:
@@ -193,12 +198,14 @@ http_api {
 # need to open the feature http of vhost.
 http_server {
     # whether http streaming service is enabled.
+    # Overwrite by env SRS_HTTP_SERVER_ENABLED
     # default: off
     enabled         on;
     # the http streaming listen entry is <[ip:]port>
     # for example, 192.168.1.100:8080
     # where the ip is optional, default to 0.0.0.0, that is 8080 equals to 0.0.0.0:8080
     # @remark, if use lower port, for instance 80, user must start srs by root.
+    # Overwrite by env SRS_HTTP_SERVER_LISTEN
     # default: 8080
     listen          8080;
     # the default dir for http root.
@@ -211,9 +218,11 @@ http_server {
     # For https_server or HTTPS Streaming.
     https {
         # Whether enable HTTPS Streaming.
+        # Overwrite by env SRS_HTTP_SERVER_HTTTPS_ENABLED
         # default: off
         enabled on;
         # The listen endpoint for HTTPS Streaming.
+        # Overwrite by env SRS_HTTP_SERVER_HTTTPS_LISTEN
         # default: 8088
         listen 8088;
         # The SSL private key file, generated by:
@@ -297,43 +306,72 @@ vhost srt.vhost.srs.com {
 #############################################################################################
 rtc_server {
     # Whether enable WebRTC server.
+    # Overwrite by env SRS_RTC_SERVER_ENABLED
     # default: off
     enabled on;
     # The udp listen port, we will reuse it for connections.
+    # Overwrite by env SRS_RTC_SERVER_LISTEN
     # default: 8000
     listen 8000;
+    # For WebRTC over TCP directly, not TURN, see https://github.com/ossrs/srs/issues/2852
+    # Some network does not support UDP, or not very well, so we use TCP like HTTP/80 port for firewall traversing.
+    tcp {
+        # Whether enable WebRTC over TCP.
+        # Overwrite by env SRS_RTC_SERVER_TCP_ENABLED
+        # Default: off
+        enabled off;
+        # The TCP listen port for WebRTC. Highly recommend is some normally used ports, such as TCP/80, TCP/443,
+        # TCP/8000, TCP/8080 etc. However SRS default to TCP/8000 corresponding to UDP/8000.
+        # Overwrite by env SRS_RTC_SERVER_TCP_LISTEN
+        # Default: 8000
+        listen 8000;
+    }
+    # The protocol for candidate to use, it can be:
+    #       udp         Generate UDP candidates. Note that UDP server is always enabled for WebRTC.
+    #       tcp         Generate TCP candidates. Fail if rtc_server.tcp(WebRTC over TCP) is disabled.
+    #       all         Generate UDP+TCP candidates. Ignore if rtc_server.tcp(WebRTC over TCP) is disabled.
+    # Note that if both are connected, we will use the first connected(DTLS done) one.
+    # Overwrite by env SRS_RTC_SERVER_PROTOCOL
+    # Default: udp
+    protocol udp;
     # The exposed candidate IPs, response in SDP candidate line. It can be:
     #       *           Retrieve server IP automatically, from all network interfaces.
     #       $CANDIDATE  Read the IP from ENV variable, use * if not set.
     #       x.x.x.x     A specified IP address or DNS name, use * if 0.0.0.0.
     # @remark For Firefox, the candidate MUST be IP, MUST NOT be DNS name, see https://bugzilla.mozilla.org/show_bug.cgi?id=1239006
     # @see https://ossrs.net/lts/zh-cn/docs/v4/doc/webrtc#config-candidate
+    # Overwrite by env SRS_RTC_SERVER_CANDIDATE
     # default: *
     candidate *;
     # If candidate is * or 0.0.0.0, means SRS could detect IP automatically, filtered by ip_family.
     # You can config this to off to disable the detecting, then SRS will try to parse the API hostname.
     # Note that browser might fail if no CANDIDATE specified.
+    # Overwrite by env SRS_RTC_SERVER_USE_AUTO_DETECT_NETWORK_IP
     # Default: on
     use_auto_detect_network_ip on;
     # 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.
+    # Overwrite by env SRS_RTC_SERVER_IP_FAMILY
     # default: ipv4
     ip_family ipv4;
     # If api_as_candidates is on, SRS would try to use the IP of api server, specified by srs.sdk.js request:
     #       api:string "http://r.ossrs.net:1985/rtc/v1/play/"
     # in this case, the r.ossrs.net and 39.107.238.185 will be added as candidates.
+    # Overwrite by env SRS_RTC_SERVER_API_AS_CANDIDATES
     # Default: on
     api_as_candidates on;
     # If use api as CANDIDATE, whether resolve the api hostname.
     # Note that use original domain name as CANDIDATE, which might make Firefox failed, see https://bugzilla.mozilla.org/show_bug.cgi?id=1239006
     # Note that if hostname is IPv4 address, always directly use it.
+    # Overwrite by env SRS_RTC_SERVER_RESOLVE_API_DOMAIN
     # Default: on
     resolve_api_domain on;
     # If use api as CANDIDATE, whether keep original api domain name as CANDIDATE.
     # Note that use original domain name as CANDIDATE, which might make Firefox failed, see https://bugzilla.mozilla.org/show_bug.cgi?id=1239006
+    # Overwrite by env SRS_RTC_SERVER_KEEP_API_DOMAIN
     # Default: off
     keep_api_domain off;
     # Whether use ECDSA certificate.
@@ -369,6 +407,7 @@ rtc_server {
 vhost rtc.vhost.srs.com {
     rtc {
         # Whether enable WebRTC server.
+        # Overwrite by env SRS_VHOST_RTC_ENABLED for all vhosts.
         # default: off
         enabled on;
         # Whether support NACK.
@@ -399,6 +438,7 @@ vhost rtc.vhost.srs.com {
         ###############################################################
         # Whether enable transmuxing RTMP to RTC.
         # If enabled, transcode aac to opus.
+        # Overwrite by env SRS_VHOST_RTC_RTMP_TO_RTC for all vhosts.
         # default: off
         rtmp_to_rtc off;
         # Whether keep B-frame, which is normal feature in live streaming,
@@ -407,6 +447,7 @@ vhost rtc.vhost.srs.com {
         keep_bframe off;
         ###############################################################
         # Whether enable transmuxing RTC to RTMP.
+        # Overwrite by env SRS_VHOST_RTC_RTC_TO_RTMP for all vhosts.
         # Default: off
         rtc_to_rtmp off;
         # The PLI interval in seconds, for RTC to RTMP.

trunk/configure

@@ -268,7 +268,7 @@ if [[ $SRS_SRT == YES ]]; then
     MODULE_FILES+=("srs_app_srt_server" "srs_app_srt_listener" "srs_app_srt_conn" "srs_app_srt_utility" "srs_app_srt_source")
 fi
 if [[ $SRS_RTC == YES ]]; then
-    MODULE_FILES+=("srs_app_rtc_conn" "srs_app_rtc_dtls" "srs_app_rtc_sdp"
+    MODULE_FILES+=("srs_app_rtc_conn" "srs_app_rtc_dtls" "srs_app_rtc_sdp" "srs_app_rtc_network"
         "srs_app_rtc_queue" "srs_app_rtc_server" "srs_app_rtc_source" "srs_app_rtc_api")
 fi
 if [[ $SRS_FFMPEG_FIT == YES ]]; then

trunk/doc/CHANGELOG.md

@@ -7,6 +7,7 @@ The changelog for SRS.
 
 ## SRS 5.0 Changelog
 
+* v5.0, 2022-09-04, Fix [#2852](https://github.com/ossrs/srs/issues/2852): WebRTC: WebRTC over TCP directly, not TURN. v5.0.60
 * v5.0, 2022-09-01, Fix [#1405](https://github.com/ossrs/srs/issues/1405): Restore the stream when parsing failed. v5.0.59
 * v5.0, 2022-09-01, Fix [#1405](https://github.com/ossrs/srs/issues/1405): Support guessing IBMF first. v5.0.58
 * v5.0, 2022-09-01, ST: Define and use a new jmpbuf. v5.0.57

trunk/src/app/srs_app_caster_flv.cpp

@@ -290,10 +290,6 @@ srs_error_t SrsDynamicHttpConn::start()
         return srs_error_wrap(err, "set cors=%d", v);
     }
 
-    if ((err = skt->initialize()) != srs_success) {
-        return srs_error_wrap(err, "init socket");
-    }
-
     return conn->start();
 }