.github/workflows/test.yml

@@ -116,6 +116,8 @@ jobs:
       # For coverage
       - name: Run SRS covergae
         if: ${{ startsWith(github.ref, 'refs/heads/') || startsWith(github.ref, 'refs/pull/') }}
+        env:
+          CODECOV_TOKEN: ${{ secrets.CODECOV_TOKEN }}
         run: |
           # The hash of commit.
           SRS_SHA=${{ github.sha }}
@@ -126,15 +128,39 @@ jobs:
           # The github.ref is, for example, refs/pull/2536/merge
           SRS_PR=$(echo ${{ github.ref }}| awk -F 'refs/pull/' '{print $2}'| awk -F '/' '{print $1}')
           echo "For ref=${{ github.ref }}, sha=${{ github.sha }}, SRS_BRANCH=$SRS_BRANCH, SRS_PR=$SRS_PR, SRS_SHA=$SRS_SHA, SRS_PROJECT=$SRS_PROJECT"
-          docker run --rm --env SRS_BRANCH=$SRS_BRANCH --env SRS_PR=$SRS_PR --env SRS_SHA=$SRS_SHA --env SRS_PROJECT=$SRS_PROJECT \
+          docker run --rm --env CODECOV_TOKEN=$CODECOV_TOKEN \
+            --env SRS_BRANCH=$SRS_BRANCH --env SRS_PR=$SRS_PR --env SRS_SHA=$SRS_SHA --env SRS_PROJECT=$SRS_PROJECT \
             srs:cov bash -c 'make utest && ./objs/srs_utest && bash auto/codecov.sh'
 
+  multile-archs:
+    name: actions-test-multile-archs
+    runs-on: ubuntu-20.04
+
+    steps:
+      - name: Checkout repository
+        uses: actions/checkout@v2
+
+      # See https://github.com/crazy-max/ghaction-docker-buildx#moved-to-docker-organization
+      # https://github.com/docker/setup-qemu-action
+      - name: Set up QEMU
+        uses: docker/setup-qemu-action@v1
+      # https://github.com/docker/setup-buildx-action
+      - name: Set up Docker Buildx
+        uses: docker/setup-buildx-action@v1
+
+      - name: Build multiple archs image
+        run: |
+          docker buildx build --platform linux/arm/v7,linux/arm64/v8,linux/amd64 \
+            --output "type=image,push=false" \
+            -f trunk/Dockerfile .
+
   done:
     name: actions-test-done
     needs:
       - build
       - utest
       - coverage
+      - multile-archs
     runs-on: ubuntu-20.04
 
     steps:

README.md

@@ -131,6 +131,12 @@ A big THANK YOU goes to:
 
 ## Releases
 
+* 2023-08-02, Release [v4.0-r5](https://github.com/ossrs/srs/releases/tag/v4.0-r5), v4.0-r5, 4.0 release5, v4.0.271, 145574 lines.
+* 2022-11-22, Release [v4.0-r4](https://github.com/ossrs/srs/releases/tag/v4.0-r4), v4.0-r4, 4.0 release4, v4.0.268, 145482 lines.
+* 2022-09-16, Release [v4.0-r3](https://github.com/ossrs/srs/releases/tag/v4.0-r3), v4.0-r3, 4.0 release3, v4.0.265, 145328 lines.
+* 2022-08-24, Release [v4.0-r2](https://github.com/ossrs/srs/releases/tag/v4.0-r2), v4.0-r2, 4.0 release2, v4.0.257, 144890 lines.
+* 2022-06-29, Release [v4.0-r1](https://github.com/ossrs/srs/releases/tag/v4.0-r1), v4.0-r1, 4.0 release1, v4.0.253, 144680 lines.
+* 2022-06-11, Release [v4.0-r0](https://github.com/ossrs/srs/releases/tag/v4.0-r0), v4.0-r0, 4.0 release0, v4.0.252, 144680 lines.
 * 2022-03-19, Release [v4.0-b10](https://github.com/ossrs/srs/releases/tag/v4.0-b10), v4.0-b10, 4.0 beta10, v4.0.251, 144665 lines.
 * 2022-02-15, Release [v4.0-b9](https://github.com/ossrs/srs/releases/tag/v4.0-b9), v4.0-b9, 4.0 beta9, v4.0.245, 144474 lines.
 * 2022-02-11, Release [v4.0-b8](https://github.com/ossrs/srs/releases/tag/v4.0-b8), v4.0-b8, 4.0 beta8, v4.0.241, 144445 lines.
@@ -433,5 +439,5 @@ Winlin
 [v4_EN_Contact]: https://github.com/ossrs/srs/wiki/v4_EN_Contact
 
 [LICENSE]: https://github.com/ossrs/srs/blob/4.0release/LICENSE
-[LicenseMixing]: https://github.com/ossrs/srs/wiki/LicenseMixing
+[LicenseMixing]: https://ossrs.net/lts/zh-cn/license
 

trunk/3rdparty/README.md

@@ -1,7 +1,7 @@
 http-parser-2.1.zip
 * for srs to support http callback.
 * https://github.com/joyent/http-parser
-* https://github.com/ossrs/srs/wiki/LicenseMixing#http-parser
+* https://ossrs.net/lts/zh-cn/license#http-parser
 
 nginx-1.5.7.zip
 * http://nginx.org/
@@ -10,7 +10,7 @@ nginx-1.5.7.zip
 srt-1-fit
 srt-1.4.1.tar.gz
 * https://github.com/Haivision/srt/releases/tag/v1.4.1
-* https://github.com/ossrs/srs/wiki/LicenseMixing#srt
+* https://ossrs.net/lts/zh-cn/license#srt
 
 openssl-1.1-fit
 openssl-1.1.1b.tar.gz
@@ -21,7 +21,7 @@ openssl-OpenSSL_1_0_2u.tar.gz
 * http://www.openssl.org/source/openssl-1.1.0e.tar.gz
 * openssl for SRS(with-ssl) RTMP complex handshake to delivery h264+aac stream.
 * SRTP depends on openssl 1.0.*, so we use both ssl versions.
-* https://github.com/ossrs/srs/wiki/LicenseMixing#openssl
+* https://ossrs.net/lts/zh-cn/license#openssl
 
 CherryPy-3.2.4.zip
 * sample api server for srs.
@@ -36,7 +36,7 @@ opus-1.3.1.tar.gz
 * http://ffmpeg.org/releases/ffmpeg-4.2.tar.gz
 * https://github.com/xiph/opus/releases/tag/v1.3.1
 * To support RTMP/WebRTC transcoding.
-* https://github.com/ossrs/srs/wiki/LicenseMixing#ffmpeg
+* https://ossrs.net/lts/zh-cn/license#ffmpeg
 
 gtest-fit
 * google test framework.
@@ -51,14 +51,14 @@ st-1.9.zip
 state-threads
 state-threads-1.9.1.tar.gz
 * Patched ST from https://github.com/ossrs/state-threads
-* https://github.com/ossrs/srs/wiki/LicenseMixing#state-threads
+* https://ossrs.net/lts/zh-cn/license#state-threads
 
 JSON
 * https://github.com/udp/json-parser
-* https://github.com/ossrs/srs/wiki/LicenseMixing#json
+* https://ossrs.net/lts/zh-cn/license#json
 
 USRSCTP
-* https://github.com/ossrs/srs/wiki/LicenseMixing#usrsctp
+* https://ossrs.net/lts/zh-cn/license#usrsctp
 
 links:
 * state-threads:

trunk/Dockerfile

@@ -1,28 +1,31 @@
-FROM ossrs/srs:dev AS build
-
-# Install depends tools.
-RUN yum install -y gcc make gcc-c++ patch unzip perl git
+ARG ARCH
+FROM ${ARCH}ossrs/srs:ubuntu20 AS build
 
+ARG BUILDPLATFORM
+ARG TARGETPLATFORM
+ARG JOBS=2
 ARG SRS_AUTO_PACKAGER
+RUN echo "BUILDPLATFORM: $BUILDPLATFORM, TARGETPLATFORM: $TARGETPLATFORM, JOBS: $JOBS, PACKAGER: ${#SRS_AUTO_PACKAGER}"
+
+# https://serverfault.com/questions/949991/how-to-install-tzdata-on-a-ubuntu-docker-image
+ENV DEBIAN_FRONTEND noninteractive
+
+# Install depends tools.
+RUN apt-get update && apt-get install -y gcc make g++ patch unzip perl git
 
 # Build and install SRS.
 COPY . /srs
 WORKDIR /srs/trunk
-RUN ./configure --srt=on --jobs=2 && make -j2 && make install
-
-# All config files for SRS.
-RUN cp -R conf /usr/local/srs/conf && \
-    cp research/api-server/static-dir/index.html /usr/local/srs/objs/nginx/html/ && \
-    cp research/api-server/static-dir/favicon.ico /usr/local/srs/objs/nginx/html/ && \
-    cp research/players/crossdomain.xml /usr/local/srs/objs/nginx/html/ && \
-    cp -R research/console /usr/local/srs/objs/nginx/html/ && \
-    cp -R research/players /usr/local/srs/objs/nginx/html/ && \
-    cp -R 3rdparty/signaling/www/demos /usr/local/srs/objs/nginx/html/
+RUN ./configure --srt=on --jobs=${JOBS} && make -j${JOBS} && make install
 
 ############################################################
 # dist
 ############################################################
-FROM centos:7 AS dist
+FROM ${ARCH}ubuntu:focal AS dist
+
+ARG BUILDPLATFORM
+ARG TARGETPLATFORM
+RUN echo "BUILDPLATFORM: $BUILDPLATFORM, TARGETPLATFORM: $TARGETPLATFORM"
 
 # Expose ports for streaming @see https://github.com/ossrs/srs#ports
 EXPOSE 1935 1985 8080 8000/udp 10080/udp
@@ -32,6 +35,12 @@ COPY --from=build /usr/local/bin/ffmpeg /usr/local/srs/objs/ffmpeg/bin/ffmpeg
 # SRS binary, config files and srs-console.
 COPY --from=build /usr/local/srs /usr/local/srs
 
+# Test the version of binaries.
+RUN ldd /usr/local/srs/objs/ffmpeg/bin/ffmpeg && \
+    /usr/local/srs/objs/ffmpeg/bin/ffmpeg -version && \
+    ldd /usr/local/srs/objs/srs && \
+    /usr/local/srs/objs/srs -v
+
 # Default workdir and command.
 WORKDIR /usr/local/srs
 CMD ["./objs/srs", "-c", "conf/docker.conf"]

trunk/auto/codecov.sh

@@ -36,9 +36,7 @@ fi
 # Note: The right path is like:
 #       https://codecov.io/gh/ossrs/srs/src/3.0release/trunk/src/protocol/srs_rtmp_stack.cpp
 #       https://codecov.io/gh/ossrs/srs/src/20fbb4466fdc8ba5d810b8570df6004063212838/trunk/src/protocol/srs_rtmp_stack.cpp
-# Remark: It takes a few minutes to sync with github, so it might not available when CircleCI is done.
-#       https://circleci.com/gh/ossrs/srs/tree/3.0release
 cd $workdir &&
-export CODECOV_TOKEN="493bba46-c468-4e73-8b45-8cdd8ff62d96" &&
-bash <(curl -s https://codecov.io/bash) $CODECOV_ARGS &&
+export CODECOV_TOKEN="$CODECOV_TOKEN" &&
+bash <(curl -s https://codecov.io/bash) $CODECOV_ARGS -f '!*gtest*' -f '!*c++*' -f '!*ffmpeg-*-fit*' &&
 echo "Done" && exit 0

trunk/auto/depends.sh

@@ -680,11 +680,8 @@ if [[ $SRS_FFMPEG_FIT == YES ]]; then
       FFMPEG_CONFIGURE="env PKG_CONFIG_PATH=$(cd ${SRS_OBJS}/${SRS_PLATFORM} && pwd)/opus/lib/pkgconfig ./configure"
     fi
 
-    # If disable nasm, disable all ASMs.
-    nasm -v >/dev/null 2>&1 && NASM_BIN_OK=YES
-    if [[ $NASM_BIN_OK != YES || $SRS_NASM == NO || $SRS_CROSS_BUILD == YES ]]; then
-        FFMPEG_OPTIONS="--disable-asm --disable-x86asm --disable-inline-asm"
-    fi
+    # Disable all asm for FFmpeg, to compatible with ARM CPU.
+    FFMPEG_OPTIONS="--disable-asm --disable-x86asm --disable-inline-asm"
     # Only build static libraries if no shared FFmpeg.
     if [[ $SRS_SHARED_FFMPEG == YES ]]; then
         FFMPEG_OPTIONS="$FFMPEG_OPTIONS --enable-shared"

trunk/auto/options.sh

@@ -40,13 +40,13 @@ SRS_DEFAULT_CONFIG=conf/srs.conf
 SRS_JOBS=1
 SRS_STATIC=NO
 # If enabled, link shared libraries for libst.so which uses MPL license.
-# See https://github.com/ossrs/srs/wiki/LicenseMixing#state-threads
+# See https://ossrs.net/lts/zh-cn/license#state-threads
 SRS_SHARED_ST=NO
 # If enabled, link shared libraries for libsrt.so which uses MPL license.
-# See https://github.com/ossrs/srs/wiki/LicenseMixing#srt
+# See https://ossrs.net/lts/zh-cn/license#srt
 SRS_SHARED_SRT=NO
 # If enabled, link shared libraries for FFmpeg which is LGPL license.
-# See https://github.com/ossrs/srs/wiki/LicenseMixing#ffmpeg
+# See https://ossrs.net/lts/zh-cn/license#ffmpeg
 SRS_SHARED_FFMPEG=NO
 # whether enable the gcov
 SRS_GCOV=NO

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_WebRTC#config-candidate
+    # @see https://ossrs.net/lts/zh-cn/docs/v4/doc/webrtc#config-candidate
     candidate $CANDIDATE;
 }
 

trunk/conf/clion.conf

@@ -25,16 +25,16 @@ 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_WebRTC#config-candidate
+    # @see https://ossrs.net/lts/zh-cn/docs/v4/doc/webrtc#config-candidate
     candidate $CANDIDATE;
 }
 
 vhost __defaultVhost__ {
     rtc {
         enabled     on;
-        # @see https://github.com/ossrs/srs/wiki/v4_CN_WebRTC#rtmp-to-rtc
+        # @see https://ossrs.net/lts/zh-cn/docs/v4/doc/webrtc#rtmp-to-rtc
         rtmp_to_rtc on;
-        # @see https://github.com/ossrs/srs/wiki/v4_CN_WebRTC#rtc-to-rtmp
+        # @see https://ossrs.net/lts/zh-cn/docs/v4/doc/webrtc#rtc-to-rtmp
         rtc_to_rtmp on;
     }
     http_remux {

trunk/conf/dash.conf

@@ -1,5 +1,5 @@
 # the config for srs to delivery dash
-# @see https://github.com/ossrs/srs/wiki/v1_CN_SampleDASH
+# @see https://ossrs.net/lts/zh-cn/docs/v4/doc/sample-dash
 # @see full.conf for detail config.
 
 listen              1935;

trunk/conf/demo.19350.conf

@@ -1,5 +1,4 @@
 # the config for srs demo
-# @see https://github.com/ossrs/srs/wiki/v1_CN_SampleDemo
 # @see full.conf for detail config.
 
 listen              19350;

trunk/conf/demo.conf

@@ -1,5 +1,4 @@
 # the config for srs demo
-# @see https://github.com/ossrs/srs/wiki/v1_CN_SampleDemo
 # @see full.conf for detail config.
 
 listen              1935;
@@ -58,8 +57,6 @@ vhost demo.srs.com {
     }
     http_hooks {
         enabled         on;
-        on_connect      http://127.0.0.1:8085/api/v1/clients;
-        on_close        http://127.0.0.1:8085/api/v1/clients;
         on_publish      http://127.0.0.1:8085/api/v1/streams;
         on_unpublish    http://127.0.0.1:8085/api/v1/streams;
         on_play         http://127.0.0.1:8085/api/v1/sessions;

trunk/conf/docker.conf

@@ -19,7 +19,7 @@ http_server {
 rtc_server {
     enabled on;
     listen 8000;
-    # @see https://github.com/ossrs/srs/wiki/v4_CN_WebRTC#config-candidate
+    # @see https://ossrs.net/lts/zh-cn/docs/v4/doc/webrtc#config-candidate
     candidate $CANDIDATE;
 }
 vhost __defaultVhost__ {
@@ -32,9 +32,9 @@ vhost __defaultVhost__ {
     }
     rtc {
         enabled     on;
-        # @see https://github.com/ossrs/srs/wiki/v4_CN_WebRTC#rtmp-to-rtc
+        # @see https://ossrs.net/lts/zh-cn/docs/v4/doc/webrtc#rtmp-to-rtc
         rtmp_to_rtc on;
-        # @see https://github.com/ossrs/srs/wiki/v4_CN_WebRTC#rtc-to-rtmp
+        # @see https://ossrs.net/lts/zh-cn/docs/v4/doc/webrtc#rtc-to-rtmp
         rtc_to_rtmp on;
     }
 }

trunk/conf/dvr.mp4.conf

@@ -1,5 +1,5 @@
 # the config for srs to dvr in session mode
-# @see https://github.com/ossrs/srs/wiki/v3_CN_DVR
+# @see https://ossrs.net/lts/zh-cn/docs/v4/doc/dvr
 # @see full.conf for detail config.
 
 listen              1935;

trunk/conf/dvr.path.conf

@@ -1,6 +1,6 @@
 # the config for srs to dvr in custom path.
-# @see https://github.com/ossrs/srs/wiki/v3_CN_DVR#custom-path
-# @see https://github.com/ossrs/srs/wiki/v3_EN_DVR#custom-path
+# @see https://ossrs.net/lts/zh-cn/docs/v4/doc/dvr#custom-path
+# @see https://ossrs.io/lts/en-us/docs/v4/doc/dvr#custom-path
 # @see full.conf for detail config.
 
 listen              1935;

trunk/conf/dvr.segment.conf

@@ -1,5 +1,5 @@
 # the config for srs to dvr in segment mode
-# @see https://github.com/ossrs/srs/wiki/v3_CN_DVR
+# @see https://ossrs.net/lts/zh-cn/docs/v4/doc/dvr
 # @see full.conf for detail config.
 
 listen              1935;

trunk/conf/dvr.session.conf

@@ -1,5 +1,5 @@
 # the config for srs to dvr in session mode
-# @see https://github.com/ossrs/srs/wiki/v3_CN_DVR
+# @see https://ossrs.net/lts/zh-cn/docs/v4/doc/dvr
 # @see full.conf for detail config.
 
 listen              1935;

trunk/conf/edge.conf

@@ -1,5 +1,5 @@
 # the config for srs origin-edge cluster
-# @see https://github.com/ossrs/srs/wiki/v1_CN_Edge
+# @see https://ossrs.net/lts/zh-cn/docs/v4/doc/edge
 # @see full.conf for detail config.
 
 listen              1935;

trunk/conf/edge.token.traverse.conf

@@ -1,5 +1,5 @@
 # the config for srs for token traverse authentication
-# @see https://github.com/ossrs/srs/wiki/v1_CN_DRM
+# @see https://ossrs.net/lts/zh-cn/docs/v4/doc/drm
 # @see full.conf for detail config.
 
 listen              1935

trunk/conf/edge2.conf

@@ -1,5 +1,5 @@
 # the config for srs origin-edge cluster
-# @see https://github.com/ossrs/srs/wiki/v1_CN_Edge
+# @see https://ossrs.net/lts/zh-cn/docs/v4/doc/edge
 # @see full.conf for detail config.
 
 listen              1935;

trunk/conf/exec.conf

@@ -1,5 +1,5 @@
 # the config for srs to support nginx-rtmp exec.
-# @see https://github.com/ossrs/srs/wiki/v3_CN_NgExec
+# @see https://ossrs.net/lts/zh-cn/docs/v4/doc/nginx-exec
 # @see full.conf for detail config.
 
 listen              1935;

trunk/conf/ffmpeg.transcode.conf

@@ -1,5 +1,5 @@
 # the config for srs use ffmpeg to transcode
-# @see https://github.com/ossrs/srs/wiki/v1_CN_SampleFFMPEG
+# @see https://ossrs.net/lts/zh-cn/docs/v4/doc/sample-ffmpeg
 # @see full.conf for detail config.
 
 listen              1935;

trunk/conf/forward.master.conf

@@ -1,5 +1,5 @@
 # the config for srs to forward
-# @see https://github.com/ossrs/srs/wiki/v1_CN_SampleForward
+# @see https://ossrs.net/lts/zh-cn/docs/v4/doc/sample-forward
 # @see full.conf for detail config.
 
 listen              1935;

trunk/conf/forward.slave.conf

@@ -1,5 +1,5 @@
 # the config for srs to forward
-# @see https://github.com/ossrs/srs/wiki/v1_CN_SampleForward
+# @see https://ossrs.net/lts/zh-cn/docs/v4/doc/sample-forward
 # @see full.conf for detail config.
 
 listen              19350;

trunk/conf/full.conf

@@ -1,5 +1,10 @@
 # all config for srs
 
+# The id of server, for stat and api identification.
+# Note that SRS will generate a random id if not configured.
+# Overwrite by env SRS_SERVER_ID
+server_id srs-ie193id;
+
 #############################################################################################
 # RTMP sections
 #############################################################################################
@@ -121,6 +126,12 @@ tcmalloc_release_rate 0.8;
 # Default: on
 query_latest_version on;
 
+# First wait when qlv(query latest version), in seconds.
+# Only available when qlv is enabled.
+# Overwrite by env SRS_FIRST_WAIT_FOR_QLV
+# Default: 300
+first_wait_for_qlv 300;
+
 # For system circuit breaker.
 circuit_breaker {
     # Whether enable the circuit breaker.
@@ -360,8 +371,8 @@ srt_server {
     # The UDP listen port for SRT.
     listen 10080;
     # For detail parameters, please read wiki:
-    #   https://github.com/ossrs/srs/wiki/v4_CN_SRTParams
-    #   https://github.com/ossrs/srs/wiki/v4_EN_SRTParams
+    # @see https://ossrs.net/lts/zh-cn/docs/v4/doc/srt-params
+    # @see https://ossrs.io/lts/en-us/docs/v4/doc/srt-params
     maxbw 1000000000;
     connect_timeout 4000;
     peerlatency 300;
@@ -383,24 +394,30 @@ rtc_server {
     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.
-    #       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 https://github.com/ossrs/srs/wiki/v4_CN_WebRTC#config-candidate
+    #       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
     # default: *
     candidate *;
     # 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.
     # 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.
+    # 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
+    # Default: off
+    keep_api_domain off;
+    # Whether use network interface IP which is detected automatically, filtered by ip_family.
+    # Note that browser might fail if no CANDIDATE specified.
+    # 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.
@@ -599,7 +616,7 @@ vhost cluster.srs.com {
         # it's strongly recommend to open the debug_srs_upnode,
         # when connect to upnode, it will take the debug info,
         # for example, the id, source id, pid.
-        # please see: https://github.com/ossrs/srs/wiki/v1_CN_SrsLog
+        # please see https://ossrs.net/lts/zh-cn/docs/v4/doc/log
         # default: on
         debug_srs_upnode    on;
 
@@ -611,7 +628,7 @@ vhost cluster.srs.com {
 
         # For origin (mode local) cluster, the co-worker's HTTP APIs.
         # This origin will connect to co-workers and communicate with them.
-        # please read: https://github.com/ossrs/srs/wiki/v3_EN_OriginCluster
+        # please see https://ossrs.io/lts/en-us/docs/v4/doc/origin-cluster
         # TODO: FIXME: Support reload.
         coworkers           127.0.0.1:9091 127.0.0.1:9092;
     }
@@ -1027,39 +1044,6 @@ vhost hooks.callback.srs.com {
         # whether the http hooks enable.
         # default off.
         enabled         on;
-        # when client connect to vhost/app, call the hook,
-        # the request in the POST data string is a object encode by json:
-        #       {
-        #           "action": "on_connect",
-        #           "client_id": 1985,
-        #           "ip": "192.168.1.10", "vhost": "video.test.com", "app": "live",
-        #           "tcUrl": "rtmp://video.test.com/live?key=d2fa801d08e3f90ed1e1670e6e52651a",
-        #           "pageUrl": "http://www.test.com/live.html", "server_id": "vid-werty"
-        #       }
-        # if valid, the hook must return HTTP code 200(Status OK) and response
-        # an int value specifies the error code(0 corresponding to success):
-        #       0
-        # support multiple api hooks, format:
-        #       on_connect http://xxx/api0 http://xxx/api1 http://xxx/apiN
-        # @remark For SRS4, the HTTPS url is supported, for example:
-        #       on_connect https://xxx/api0 https://xxx/api1 https://xxx/apiN
-        on_connect      http://127.0.0.1:8085/api/v1/clients http://localhost:8085/api/v1/clients;
-        # when client close/disconnect to vhost/app/stream, call the hook,
-        # the request in the POST data string is a object encode by json:
-        #       {
-        #           "action": "on_close",
-        #           "client_id": 1985,
-        #           "ip": "192.168.1.10", "vhost": "video.test.com", "app": "live",
-        #           "send_bytes": 10240, "recv_bytes": 10240, "server_id": "vid-werty"
-        #       }
-        # if valid, the hook must return HTTP code 200(Status OK) and response
-        # an int value specifies the error code(0 corresponding to success):
-        #       0
-        # support multiple api hooks, format:
-        #       on_close http://xxx/api0 http://xxx/api1 http://xxx/apiN
-        # @remark For SRS4, the HTTPS url is supported, for example:
-        #       on_close https://xxx/api0 https://xxx/api1 https://xxx/apiN
-        on_close        http://127.0.0.1:8085/api/v1/clients http://localhost:8085/api/v1/clients;
         # when client(encoder) publish to vhost/app/stream, call the hook,
         # the request in the POST data string is a object encode by json:
         #       {
@@ -1296,8 +1280,8 @@ vhost hls.srs.com {
         #       [timestamp],replace this const to current UNIX timestamp in ms.
         #       [seq], the sequence number of ts.
         #       [duration], replace this const to current ts duration.
-        # @see https://github.com/ossrs/srs/wiki/v2_CN_DVR#custom-path
-        # @see https://github.com/ossrs/srs/wiki/v2_CN_DeliveryHLS#hls-config
+        # @see https://ossrs.net/lts/zh-cn/docs/v4/doc/dvr#custom-path
+        # @see https://ossrs.net/lts/zh-cn/docs/v4/doc/delivery-hls#hls-config
         # default: [app]/[stream]-[seq].ts
         hls_ts_file     [app]/[stream]-[seq].ts;
         # whether use floor for the hls_ts_file path generation.
@@ -1381,8 +1365,8 @@ vhost hls.srs.com {
 
         # on_hls, never config in here, should config in http_hooks.
         # for the hls http callback, @see http_hooks.on_hls of vhost hooks.callback.srs.com
-        # @read https://github.com/ossrs/srs/wiki/v2_CN_DeliveryHLS#http-callback
-        # @read https://github.com/ossrs/srs/wiki/v2_EN_DeliveryHLS#http-callback
+        # @see https://ossrs.net/lts/zh-cn/docs/v4/doc/delivery-hls#http-callback
+        # @see https://ossrs.io/lts/en-us/docs/v4/doc/delivery-hls#http-callback
 
         # on_hls_notify, never config in here, should config in http_hooks.
         # we support the variables to generate the notify url:
@@ -1391,8 +1375,8 @@ vhost hls.srs.com {
         #       [param], replace with the param.
         #       [ts_url], replace with the ts url.
         # for the hls http callback, @see http_hooks.on_hls_notify of vhost hooks.callback.srs.com
-        # @read https://github.com/ossrs/srs/wiki/v2_CN_DeliveryHLS#on-hls-notify
-        # @read https://github.com/ossrs/srs/wiki/v2_EN_DeliveryHLS#on-hls-notify
+        # @see https://ossrs.net/lts/zh-cn/docs/v4/doc/delivery-hls#on-hls-notify
+        # @see https://ossrs.io/lts/en-us/docs/v4/doc/delivery-hls#on-hls-notify
     }
 }
 # the vhost with hls disabled.
@@ -1481,8 +1465,8 @@ vhost dvr.srs.com {
         #       dvr_path ./objs/nginx/html/[app]/[stream].[timestamp].mp4;
         #       =>
         #       dvr_path ./objs/nginx/html/live/livestream.1420254068776.mp4;
-        # @see https://github.com/ossrs/srs/wiki/v3_CN_DVR#custom-path
-        # @see https://github.com/ossrs/srs/wiki/v3_EN_DVR#custom-path
+        # @see https://ossrs.net/lts/zh-cn/docs/v4/doc/dvr#custom-path
+        # @see https://ossrs.io/lts/en-us/docs/v4/doc/dvr#custom-path
         #       segment,session apply it.
         # default: ./objs/nginx/html/[app]/[stream].[timestamp].flv
         dvr_path        ./objs/nginx/html/[app]/[stream].[timestamp].flv;
@@ -1514,8 +1498,8 @@ vhost dvr.srs.com {
 
         # on_dvr, never config in here, should config in http_hooks.
         # for the dvr http callback, @see http_hooks.on_dvr of vhost hooks.callback.srs.com
-        # @read https://github.com/ossrs/srs/wiki/v2_CN_DVR#http-callback
-        # @read https://github.com/ossrs/srs/wiki/v2_EN_DVR#http-callback
+        # @see https://ossrs.net/lts/zh-cn/docs/v4/doc/dvr#http-callback
+        # @see https://ossrs.io/lts/en-us/docs/v4/doc/dvr#http-callback
     }
 }
 

trunk/conf/go-oryx-edge.conf

@@ -1,5 +1,5 @@
 # the config for srs origin-edge cluster
-# @see https://github.com/ossrs/srs/wiki/v1_CN_Edge
+# @see https://ossrs.net/lts/zh-cn/docs/v4/doc/edge
 # @see full.conf for detail config.
 
 listen              19351;

trunk/conf/go-oryx-edge2.conf

@@ -1,5 +1,5 @@
 # the config for srs origin-edge cluster
-# @see https://github.com/ossrs/srs/wiki/v1_CN_Edge
+# @see https://ossrs.net/lts/zh-cn/docs/v4/doc/edge
 # @see full.conf for detail config.
 
 listen              19352;

trunk/conf/hds.conf

@@ -1,5 +1,4 @@
 # the config for srs to delivery hds
-# @see https://github.com/ossrs/srs/wiki/v1_CN_SampleHDS
 # @see full.conf for detail config.
 
 listen              1935;

trunk/conf/hls.conf

@@ -1,5 +1,5 @@
 # the config for srs to delivery hls
-# @see https://github.com/ossrs/srs/wiki/v1_CN_SampleHLS
+# @see https://ossrs.net/lts/zh-cn/docs/v4/doc/sample-hls
 # @see full.conf for detail config.
 
 listen              1935;

trunk/conf/hls.realtime.conf

@@ -1,5 +1,5 @@
 # the config for srs to delivery realtime RTMP stream
-# @see https://github.com/ossrs/srs/wiki/v2_CN_SampleRealtime
+# @see https://ossrs.net/lts/zh-cn/docs/v4/doc/sample-realtime
 # @see full.conf for detail config.
 
 listen              1935;

trunk/conf/http.aac.live.conf

@@ -1,5 +1,5 @@
 # the config for srs to remux rtmp to aac live stream.
-# @see https://github.com/ossrs/srs/wiki/v2_CN_DeliveryHttpStream
+# @see https://ossrs.net/lts/zh-cn/docs/v4/doc/delivery-http-flv
 # @see full.conf for detail config.
 
 listen              1935;

trunk/conf/http.flv.live.conf

@@ -1,5 +1,5 @@
 # the config for srs to remux rtmp to flv live stream.
-# @see https://github.com/ossrs/srs/wiki/v2_CN_DeliveryHttpStream
+# @see https://ossrs.net/lts/zh-cn/docs/v4/doc/delivery-http-flv
 # @see full.conf for detail config.
 
 listen              1935;

trunk/conf/http.flv.live.edge1.conf

@@ -1,5 +1,5 @@
 # the config for srs to remux rtmp to flv live stream.
-# @see https://github.com/ossrs/srs/wiki/v2_CN_DeliveryHttpStream
+# @see https://ossrs.net/lts/zh-cn/docs/v4/doc/delivery-http-flv
 # @see full.conf for detail config.
 
 listen              19351;

trunk/conf/http.flv.live.edge2.conf

@@ -1,5 +1,5 @@
 # the config for srs to remux rtmp to flv live stream.
-# @see https://github.com/ossrs/srs/wiki/v2_CN_DeliveryHttpStream
+# @see https://ossrs.net/lts/zh-cn/docs/v4/doc/delivery-http-flv
 # @see full.conf for detail config.
 
 listen              19352;

trunk/conf/http.hls.conf

@@ -1,5 +1,5 @@
 # the config for srs to delivery hls
-# @see https://github.com/ossrs/srs/wiki/v1_CN_SampleHLS
+# @see https://ossrs.net/lts/zh-cn/docs/v4/doc/sample-hls
 # @see full.conf for detail config.
 
 listen              1935;

trunk/conf/http.hooks.callback.conf

@@ -8,8 +8,6 @@ srs_log_tank        console;
 vhost __defaultVhost__ {
     http_hooks {
         enabled         on;
-        on_connect      http://127.0.0.1:8085/api/v1/clients http://localhost:8085/api/v1/clients;
-        on_close        http://127.0.0.1:8085/api/v1/clients http://localhost:8085/api/v1/clients;
         on_publish      http://127.0.0.1:8085/api/v1/streams http://localhost:8085/api/v1/streams;
         on_unpublish    http://127.0.0.1:8085/api/v1/streams http://localhost:8085/api/v1/streams;
         on_play         http://127.0.0.1:8085/api/v1/sessions http://localhost:8085/api/v1/sessions;

trunk/conf/http.mp3.live.conf

@@ -1,5 +1,5 @@
 # the config for srs to remux rtmp to mp3 live stream.
-# @see https://github.com/ossrs/srs/wiki/v2_CN_DeliveryHttpStream
+# @see https://ossrs.net/lts/zh-cn/docs/v4/doc/delivery-http-flv
 # @see full.conf for detail config.
 
 listen              1935;

trunk/conf/http.ts.live.conf

@@ -1,5 +1,5 @@
 # the config for srs to remux rtmp to ts live stream.
-# @see https://github.com/ossrs/srs/wiki/v2_CN_DeliveryHttpStream
+# @see https://ossrs.net/lts/zh-cn/docs/v4/doc/delivery-http-flv
 # @see full.conf for detail config.
 
 listen              1935;

trunk/conf/https.docker.conf

@@ -32,7 +32,7 @@ stats {
 rtc_server {
     enabled         on;
     listen 8000; # UDP port
-    # @see https://github.com/ossrs/srs/wiki/v4_CN_WebRTC#config-candidate
+    # @see https://ossrs.net/lts/zh-cn/docs/v4/doc/webrtc#config-candidate
     candidate       $CANDIDATE;
 }
 
@@ -46,9 +46,9 @@ vhost __defaultVhost__ {
     }
     rtc {
         enabled     on;
-        # @see https://github.com/ossrs/srs/wiki/v4_CN_WebRTC#rtmp-to-rtc
+        # @see https://ossrs.net/lts/zh-cn/docs/v4/doc/webrtc#rtmp-to-rtc
         rtmp_to_rtc off;
-        # @see https://github.com/ossrs/srs/wiki/v4_CN_WebRTC#rtc-to-rtmp
+        # @see https://ossrs.net/lts/zh-cn/docs/v4/doc/webrtc#rtc-to-rtmp
         rtc_to_rtmp off;
     }
 }

trunk/conf/https.hooks.callback.conf

@@ -9,8 +9,6 @@ srs_log_tank        console;
 vhost __defaultVhost__ {
     http_hooks {
         enabled         on;
-        on_connect      https://127.0.0.1:443/api/v1/clients;
-        on_close        https://127.0.0.1:443/api/v1/clients;
         on_publish      https://127.0.0.1:443/api/v1/streams;
         on_unpublish    https://127.0.0.1:443/api/v1/streams;
         on_play         https://127.0.0.1:443/api/v1/sessions;

trunk/conf/https.rtc.conf

@@ -32,16 +32,16 @@ stats {
 rtc_server {
     enabled         on;
     listen 8000; # UDP port
-    # @see https://github.com/ossrs/srs/wiki/v4_CN_WebRTC#config-candidate
+    # @see https://ossrs.net/lts/zh-cn/docs/v4/doc/webrtc#config-candidate
     candidate       $CANDIDATE;
 }
 
 vhost __defaultVhost__ {
     rtc {
         enabled     on;
-        # @see https://github.com/ossrs/srs/wiki/v4_CN_WebRTC#rtmp-to-rtc
+        # @see https://ossrs.net/lts/zh-cn/docs/v4/doc/webrtc#rtmp-to-rtc
         rtmp_to_rtc off;
-        # @see https://github.com/ossrs/srs/wiki/v4_CN_WebRTC#rtc-to-rtmp
+        # @see https://ossrs.net/lts/zh-cn/docs/v4/doc/webrtc#rtc-to-rtmp
         rtc_to_rtmp off;
     }
     http_remux {

trunk/conf/https.rtmp2rtc.conf

@@ -32,16 +32,16 @@ stats {
 rtc_server {
     enabled         on;
     listen 8000; # UDP port
-    # @see https://github.com/ossrs/srs/wiki/v4_CN_WebRTC#config-candidate
+    # @see https://ossrs.net/lts/zh-cn/docs/v4/doc/webrtc#config-candidate
     candidate       $CANDIDATE;
 }
 
 vhost __defaultVhost__ {
     rtc {
         enabled     on;
-        # @see https://github.com/ossrs/srs/wiki/v4_CN_WebRTC#rtmp-to-rtc
+        # @see https://ossrs.net/lts/zh-cn/docs/v4/doc/webrtc#rtmp-to-rtc
         rtmp_to_rtc on;
-        # @see https://github.com/ossrs/srs/wiki/v4_CN_WebRTC#rtc-to-rtmp
+        # @see https://ossrs.net/lts/zh-cn/docs/v4/doc/webrtc#rtc-to-rtmp
         rtc_to_rtmp on;
     }
     http_remux {

trunk/conf/https.srs.conf

@@ -32,7 +32,7 @@ stats {
 rtc_server {
     enabled         on;
     listen 8000; # UDP port
-    # @see https://github.com/ossrs/srs/wiki/v4_CN_WebRTC#config-candidate
+    # @see https://ossrs.net/lts/zh-cn/docs/v4/doc/webrtc#config-candidate
     candidate       $CANDIDATE;
 }
 
@@ -46,9 +46,9 @@ vhost __defaultVhost__ {
     }
     rtc {
         enabled     on;
-        # @see https://github.com/ossrs/srs/wiki/v4_CN_WebRTC#rtmp-to-rtc
+        # @see https://ossrs.net/lts/zh-cn/docs/v4/doc/webrtc#rtmp-to-rtc
         rtmp_to_rtc off;
-        # @see https://github.com/ossrs/srs/wiki/v4_CN_WebRTC#rtc-to-rtmp
+        # @see https://ossrs.net/lts/zh-cn/docs/v4/doc/webrtc#rtc-to-rtmp
         rtc_to_rtmp off;
     }
 }

trunk/conf/ingest.conf

@@ -1,5 +1,5 @@
 # use ffmpeg to ingest file/stream/device to SRS
-# @see https://github.com/ossrs/srs/wiki/v1_CN_SampleIngest
+# @see https://ossrs.net/lts/zh-cn/docs/v4/doc/sample-ingest
 # @see full.conf for detail config.
 
 listen              1935;

trunk/conf/ingest.rtsp.conf

@@ -1,5 +1,5 @@
 # use ffmpeg to ingest file/stream/device to SRS
-# @see https://github.com/ossrs/srs/wiki/v1_CN_SampleIngest
+# @see https://ossrs.net/lts/zh-cn/docs/v4/doc/sample-ingest
 # @see full.conf for detail config.
 
 listen              1935;

trunk/conf/origin.cluster.edge.conf

@@ -1,5 +1,5 @@
 # the config for srs origin-edge cluster
-# @see https://github.com/ossrs/srs/wiki/v3_EN_OriginCluster
+# @see https://ossrs.io/lts/en-us/docs/v4/doc/origin-cluster
 # @see full.conf for detail config.
 
 listen              1935;

trunk/conf/origin.cluster.serverA.conf

@@ -1,5 +1,5 @@
 # the config for srs origin-origin cluster
-# @see https://github.com/ossrs/srs/wiki/v3_EN_OriginCluster
+# @see https://ossrs.io/lts/en-us/docs/v4/doc/origin-cluster
 # @see full.conf for detail config.
 
 listen              19350;

trunk/conf/origin.cluster.serverB.conf

@@ -1,5 +1,5 @@
 # the config for srs origin-origin cluster
-# @see https://github.com/ossrs/srs/wiki/v3_EN_OriginCluster
+# @see https://ossrs.io/lts/en-us/docs/v4/doc/origin-cluster
 # @see full.conf for detail config.
 
 listen              19351;

trunk/conf/origin.cluster.serverC.conf

@@ -1,5 +1,5 @@
 # the config for srs origin-origin cluster
-# @see https://github.com/ossrs/srs/wiki/v3_EN_OriginCluster
+# @see https://ossrs.io/lts/en-us/docs/v4/doc/origin-cluster
 # @see full.conf for detail config.
 
 listen              19352;

trunk/conf/origin.conf

@@ -1,5 +1,5 @@
 # the config for srs origin-edge cluster
-# @see https://github.com/ossrs/srs/wiki/v1_CN_Edge
+# @see https://ossrs.net/lts/zh-cn/docs/v4/doc/edge
 # @see full.conf for detail config.
 
 listen              19350;

trunk/conf/push.flv.conf

@@ -1,5 +1,5 @@
 # push HTTP FLV to SRS.
-# @see https://github.com/ossrs/srs/wiki/v2_CN_Streamer#push-http-flv-to-srs
+# @see https://ossrs.net/lts/zh-cn/docs/v4/doc/streamer#push-http-flv-to-srs
 # @see full.conf for detail config.
 
 listen              1935;

trunk/conf/push.mpegts.over.udp.conf

@@ -1,5 +1,5 @@
 # push MPEG-TS over UDP to SRS.
-# @see https://github.com/ossrs/srs/wiki/v2_CN_Streamer#push-mpeg-ts-over-udp
+# @see https://ossrs.net/lts/zh-cn/docs/v4/doc/streamer#push-mpeg-ts-over-udp
 # @see https://github.com/ossrs/srs/issues/250#issuecomment-72321769
 # @see full.conf for detail config.
 

trunk/conf/realtime.conf

@@ -1,5 +1,5 @@
 # the config for srs to delivery realtime RTMP stream
-# @see https://github.com/ossrs/srs/wiki/v2_CN_SampleRealtime
+# @see https://ossrs.net/lts/zh-cn/docs/v4/doc/sample-realtime
 # @see full.conf for detail config.
 
 listen              1935;

trunk/conf/realtime.flv.conf

@@ -1,5 +1,5 @@
 # the config for srs to delivery realtime RTMP stream
-# @see https://github.com/ossrs/srs/wiki/v2_CN_SampleRealtime
+# @see https://ossrs.net/lts/zh-cn/docs/v4/doc/sample-realtime
 # @see full.conf for detail config.
 
 listen              1935;

trunk/conf/rtc.conf

@@ -22,16 +22,16 @@ stats {
 rtc_server {
     enabled on;
     listen 8000; # UDP port
-    # @see https://github.com/ossrs/srs/wiki/v4_CN_WebRTC#config-candidate
+    # @see https://ossrs.net/lts/zh-cn/docs/v4/doc/webrtc#config-candidate
     candidate $CANDIDATE;
 }
 
 vhost __defaultVhost__ {
     rtc {
         enabled     on;
-        # @see https://github.com/ossrs/srs/wiki/v4_CN_WebRTC#rtmp-to-rtc
+        # @see https://ossrs.net/lts/zh-cn/docs/v4/doc/webrtc#rtmp-to-rtc
         rtmp_to_rtc off;
-        # @see https://github.com/ossrs/srs/wiki/v4_CN_WebRTC#rtc-to-rtmp
+        # @see https://ossrs.net/lts/zh-cn/docs/v4/doc/webrtc#rtc-to-rtmp
         rtc_to_rtmp off;
     }
     http_remux {

trunk/conf/rtc2rtmp.conf

@@ -20,16 +20,16 @@ stats {
 rtc_server {
     enabled on;
     listen 8000; # UDP port
-    # @see https://github.com/ossrs/srs/wiki/v4_CN_WebRTC#config-candidate
+    # @see https://ossrs.net/lts/zh-cn/docs/v4/doc/webrtc#config-candidate
     candidate $CANDIDATE;
 }
 
 vhost __defaultVhost__ {
     rtc {
         enabled     on;
-        # @see https://github.com/ossrs/srs/wiki/v4_CN_WebRTC#rtmp-to-rtc
+        # @see https://ossrs.net/lts/zh-cn/docs/v4/doc/webrtc#rtmp-to-rtc
         rtmp_to_rtc on;
-        # @see https://github.com/ossrs/srs/wiki/v4_CN_WebRTC#rtc-to-rtmp
+        # @see https://ossrs.net/lts/zh-cn/docs/v4/doc/webrtc#rtc-to-rtmp
         rtc_to_rtmp on;
     }
     http_remux {

trunk/conf/rtmp.conf

@@ -1,5 +1,5 @@
 # the config for srs to delivery RTMP
-# @see https://github.com/ossrs/srs/wiki/v1_CN_SampleRTMP
+# @see https://ossrs.net/lts/zh-cn/docs/v4/doc/sample-rtmp
 # @see full.conf for detail config.
 
 listen              1935;

trunk/conf/rtmp2rtc.conf

@@ -20,16 +20,16 @@ stats {
 rtc_server {
     enabled on;
     listen 8000; # UDP port
-    # @see https://github.com/ossrs/srs/wiki/v4_CN_WebRTC#config-candidate
+    # @see https://ossrs.net/lts/zh-cn/docs/v4/doc/webrtc#config-candidate
     candidate $CANDIDATE;
 }
 
 vhost __defaultVhost__ {
     rtc {
         enabled     on;
-        # @see https://github.com/ossrs/srs/wiki/v4_CN_WebRTC#rtmp-to-rtc
+        # @see https://ossrs.net/lts/zh-cn/docs/v4/doc/webrtc#rtmp-to-rtc
         rtmp_to_rtc on;
-        # @see https://github.com/ossrs/srs/wiki/v4_CN_WebRTC#rtc-to-rtmp
+        # @see https://ossrs.net/lts/zh-cn/docs/v4/doc/webrtc#rtc-to-rtmp
         rtc_to_rtmp on;
     }
     http_remux {

trunk/conf/srs.conf

@@ -18,7 +18,7 @@ http_server {
 rtc_server {
     enabled on;
     listen 8000; # UDP port
-    # @see https://github.com/ossrs/srs/wiki/v4_CN_WebRTC#config-candidate
+    # @see https://ossrs.net/lts/zh-cn/docs/v4/doc/webrtc#config-candidate
     candidate $CANDIDATE;
 }
 vhost __defaultVhost__ {
@@ -31,9 +31,9 @@ vhost __defaultVhost__ {
     }
     rtc {
         enabled     on;
-        # @see https://github.com/ossrs/srs/wiki/v4_CN_WebRTC#rtmp-to-rtc
+        # @see https://ossrs.net/lts/zh-cn/docs/v4/doc/webrtc#rtmp-to-rtc
         rtmp_to_rtc off;
-        # @see https://github.com/ossrs/srs/wiki/v4_CN_WebRTC#rtc-to-rtmp
+        # @see https://ossrs.net/lts/zh-cn/docs/v4/doc/webrtc#rtc-to-rtmp
         rtc_to_rtmp off;
     }
 }

trunk/conf/srt2rtc.conf

@@ -32,17 +32,17 @@ 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_WebRTC#config-candidate
+    # @see https://ossrs.net/lts/zh-cn/docs/v4/doc/webrtc#config-candidate
     candidate $CANDIDATE;
 }
 
 # @doc https://github.com/ossrs/srs/issues/1147#issuecomment-577607026
 vhost __defaultVhost__ {
     rtc {
         enabled     on;
-        # @see https://github.com/ossrs/srs/wiki/v4_CN_WebRTC#rtmp-to-rtc
+        # @see https://ossrs.net/lts/zh-cn/docs/v4/doc/webrtc#rtmp-to-rtc
         rtmp_to_rtc off;
-        # @see https://github.com/ossrs/srs/wiki/v4_CN_WebRTC#rtc-to-rtmp
+        # @see https://ossrs.net/lts/zh-cn/docs/v4/doc/webrtc#rtc-to-rtmp
         rtc_to_rtmp off;
     }
     http_remux {

trunk/conf/transcode2hls.audio.only.conf

@@ -1,5 +1,5 @@
 # the config for srs to delivery hls
-# @see https://github.com/ossrs/srs/wiki/v1_CN_SampleHLS
+# @see https://ossrs.net/lts/zh-cn/docs/v4/doc/sample-hls
 # @see full.conf for detail config.
 
 listen              1935;

trunk/conf/transform.edge.conf

@@ -1,5 +1,5 @@
 # the config for srs origin-edge cluster
-# @see https://github.com/ossrs/srs/wiki/v2_CN_Edge#transform-vhost
+# @see https://ossrs.net/lts/zh-cn/docs/v4/doc/edge#transform-vhost
 # @see full.conf for detail config.
 
 listen              1935;

trunk/conf/vm.conf

@@ -25,7 +25,7 @@ stats {
 rtc_server {
     enabled on;
     listen 8000; # UDP port
-    # @see https://github.com/ossrs/srs/wiki/v4_CN_WebRTC#config-candidate
+    # @see https://ossrs.net/lts/zh-cn/docs/v4/doc/webrtc#config-candidate
     candidate $CANDIDATE;
 }
 
@@ -39,9 +39,9 @@ vhost __defaultVhost__ {
     }
     rtc {
         enabled     on;
-        # @see https://github.com/ossrs/srs/wiki/v4_CN_WebRTC#rtmp-to-rtc
+        # @see https://ossrs.net/lts/zh-cn/docs/v4/doc/webrtc#rtmp-to-rtc
         rtmp_to_rtc on;
-        # @see https://github.com/ossrs/srs/wiki/v4_CN_WebRTC#rtc-to-rtmp
+        # @see https://ossrs.net/lts/zh-cn/docs/v4/doc/webrtc#rtc-to-rtmp
         rtc_to_rtmp on;
     }
 }

trunk/doc/Architecture.md

@@ -293,7 +293,7 @@ Supported operating systems and hardware:
 [v4_EN_Contact]: https://github.com/ossrs/srs/wiki/v4_EN_Contact
 
 [LICENSE]: https://github.com/ossrs/srs/blob/4.0release/LICENSE
-[LicenseMixing]: https://github.com/ossrs/srs/wiki/LicenseMixing
+[LicenseMixing]: https://ossrs.net/lts/zh-cn/license
 
 [release2]: https://github.com/ossrs/srs/wiki/v4_CN_Product#release20
 [release3]: https://github.com/ossrs/srs/wiki/v4_CN_Product#release30

trunk/doc/CHANGELOG.md

@@ -8,10 +8,31 @@ The changelog for SRS.
 
 ## SRS 4.0 Changelog
 
+* v4.0, 2023-08-02, Fix [#3749](https://github.com/ossrs/srs/issues/3749): HLS: Ignore empty NALU to avoid error. v4.0.271
+* v4.0, 2023-07-21, Merge [#3695](https://github.com/ossrs/srs/pull/3695): API: Fix HTTPS callback issue using SNI in TLS client handshake. v4.0.270 (#3695)
+* v4.0, 2022-12-24, For [#296](https://github.com/ossrs/srs/issues/296): MP3: Fix bug for TS or HLS with mp3 codec. v4.0.269
+* v4.0, 2022-11-22, Pick [#3079](https://github.com/ossrs/srs/issues/3079): WebRTC: Fix no audio and video issue for Firefox. v4.0.268
+* v4.0, 2022-10-10, For [#2901](https://github.com/ossrs/srs/issues/2901): Edge: Fast disconnect and reconnect. v4.0.267
+* v4.0, 2022-09-27, For [#3167](https://github.com/ossrs/srs/issues/3167): WebRTC: Refine sequence jitter algorithm. v4.0.266
+* v4.0, 2022-09-16, For [#3179](https://github.com/ossrs/srs/issues/3179): WebRTC: Make sure the same m-lines order for offer and answer. v4.0.265
+* v4.0, 2022-09-09, For [#3174](https://github.com/ossrs/srs/issues/3174): WebRTC: Support Unity to publish or play stream. v4.0.264
+* v4.0, 2022-09-09, Fix [#3093](https://github.com/ossrs/srs/issues/3093): WebRTC: Ignore unknown fmtp for h.264. v4.0.263
+* v4.0, 2022-09-06, Fix [#3170](https://github.com/ossrs/srs/issues/3170): WebRTC: Support WHIP(WebRTC-HTTP ingestion protocol). v4.0.262
+* v4.0, 2022-09-03, Fix HTTP url parsing bug. v4.0.261
+* v4.0, 2022-09-03, For [#3167](https://github.com/ossrs/srs/issues/3167): WebRTC: Play stucked when republish. v4.0.260
+* v4.0, 2022-09-02, For [#307](https://github.com/ossrs/srs/issues/307): WebRTC: Support use domain name as CANDIDATE. v4.0.259
+* v4.0, 2022-08-29, Copy libxml2-dev for FFmpeg. v4.0.258
+* v4.0, 2022-08-24, STAT: Support config server_id and generate one if empty. v4.0.257
+* v4.0, 2022-08-24, For [#2136](https://github.com/ossrs/srs/issues/2136): API: Cleanup no active streams for statistics. v4.0.256
+* v4.0, 2022-08-17, RTMP URL supports domain in stream parameters. v4.0.255
+* v4.0, 2022-08-10, Fix server id generator bug. v4.0.254
+* v4.0, 2022-06-29, Update SRS image for r.ossrs.net. v4.0.253
+* v4.0, 2022-06-11, For [#3058](https://github.com/ossrs/srs/pull/3058): Docker: Support x86_64, armv7 and aarch64 docker image (#3058). v4.0.252
 * v4.0, 2022-03-19, For [#2893](https://github.com/ossrs/srs/pull/2893): SRT: Decouple publish with play url (#2893). v4.0.251
 * v4.0, 2022-03-19, Merge [#2908](https://github.com/ossrs/srs/pull/2908): SRT: url supports multiple QueryStrings (#2908). v4.0.250
 * v4.0, 2022-03-17, SRT: Support debug and run with CLion. v4.0.249
 * v4.0, 2022-03-15, Merge [#2966](https://github.com/ossrs/srs/pull/2966): Bugfix: Fix rtcp nack blp encode bug (#2966). v4.0.248
+* v4.0, 2022-03-11, Merge [#2914](https://github.com/ossrs/srs/pull/2914): Security: Enable CIDR for allow/deny play/publish.
 * v4.0, 2022-03-07, RTC: Identify the WebRTC publisher in param for hooks. v4.0.247
 * v4.0, 2022-03-07, SRT: Append vhost to stream, not app. v4.0.246
 * v4.0, 2022-02-15, Fix warnings for uuid. v4.0.245

trunk/doc/Features.md

@@ -304,7 +304,7 @@ The features of SRS.
 [v4_EN_Contact]: https://github.com/ossrs/srs/wiki/v4_EN_Contact
 
 [LICENSE]: https://github.com/ossrs/srs/blob/4.0release/LICENSE
-[LicenseMixing]: https://github.com/ossrs/srs/wiki/LicenseMixing
+[LicenseMixing]: https://ossrs.net/lts/zh-cn/license
 
 [release2]: https://github.com/ossrs/srs/wiki/v4_CN_Product#release20
 [release3]: https://github.com/ossrs/srs/wiki/v4_CN_Product#release30

trunk/doc/PERFORMANCE.md

@@ -408,7 +408,7 @@ Winlin 2021
 [v4_EN_Contact]: https://github.com/ossrs/srs/wiki/v4_EN_Contact
 
 [LICENSE]: https://github.com/ossrs/srs/blob/4.0release/LICENSE
-[LicenseMixing]: https://github.com/ossrs/srs/wiki/LicenseMixing
+[LicenseMixing]: https://ossrs.net/lts/zh-cn/license
 
 [release2]: https://github.com/ossrs/srs/wiki/v4_CN_Product#release20
 [release3]: https://github.com/ossrs/srs/wiki/v4_CN_Product#release30

trunk/doc/Resources.md

@@ -318,7 +318,7 @@ git clone https://github.com/ossrs/srs.git
 [v4_EN_Contact]: https://github.com/ossrs/srs/wiki/v4_EN_Contact
 
 [LICENSE]: https://github.com/ossrs/srs/blob/4.0release/LICENSE
-[LicenseMixing]: https://github.com/ossrs/srs/wiki/LicenseMixing
+[LicenseMixing]: https://ossrs.net/lts/zh-cn/license
 
 [release2]: https://github.com/ossrs/srs/wiki/v4_CN_Product#release20
 [release3]: https://github.com/ossrs/srs/wiki/v4_CN_Product#release30

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

@@ -54,6 +54,8 @@ function SrsRtcPublisherAsync() {
         var conf = self.__internal.prepareUrl(url);
         self.pc.addTransceiver("audio", {direction: "sendonly"});
         self.pc.addTransceiver("video", {direction: "sendonly"});
+        //self.pc.addTransceiver("video", {direction: "sendonly"});
+        //self.pc.addTransceiver("audio", {direction: "sendonly"});
 
         if (!navigator.mediaDevices && window.location.protocol === 'http:' && window.location.hostname !== 'localhost') {
             throw new SrsError('HttpsRequiredError', `Please use HTTPS or localhost to publish, read https://github.com/ossrs/srs/issues/2762#issuecomment-983147576`);
@@ -303,6 +305,8 @@ function SrsRtcPlayerAsync() {
         var conf = self.__internal.prepareUrl(url);
         self.pc.addTransceiver("audio", {direction: "recvonly"});
         self.pc.addTransceiver("video", {direction: "recvonly"});
+        //self.pc.addTransceiver("video", {direction: "recvonly"});
+        //self.pc.addTransceiver("audio", {direction: "recvonly"});
 
         var offer = await self.pc.createOffer();
         await self.pc.setLocalDescription(offer);

trunk/src/app/srs_app_config.cpp

@@ -58,6 +58,9 @@ const char* _srs_version = "XCORE-" RTMP_SIG_SRS_SERVER;
 // '\r'
 #define SRS_CR (char)SRS_CONSTS_CR
 
+// Overwrite the config by env.
+#define SRS_OVERWRITE_BY_ENV_SECONDS(key) if (getenv(key)) return ::atoi(getenv(key)) * SRS_UTIME_SECONDS
+
 /**
  * dumps the ingest/transcode-engine in @param dir to amf0 object @param engine.
  * @param dir the transcode or ingest config directive.
@@ -2466,11 +2469,11 @@ srs_error_t SrsConfig::check_normal_config()
             && n != "max_connections" && n != "daemon" && n != "heartbeat"
             && n != "http_api" && n != "stats" && n != "vhost" && n != "pithy_print_ms"
             && n != "http_server" && n != "stream_caster" && n != "rtc_server" && n != "srt_server"
-            && n != "utc_time" && n != "work_dir" && n != "asprocess"
+            && n != "utc_time" && n != "work_dir" && n != "asprocess" && n != "server_id"
             && n != "ff_log_level" && n != "grace_final_wait" && n != "force_grace_quit"
             && n != "grace_start_wait" && n != "empty_ip_ok" && n != "disable_daemon_for_docker"
             && n != "inotify_auto_reload" && n != "auto_reload_for_docker" && n != "tcmalloc_release_rate"
-            && n != "query_latest_version"
+            && n != "query_latest_version" && n != "first_wait_for_qlv"
             && n != "circuit_breaker" && n != "is_full" && n != "in_docker"
             ) {
             return srs_error_new(ERROR_SYSTEM_CONFIG_INVALID, "illegal directive %s", n.c_str());
@@ -2542,7 +2545,8 @@ srs_error_t SrsConfig::check_normal_config()
             string n = conf->at(i)->name;
             if (n != "enabled" && n != "listen" && n != "dir" && n != "candidate" && n != "ecdsa"
                 && n != "encrypt" && n != "reuseport" && n != "merge_nalus" && n != "black_hole"
-                && n != "ip_family" && n != "api_as_candidates") {
+                && n != "ip_family" && n != "api_as_candidates" && n != "resolve_api_domain"
+                && n != "keep_api_domain" && n != "use_auto_detect_network_ip") {
                 return srs_error_new(ERROR_SYSTEM_CONFIG_INVALID, "illegal rtc_server.%s", n.c_str());
             }
         }
@@ -3028,6 +3032,90 @@ SrsConfDirective* SrsConfig::get_root()
     return root;
 }
 
+string srs_server_id_path(string pid_file)
+{
+    string path = srs_string_replace(pid_file, ".pid", ".id");
+    if (!srs_string_ends_with(path, ".id")) {
+        path += ".id";
+    }
+    return path;
+}
+
+string srs_try_read_file(string path) {
+    srs_error_t err = srs_success;
+
+    SrsFileReader r;
+    if ((err = r.open(path)) != srs_success) {
+        srs_freep(err);
+        return "";
+    }
+
+    static char buf[1024];
+    ssize_t nn = 0;
+    if ((err = r.read(buf, sizeof(buf), &nn)) != srs_success) {
+        srs_freep(err);
+        return "";
+    }
+
+    if (nn > 0) {
+        return string(buf, nn);
+    }
+    return "";
+}
+
+void srs_try_write_file(string path, string content) {
+    srs_error_t err = srs_success;
+
+    SrsFileWriter w;
+    if ((err = w.open(path)) != srs_success) {
+        srs_freep(err);
+        return;
+    }
+
+    if ((err = w.write((void*)content.data(), content.length(), NULL)) != srs_success) {
+        srs_freep(err);
+        return;
+    }
+}
+
+string SrsConfig::get_server_id()
+{
+    static string DEFAULT = "";
+
+    // Try to read DEFAULT from server id file.
+    if (DEFAULT.empty()) {
+        DEFAULT = srs_try_read_file(srs_server_id_path(get_pid_file()));
+    }
+
+    // Generate a random one if empty.
+    if (DEFAULT.empty()) {
+        DEFAULT = srs_generate_stat_vid();
+    }
+
+    // Get the server id from env, config or DEFAULT.
+    string server_id;
+
+    if (getenv("SRS_SERVER_ID")) {
+        server_id = getenv("SRS_SERVER_ID");
+    }
+
+    SrsConfDirective* conf = root->get("server_id");
+    if (conf) {
+        server_id = conf->arg0();
+    }
+
+    if (server_id.empty()) {
+        server_id = DEFAULT;
+    }
+
+    // Write server id to tmp file.
+    if (!server_id.empty()) {
+        srs_try_write_file(srs_server_id_path(get_pid_file()), server_id);
+    }
+
+    return server_id;
+}
+
 int SrsConfig::get_max_connections()
 {
     static int DEFAULT = 1000;
@@ -3128,6 +3216,20 @@ bool SrsConfig::whether_query_latest_version()
     return SRS_CONF_PERFER_TRUE(conf->arg0());
 }
 
+srs_utime_t SrsConfig::first_wait_for_qlv()
+{
+    SRS_OVERWRITE_BY_ENV_SECONDS("SRS_FIRST_WAIT_FOR_QLV");
+
+    static srs_utime_t DEFAULT = 5 * 60 * SRS_UTIME_SECONDS;
+
+    SrsConfDirective* conf = root->get("first_wait_for_qlv");
+    if (!conf) {
+        return DEFAULT;
+    }
+
+    return ::atoi(conf->arg0().c_str()) * SRS_UTIME_SECONDS;
+}
+
 bool SrsConfig::empty_ip_ok()
 {
     static bool DEFAULT = true;
@@ -3561,6 +3663,57 @@ bool SrsConfig::get_api_as_candidates()
     return SRS_CONF_PERFER_TRUE(conf->arg0());
 }
 
+bool SrsConfig::get_resolve_api_domain()
+{
+    static bool DEFAULT = true;
+
+    SrsConfDirective* conf = root->get("rtc_server");
+    if (!conf) {
+        return DEFAULT;
+    }
+
+    conf = conf->get("resolve_api_domain");
+    if (!conf || conf->arg0().empty()) {
+        return DEFAULT;
+    }
+
+    return SRS_CONF_PERFER_TRUE(conf->arg0());
+}
+
+bool SrsConfig::get_keep_api_domain()
+{
+    static bool DEFAULT = false;
+
+    SrsConfDirective* conf = root->get("rtc_server");
+    if (!conf) {
+        return DEFAULT;
+    }
+
+    conf = conf->get("keep_api_domain");
+    if (!conf || conf->arg0().empty()) {
+        return DEFAULT;
+    }
+
+    return SRS_CONF_PERFER_FALSE(conf->arg0());
+}
+
+bool SrsConfig::get_use_auto_detect_network_ip()
+{
+    static bool DEFAULT = true;
+
+    SrsConfDirective* conf = root->get("rtc_server");
+    if (!conf) {
+        return DEFAULT;
+    }
+
+    conf = conf->get("use_auto_detect_network_ip");
+    if (!conf || conf->arg0().empty()) {
+        return DEFAULT;
+    }
+
+    return SRS_CONF_PERFER_TRUE(conf->arg0());
+}
+
 std::string SrsConfig::get_rtc_server_ip_family()
 {
     static string DEFAULT = "ipv4";

trunk/src/app/srs_app_config.hpp

@@ -388,6 +388,8 @@ class SrsConfig
     // Whether user use full.conf
     virtual bool is_full_config();
 public:
+    // Get the server id, generated a random one if not configured.
+    virtual std::string get_server_id();
     // Get the max connections limit of system.
     // If  exceed the max connection, SRS will disconnect the connection.
     // @remark, linux will limit the connections of each process,
@@ -419,6 +421,7 @@ class SrsConfig
     virtual bool get_asprocess();
     // Whether query the latest available version of SRS.
     virtual bool whether_query_latest_version();
+    virtual srs_utime_t first_wait_for_qlv();
     // Whether empty client IP is ok.
     virtual bool empty_ip_ok();
     // Get the start wait in ms for gracefully quit.
@@ -470,6 +473,9 @@ class SrsConfig
     virtual int get_rtc_server_listen();
     virtual std::string get_rtc_server_candidates();
     virtual bool get_api_as_candidates();
+    virtual bool get_resolve_api_domain();
+    virtual bool get_keep_api_domain();
+    virtual bool get_use_auto_detect_network_ip();
     virtual std::string get_rtc_server_ip_family();
     virtual bool get_rtc_server_ecdsa();
     virtual bool get_rtc_server_encrypt();

trunk/src/app/srs_app_edge.cpp

@@ -201,8 +201,8 @@ void SrsEdgeIngester::stop()
 {
     trd->stop();
     upstream->close();
-    
-    // notice to unpublish.
+
+    // Notify source to un-publish if exists.
     if (source) {
         source->on_unpublish();
     }
@@ -232,6 +232,11 @@ srs_error_t SrsEdgeIngester::cycle()
             srs_freep(err);
         }
 
+        // Check whether coroutine is stopped, see https://github.com/ossrs/srs/issues/2901
+        if ((err = trd->pull()) != srs_success) {
+            return srs_error_wrap(err, "edge ingester");
+        }
+
         srs_usleep(SRS_EDGE_INGESTER_CIMS);
     }
 

trunk/src/app/srs_app_hls.cpp

@@ -202,6 +202,7 @@ SrsHlsMuxer::SrsHlsMuxer()
     async = new SrsAsyncCallWorker();
     context = new SrsTsContext();
     segments = new SrsFragmentWindow();
+    latest_acodec_ = SrsAudioCodecIdForbidden;
 
     memset(key, 0, 16);
     memset(iv, 0, 16);
@@ -263,6 +264,24 @@ int SrsHlsMuxer::deviation()
     return deviation_ts;
 }
 
+SrsAudioCodecId SrsHlsMuxer::latest_acodec()
+{
+    // If current context writer exists, we query from it.
+    if (current && current->tscw) return current->tscw->acodec();
+
+    // Get the configured or updated config.
+    return latest_acodec_;
+}
+
+void SrsHlsMuxer::set_latest_acodec(SrsAudioCodecId v)
+{
+    // Refresh the codec in context writer for current segment.
+    if (current && current->tscw) current->tscw->set_acodec(v);
+
+    // Refresh the codec for future segments.
+    latest_acodec_ = v;
+}
+
 srs_error_t SrsHlsMuxer::initialize()
 {
     return srs_success;
@@ -371,6 +390,8 @@ srs_error_t SrsHlsMuxer::segment_open()
             srs_warn("hls: use aac for other codec=%s", default_acodec_str.c_str());
         }
     }
+    // Now that we know the latest audio codec in stream, use it.
+    if (latest_acodec_ != SrsAudioCodecIdForbidden) default_acodec = latest_acodec_;
 
     // load the default vcodec from config.
     SrsVideoCodecId default_vcodec = SrsVideoCodecIdAVC;
@@ -963,6 +984,13 @@ srs_error_t SrsHlsController::on_sequence_header()
 srs_error_t SrsHlsController::write_audio(SrsAudioFrame* frame, int64_t pts)
 {
     srs_error_t err = srs_success;
+
+    // Refresh the codec ASAP.
+    if (muxer->latest_acodec() != frame->acodec()->id) {
+        srs_trace("HLS: Switch audio codec %d(%s) to %d(%s)", muxer->latest_acodec(), srs_audio_codec_id2str(muxer->latest_acodec()).c_str(),
+            frame->acodec()->id, srs_audio_codec_id2str(frame->acodec()->id).c_str());
+        muxer->set_latest_acodec(frame->acodec()->id);
+    }
 
     // write audio to cache.
     if ((err = tsmc->cache_audio(frame, pts)) != srs_success) {

trunk/src/app/srs_app_hls.hpp

@@ -156,6 +156,9 @@ class SrsHlsMuxer
     SrsHlsSegment* current;
     // The ts context, to keep cc continous between ts.
     SrsTsContext* context;
+private:
+    // Latest audio codec, parsed from stream.
+    SrsAudioCodecId latest_acodec_;
 public:
     SrsHlsMuxer();
     virtual ~SrsHlsMuxer();
@@ -166,6 +169,9 @@ class SrsHlsMuxer
     virtual std::string ts_url();
     virtual srs_utime_t duration();
     virtual int deviation();
+public:
+    SrsAudioCodecId latest_acodec();
+    void set_latest_acodec(SrsAudioCodecId v);
 public:
     // Initialize the hls muxer.
     virtual srs_error_t initialize();

trunk/src/app/srs_app_http_stream.cpp

@@ -773,7 +773,9 @@ void SrsLiveStream::http_hooks_on_stop(ISrsHttpMessage* r)
 srs_error_t SrsLiveStream::streaming_send_messages(ISrsBufferEncoder* enc, SrsSharedPtrMessage** msgs, int nb_msgs)
 {
     srs_error_t err = srs_success;
-
+
+    // TODO: In gop cache, we know both the audio and video codec, so we should notice the encoder, which might depends
+    // on setting the correct codec information, for example, HTTP-TS or HLS will write PMT.
     for (int i = 0; i < nb_msgs; i++) {
         SrsSharedPtrMessage* msg = msgs[i];
 

trunk/src/app/srs_app_latest_version.cpp

@@ -34,6 +34,7 @@ CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
 #include <srs_app_http_client.hpp>
 #include <srs_app_utility.hpp>
 #include <srs_app_uuid.hpp>
+#include <srs_app_statistic.hpp>
 
 #include <unistd.h>
 #include <sstream>
@@ -189,16 +190,8 @@ srs_error_t SrsLatestVersion::start()
         return srs_success;
     }
 
-    if (true) {
-        uuid_t uuid;
-        uuid_generate_time(uuid);
-
-        char buf[32];
-        for (int i = 0; i < 16; i++) {
-            snprintf(buf + i * 2, sizeof(buf), "%02x", uuid[i]);
-        }
-        server_id_ = string(buf, sizeof(buf));
-    }
+    server_id_ = SrsStatistic::instance()->server_id();
+    session_id_ = srs_generate_stat_vid();
 
     return trd_->start();
 }
@@ -208,15 +201,10 @@ srs_error_t SrsLatestVersion::cycle()
     srs_error_t err = srs_success;
 
     if (true) {
-        srs_utime_t first_random_wait = 0;
-        srs_random_generate((char *) &first_random_wait, 8);
-        first_random_wait = srs_utime_t(uint64_t((first_random_wait + srs_update_system_time() + getpid())) % (5 * 60)) * SRS_UTIME_SECONDS; // in s.
-
-        // Only report after 5+ minutes.
-        first_random_wait += 5 * 60 * SRS_UTIME_SECONDS;
-
-        srs_trace("Startup query id=%s, eip=%s, wait=%ds", server_id_.c_str(), srs_get_public_internet_address().c_str(), srsu2msi(first_random_wait) / 1000);
-        srs_usleep(first_random_wait);
+        srs_utime_t first_wait_for_qlv = _srs_config->first_wait_for_qlv();
+        string pip = srs_get_public_internet_address();
+        srs_trace("Startup query id=%s, session=%s, eip=%s, wait=%ds", server_id_.c_str(), session_id_.c_str(), pip.c_str(), srsu2msi(first_wait_for_qlv) / 1000);
+        srs_usleep(first_wait_for_qlv);
     }
 
     while (true) {
@@ -231,8 +219,8 @@ srs_error_t SrsLatestVersion::cycle()
             srs_freep(err); // Ignore any error.
         }
 
-        srs_trace("Finish query id=%s, eip=%s, match=%s, stable=%s, cost=%dms, url=%s",
-            server_id_.c_str(), srs_get_public_internet_address().c_str(), match_version_.c_str(),
+        srs_trace("Finish query id=%s, session=%s, eip=%s, match=%s, stable=%s, cost=%dms, url=%s",
+            server_id_.c_str(), session_id_.c_str(), srs_get_public_internet_address().c_str(), match_version_.c_str(),
             stable_version_.c_str(), srsu2msi(srs_update_system_time() - starttime), url.c_str());
 
         srs_usleep(3600 * SRS_UTIME_SECONDS); // Every an hour.
@@ -249,11 +237,12 @@ srs_error_t SrsLatestVersion::query_latest_version(string& url)
     stringstream ss;
     ss << "http://api.ossrs.net/service/v1/releases?"
           << "version=v" << VERSION_MAJOR << "." << VERSION_MINOR << "." << VERSION_REVISION
-          << "&id=" << server_id_ << "&role=srs"
+          << "&id=" << server_id_ << "&session=" << session_id_ << "&role=srs"
           << "&eip=" << srs_get_public_internet_address()
           << "&ts=" << srs_get_system_time()
           << "&alive=" << srsu2ms(srs_get_system_time() - srs_get_system_startup_time()) / 1000;
     srs_build_features(ss);
+    SrsStatistic::instance()->dumps_hints_kv(ss);
     url = ss.str();
 
     SrsHttpUri uri;

trunk/src/app/srs_app_latest_version.hpp

@@ -39,6 +39,7 @@ class SrsLatestVersion : public ISrsCoroutineHandler
 private:
     SrsCoroutine* trd_;
     std::string server_id_;
+    std::string session_id_;
 private:
     std::string match_version_;
     std::string stable_version_;

trunk/src/app/srs_app_rtc_api.cpp

@@ -121,7 +121,7 @@ srs_error_t SrsGoApiRtcPlay::do_serve_http(ISrsHttpResponseWriter* w, ISrsHttpMe
 
     srs_parse_rtmp_url(streamurl, ruc.req_->tcUrl, ruc.req_->stream);
 
-    srs_discovery_tc_url(ruc.req_->tcUrl, ruc.req_->schema, ruc.req_->host, ruc.req_->vhost, 
+    srs_discovery_tc_url(ruc.req_->tcUrl, ruc.req_->schema, ruc.req_->host, ruc.req_->vhost,
                          ruc.req_->app, ruc.req_->stream, ruc.req_->port, ruc.req_->param);
 
     // discovery vhost, resolve the vhost from config
@@ -130,10 +130,6 @@ srs_error_t SrsGoApiRtcPlay::do_serve_http(ISrsHttpResponseWriter* w, ISrsHttpMe
         ruc.req_->vhost = parsed_vhost->arg0();
     }
 
-    if ((err = http_hooks_on_play(ruc.req_)) != srs_success) {
-        return srs_error_wrap(err, "RTC: http_hooks_on_play");
-    }
-
     // For client to specifies the candidate(EIP) of server.
     string eip = r->query_get("eip");
     if (eip.empty()) {
@@ -144,9 +140,11 @@ srs_error_t SrsGoApiRtcPlay::do_serve_http(ISrsHttpResponseWriter* w, ISrsHttpMe
     string srtp = r->query_get("encrypt");
     string dtls = r->query_get("dtls");
 
-    srs_trace("RTC play %s, api=%s, tid=%s, clientip=%s, app=%s, stream=%s, offer=%dB, eip=%s, codec=%s, srtp=%s, dtls=%s",
-        streamurl.c_str(), api.c_str(), tid.c_str(), clientip.c_str(), ruc.req_->app.c_str(), ruc.req_->stream.c_str(), remote_sdp_str.length(),
-        eip.c_str(), codec.c_str(), srtp.c_str(), dtls.c_str()
+    srs_trace(
+            "RTC play %s, api=%s, tid=%s, clientip=%s, app=%s, stream=%s, offer=%dB, eip=%s, codec=%s, srtp=%s, dtls=%s",
+            streamurl.c_str(), api.c_str(), tid.c_str(), clientip.c_str(), ruc.req_->app.c_str(),
+            ruc.req_->stream.c_str(), remote_sdp_str.length(),
+            eip.c_str(), codec.c_str(), srtp.c_str(), dtls.c_str()
     );
 
     ruc.eip_ = eip;
@@ -161,6 +159,7 @@ srs_error_t SrsGoApiRtcPlay::do_serve_http(ISrsHttpResponseWriter* w, ISrsHttpMe
     }
 
     // TODO: FIXME: It seems remote_sdp doesn't represents the full SDP information.
+    ruc.remote_sdp_str_ = remote_sdp_str;
     if ((err = ruc.remote_sdp_.parse(remote_sdp_str)) != srs_success) {
         return srs_error_wrap(err, "parse sdp failed: %s", remote_sdp_str.c_str());
     }
@@ -169,42 +168,64 @@ srs_error_t SrsGoApiRtcPlay::do_serve_http(ISrsHttpResponseWriter* w, ISrsHttpMe
         return srs_error_wrap(err, "remote sdp check failed");
     }
 
+    if ((err = http_hooks_on_play(ruc.req_)) != srs_success) {
+        return srs_error_wrap(err, "RTC: http_hooks_on_play");
+    }
+
+    if ((err = serve_http(w, r, &ruc)) != srs_success) {
+        return srs_error_wrap(err, "serve");
+    }
+
+    res->set("code", SrsJsonAny::integer(ERROR_SUCCESS));
+    res->set("server", SrsJsonAny::str(SrsStatistic::instance()->server_id().c_str()));
+
+    // TODO: add candidates in response json?
+    res->set("sdp", SrsJsonAny::str(ruc.local_sdp_str_.c_str()));
+    res->set("sessionid", SrsJsonAny::str(ruc.session_id_.c_str()));
+
+    return err;
+}
+
+srs_error_t SrsGoApiRtcPlay::serve_http(ISrsHttpResponseWriter* w, ISrsHttpMessage* r, SrsRtcUserConfig* ruc)
+{
+    srs_error_t  err = srs_success;
+
     SrsSdp local_sdp;
 
     // Config for SDP and session.
-    local_sdp.session_config_.dtls_role = _srs_config->get_rtc_dtls_role(ruc.req_->vhost);
-    local_sdp.session_config_.dtls_version = _srs_config->get_rtc_dtls_version(ruc.req_->vhost);
+    local_sdp.session_config_.dtls_role = _srs_config->get_rtc_dtls_role(ruc->req_->vhost);
+    local_sdp.session_config_.dtls_version = _srs_config->get_rtc_dtls_version(ruc->req_->vhost);
 
     // Whether enabled.
     bool server_enabled = _srs_config->get_rtc_server_enabled();
-    bool rtc_enabled = _srs_config->get_rtc_enabled(ruc.req_->vhost);
+    bool rtc_enabled = _srs_config->get_rtc_enabled(ruc->req_->vhost);
     if (server_enabled && !rtc_enabled) {
-        srs_warn("RTC disabled in vhost %s", ruc.req_->vhost.c_str());
+        srs_warn("RTC disabled in vhost %s", ruc->req_->vhost.c_str());
     }
     if (!server_enabled || !rtc_enabled) {
         return srs_error_new(ERROR_RTC_DISABLED, "Disabled server=%d, rtc=%d, vhost=%s",
-            server_enabled, rtc_enabled, ruc.req_->vhost.c_str());
+            server_enabled, rtc_enabled, ruc->req_->vhost.c_str());
     }
 
     // Whether RTC stream is active.
     bool is_rtc_stream_active = false;
     if (true) {
-        SrsRtcSource* source = _srs_rtc_sources->fetch(ruc.req_);
+        SrsRtcSource* source = _srs_rtc_sources->fetch(ruc->req_);
         is_rtc_stream_active = (source && !source->can_publish());
     }
 
     // For RTMP to RTC, fail if disabled and RTMP is active, see https://github.com/ossrs/srs/issues/2728
-    if (!is_rtc_stream_active && !_srs_config->get_rtc_from_rtmp(ruc.req_->vhost)) {
-        SrsLiveSource* rtmp = _srs_sources->fetch(ruc.req_);
+    if (!is_rtc_stream_active && !_srs_config->get_rtc_from_rtmp(ruc->req_->vhost)) {
+        SrsLiveSource* rtmp = _srs_sources->fetch(ruc->req_);
         if (rtmp && !rtmp->inactive()) {
-            return srs_error_new(ERROR_RTC_DISABLED, "Disabled rtmp_to_rtc of %s, see #2728", ruc.req_->vhost.c_str());
+            return srs_error_new(ERROR_RTC_DISABLED, "Disabled rtmp_to_rtc of %s, see #2728", ruc->req_->vhost.c_str());
         }
     }
 
     // TODO: FIXME: When server enabled, but vhost disabled, should report error.
     SrsRtcConnection* session = NULL;
-    if ((err = server_->create_session(&ruc, local_sdp, &session)) != srs_success) {
-        return srs_error_wrap(err, "create session, dtls=%u, srtp=%u, eip=%s", ruc.dtls_, ruc.srtp_, eip.c_str());
+    if ((err = server_->create_session(ruc, local_sdp, &session)) != srs_success) {
+        return srs_error_wrap(err, "create session, dtls=%u, srtp=%u, eip=%s", ruc->dtls_, ruc->srtp_, ruc->eip_.c_str());
     }
 
     ostringstream os;
@@ -216,17 +237,12 @@ srs_error_t SrsGoApiRtcPlay::do_serve_http(ISrsHttpResponseWriter* w, ISrsHttpMe
     // Filter the \r\n to \\r\\n for JSON.
     string local_sdp_escaped = srs_string_replace(local_sdp_str.c_str(), "\r\n", "\\r\\n");
 
-    res->set("code", SrsJsonAny::integer(ERROR_SUCCESS));
-    res->set("server", SrsJsonAny::str(SrsStatistic::instance()->server_id().c_str()));
-
-    // TODO: add candidates in response json?
-
-    res->set("sdp", SrsJsonAny::str(local_sdp_str.c_str()));
-    res->set("sessionid", SrsJsonAny::str(session->username().c_str()));
+    ruc->local_sdp_str_ = local_sdp_str;
+    ruc->session_id_ = session->username();
 
     srs_trace("RTC username=%s, dtls=%u, srtp=%u, offer=%dB, answer=%dB", session->username().c_str(),
-        ruc.dtls_, ruc.srtp_, remote_sdp_str.length(), local_sdp_escaped.length());
-    srs_trace("RTC remote offer: %s", srs_string_replace(remote_sdp_str.c_str(), "\r\n", "\\r\\n").c_str());
+        ruc->dtls_, ruc->srtp_, ruc->remote_sdp_str_.length(), local_sdp_escaped.length());
+    srs_trace("RTC remote offer: %s", srs_string_replace(ruc->remote_sdp_str_.c_str(), "\r\n", "\\r\\n").c_str());
     srs_trace("RTC local answer: %s", local_sdp_escaped.c_str());
 
     return err;
@@ -333,8 +349,7 @@ srs_error_t SrsGoApiRtcPublish::do_serve_http(ISrsHttpResponseWriter* w, ISrsHtt
     srs_error_t err = srs_success;
 
     // For each RTC session, we use short-term HTTP connection.
-    SrsHttpHeader* hdr = w->header();
-    hdr->set("Connection", "Close");
+    w->header()->set("Connection", "Close");
 
     // Parse req, the request json object, from body.
     SrsJsonObject* req = NULL;
@@ -406,10 +421,6 @@ srs_error_t SrsGoApiRtcPublish::do_serve_http(ISrsHttpResponseWriter* w, ISrsHtt
         ruc.req_->vhost = parsed_vhost->arg0();
     }
 
-	if ((err = http_hooks_on_publish(ruc.req_)) != srs_success) {
-        return srs_error_wrap(err, "RTC: http_hooks_on_publish");
-    }
-
     // For client to specifies the candidate(EIP) of server.
     string eip = r->query_get("eip");
     if (eip.empty()) {
@@ -428,35 +439,58 @@ srs_error_t SrsGoApiRtcPublish::do_serve_http(ISrsHttpResponseWriter* w, ISrsHtt
     ruc.dtls_ = ruc.srtp_ = true;
 
     // TODO: FIXME: It seems remote_sdp doesn't represents the full SDP information.
+    ruc.remote_sdp_str_ = remote_sdp_str;
     if ((err = ruc.remote_sdp_.parse(remote_sdp_str)) != srs_success) {
         return srs_error_wrap(err, "parse sdp failed: %s", remote_sdp_str.c_str());
     }
 
-    if ((err = check_remote_sdp(ruc.remote_sdp_)) != srs_success) {
+    if ((err = serve_http(w, r, &ruc)) != srs_success) {
+        return srs_error_wrap(err, "serve");
+    }
+
+    res->set("code", SrsJsonAny::integer(ERROR_SUCCESS));
+    res->set("server", SrsJsonAny::str(SrsStatistic::instance()->server_id().c_str()));
+
+    // TODO: add candidates in response json?
+    res->set("sdp", SrsJsonAny::str(ruc.local_sdp_str_.c_str()));
+    res->set("sessionid", SrsJsonAny::str(ruc.session_id_.c_str()));
+
+    return err;
+}
+
+srs_error_t SrsGoApiRtcPublish::serve_http(ISrsHttpResponseWriter* w, ISrsHttpMessage* r, SrsRtcUserConfig* ruc)
+{
+    srs_error_t  err = srs_success;
+
+    if ((err = check_remote_sdp(ruc->remote_sdp_)) != srs_success) {
         return srs_error_wrap(err, "remote sdp check failed");
     }
 
+    if ((err = http_hooks_on_publish(ruc->req_)) != srs_success) {
+        return srs_error_wrap(err, "RTC: http_hooks_on_publish");
+    }
+
     SrsSdp local_sdp;
 
     // TODO: FIXME: move to create_session.
     // Config for SDP and session.
-    local_sdp.session_config_.dtls_role = _srs_config->get_rtc_dtls_role(ruc.req_->vhost);
-    local_sdp.session_config_.dtls_version = _srs_config->get_rtc_dtls_version(ruc.req_->vhost);
+    local_sdp.session_config_.dtls_role = _srs_config->get_rtc_dtls_role(ruc->req_->vhost);
+    local_sdp.session_config_.dtls_version = _srs_config->get_rtc_dtls_version(ruc->req_->vhost);
 
     // Whether enabled.
     bool server_enabled = _srs_config->get_rtc_server_enabled();
-    bool rtc_enabled = _srs_config->get_rtc_enabled(ruc.req_->vhost);
+    bool rtc_enabled = _srs_config->get_rtc_enabled(ruc->req_->vhost);
     if (server_enabled && !rtc_enabled) {
-        srs_warn("RTC disabled in vhost %s", ruc.req_->vhost.c_str());
+        srs_warn("RTC disabled in vhost %s", ruc->req_->vhost.c_str());
     }
     if (!server_enabled || !rtc_enabled) {
         return srs_error_new(ERROR_RTC_DISABLED, "Disabled server=%d, rtc=%d, vhost=%s",
-            server_enabled, rtc_enabled, ruc.req_->vhost.c_str());
+            server_enabled, rtc_enabled, ruc->req_->vhost.c_str());
     }
 
     // TODO: FIXME: When server enabled, but vhost disabled, should report error.
     SrsRtcConnection* session = NULL;
-    if ((err = server_->create_session(&ruc, local_sdp, &session)) != srs_success) {
+    if ((err = server_->create_session(ruc, local_sdp, &session)) != srs_success) {
         return srs_error_wrap(err, "create session");
     }
 
@@ -469,17 +503,12 @@ srs_error_t SrsGoApiRtcPublish::do_serve_http(ISrsHttpResponseWriter* w, ISrsHtt
     // Filter the \r\n to \\r\\n for JSON.
     string local_sdp_escaped = srs_string_replace(local_sdp_str.c_str(), "\r\n", "\\r\\n");
 
-    res->set("code", SrsJsonAny::integer(ERROR_SUCCESS));
-    res->set("server", SrsJsonAny::str(SrsStatistic::instance()->server_id().c_str()));
-
-    // TODO: add candidates in response json?
-
-    res->set("sdp", SrsJsonAny::str(local_sdp_str.c_str()));
-    res->set("sessionid", SrsJsonAny::str(session->username().c_str()));
+    ruc->local_sdp_str_ = local_sdp_str;
+    ruc->session_id_ = session->username();
 
     srs_trace("RTC username=%s, offer=%dB, answer=%dB", session->username().c_str(),
-        remote_sdp_str.length(), local_sdp_escaped.length());
-    srs_trace("RTC remote offer: %s", srs_string_replace(remote_sdp_str.c_str(), "\r\n", "\\r\\n").c_str());
+        ruc->remote_sdp_str_.length(), local_sdp_escaped.length());
+    srs_trace("RTC remote offer: %s", srs_string_replace(ruc->remote_sdp_str_.c_str(), "\r\n", "\\r\\n").c_str());
     srs_trace("RTC local answer: %s", local_sdp_escaped.c_str());
 
     return err;
@@ -545,6 +574,100 @@ srs_error_t SrsGoApiRtcPublish::http_hooks_on_publish(SrsRequest* req)
     return err;
 }
 
+SrsGoApiRtcWhip::SrsGoApiRtcWhip(SrsRtcServer* server)
+{
+    publish_ = new SrsGoApiRtcPublish(server);
+    play_ = new SrsGoApiRtcPlay(server);
+}
+
+SrsGoApiRtcWhip::~SrsGoApiRtcWhip()
+{
+    srs_freep(publish_);
+    srs_freep(play_);
+}
+
+srs_error_t SrsGoApiRtcWhip::serve_http(ISrsHttpResponseWriter* w, ISrsHttpMessage* r)
+{
+    srs_error_t err = srs_success;
+
+    // For each RTC session, we use short-term HTTP connection.
+    w->header()->set("Connection", "Close");
+
+    string remote_sdp_str;
+    if ((err = r->body_read_all(remote_sdp_str)) != srs_success) {
+        return srs_error_wrap(err, "read sdp");
+    }
+
+    string clientip;
+    if (clientip.empty()){
+        clientip = dynamic_cast<SrsHttpMessage*>(r)->connection()->remote_ip();
+        // Overwrite by ip from proxy.
+        string oip = srs_get_original_ip(r);
+        if (!oip.empty()) {
+            clientip = oip;
+        }
+    }
+
+    // For client to specifies the candidate(EIP) of server.
+    string eip = r->query_get("eip");
+    if (eip.empty()) {
+        eip = r->query_get("candidate");
+    }
+    string codec = r->query_get("codec");
+    string app = r->query_get("app");
+    string stream = r->query_get("stream");
+    string action = r->query_get("action");
+    if (action.empty()) {
+        action = "publish";
+    }
+    if (srs_string_ends_with(r->path(), "/whip-play/")) {
+        action = "play";
+    }
+
+    // The RTC user config object.
+    SrsRtcUserConfig ruc;
+    ruc.req_->ip = clientip;
+    ruc.req_->host = r->host();
+    ruc.req_->vhost = ruc.req_->host;
+    ruc.req_->app = app.empty() ? "live" : app;
+    ruc.req_->stream = stream.empty() ? "livestream" : stream;
+
+    // discovery vhost, resolve the vhost from config
+    SrsConfDirective* parsed_vhost = _srs_config->get_vhost(ruc.req_->vhost);
+    if (parsed_vhost) {
+        ruc.req_->vhost = parsed_vhost->arg0();
+    }
+
+    srs_trace("RTC whip %s %s, clientip=%s, app=%s, stream=%s, offer=%dB, eip=%s, codec=%s",
+        action.c_str(), ruc.req_->get_stream_url().c_str(), clientip.c_str(), ruc.req_->app.c_str(), ruc.req_->stream.c_str(),
+        remote_sdp_str.length(), eip.c_str(), codec.c_str()
+    );
+
+    ruc.eip_ = eip;
+    ruc.codec_ = codec;
+    ruc.publish_ = (action == "publish");
+    ruc.dtls_ = ruc.srtp_ = true;
+
+    // TODO: FIXME: It seems remote_sdp doesn't represents the full SDP information.
+    ruc.remote_sdp_str_ = remote_sdp_str;
+    if ((err = ruc.remote_sdp_.parse(remote_sdp_str)) != srs_success) {
+        return srs_error_wrap(err, "parse sdp failed: %s", remote_sdp_str.c_str());
+    }
+
+    err = action == "publish" ? publish_->serve_http(w, r, &ruc) : play_->serve_http(w, r, &ruc);
+    if (err != srs_success) {
+        return srs_error_wrap(err, "serve");
+    }
+
+    if (ruc.local_sdp_str_.empty()) {
+        return srs_go_http_error(w, SRS_CONSTS_HTTP_InternalServerError);
+    }
+
+    string sdp = ruc.local_sdp_str_;
+    w->header()->set("Content-Type", "application/sdp");
+    return w->write((char*)sdp.data(), (int)sdp.length());
+}
+
 SrsGoApiRtcNACK::SrsGoApiRtcNACK(SrsRtcServer* server)
 {
     server_ = server;

trunk/src/app/srs_app_rtc_api.hpp

@@ -14,6 +14,7 @@
 class SrsRtcServer;
 class SrsRequest;
 class SrsSdp;
+class SrsRtcUserConfig;
 
 class SrsGoApiRtcPlay : public ISrsHttpHandler
 {
@@ -26,6 +27,9 @@ class SrsGoApiRtcPlay : public ISrsHttpHandler
     virtual srs_error_t serve_http(ISrsHttpResponseWriter* w, ISrsHttpMessage* r);
 private:
     virtual srs_error_t do_serve_http(ISrsHttpResponseWriter* w, ISrsHttpMessage* r, SrsJsonObject* res);
+public:
+    virtual srs_error_t serve_http(ISrsHttpResponseWriter* w, ISrsHttpMessage* r, SrsRtcUserConfig* ruc);
+private:
     srs_error_t check_remote_sdp(const SrsSdp& remote_sdp);
 private:
     virtual srs_error_t http_hooks_on_play(SrsRequest* req);
@@ -42,11 +46,27 @@ class SrsGoApiRtcPublish : public ISrsHttpHandler
     virtual srs_error_t serve_http(ISrsHttpResponseWriter* w, ISrsHttpMessage* r);
 private:
     virtual srs_error_t do_serve_http(ISrsHttpResponseWriter* w, ISrsHttpMessage* r, SrsJsonObject* res);
+public:
+    virtual srs_error_t serve_http(ISrsHttpResponseWriter* w, ISrsHttpMessage* r, SrsRtcUserConfig* ruc);
+private:
     srs_error_t check_remote_sdp(const SrsSdp& remote_sdp);
 private:
     virtual srs_error_t http_hooks_on_publish(SrsRequest* req);
 };
 
+// See https://datatracker.ietf.org/doc/draft-ietf-wish-whip/
+class SrsGoApiRtcWhip : public ISrsHttpHandler
+{
+private:
+    SrsGoApiRtcPublish* publish_;
+    SrsGoApiRtcPlay* play_;
+public:
+    SrsGoApiRtcWhip(SrsRtcServer* server);
+    virtual ~SrsGoApiRtcWhip();
+public:
+    virtual srs_error_t serve_http(ISrsHttpResponseWriter* w, ISrsHttpMessage* r);
+};
+
 class SrsGoApiRtcNACK : public ISrsHttpHandler
 {
 private:

trunk/src/app/srs_app_rtc_conn.cpp

@@ -510,10 +510,12 @@ srs_error_t SrsRtcPlayStream::initialize(SrsRequest* req, std::map<uint32_t, Srs
 
 void SrsRtcPlayStream::on_stream_change(SrsRtcSourceDescription* desc)
 {
+    if (!desc) return;
+
     // Refresh the relation for audio.
     // TODO: FIMXE: Match by label?
     if (desc && desc->audio_track_desc_ && audio_tracks_.size() == 1) {
-        if (! audio_tracks_.empty()) {
+        if (!audio_tracks_.empty()) {
             uint32_t ssrc = desc->audio_track_desc_->ssrc_;
             SrsRtcAudioSendTrack* track = audio_tracks_.begin()->second;
 
@@ -525,7 +527,7 @@ void SrsRtcPlayStream::on_stream_change(SrsRtcSourceDescription* desc)
     // Refresh the relation for video.
     // TODO: FIMXE: Match by label?
     if (desc && desc->video_track_descs_.size() == 1) {
-        if (! video_tracks_.empty()) {
+        if (!video_tracks_.empty()) {
             SrsRtcTrackDescription* vdesc = desc->video_track_descs_.at(0);
             uint32_t ssrc = vdesc->ssrc_;
             SrsRtcVideoSendTrack* track = video_tracks_.begin()->second;
@@ -534,6 +536,15 @@ void SrsRtcPlayStream::on_stream_change(SrsRtcSourceDescription* desc)
             video_tracks_.insert(make_pair(ssrc, track));
         }
     }
+
+    // Request keyframe(PLI) when stream changed.
+    if (desc->audio_track_desc_) {
+        pli_worker_->request_keyframe(desc->audio_track_desc_->ssrc_, cid_);
+    }
+    for (vector<SrsRtcTrackDescription*>::iterator it = desc->video_track_descs_.begin(); it != desc->video_track_descs_.end(); ++it) {
+        SrsRtcTrackDescription* vdesc = *it;
+        pli_worker_->request_keyframe(vdesc->ssrc_, cid_);
+    }
 }
 
 srs_error_t SrsRtcPlayStream::on_reload_vhost_play(string vhost)
@@ -1981,7 +1992,7 @@ srs_error_t SrsRtcConnection::add_publisher(SrsRtcUserConfig* ruc, SrsSdp& local
         return srs_error_wrap(err, "publish negotiate");
     }
 
-    if ((err = generate_publish_local_sdp(req, local_sdp, stream_desc, ruc->remote_sdp_.is_unified())) != srs_success) {
+    if ((err = generate_publish_local_sdp(req, local_sdp, stream_desc, ruc->remote_sdp_.is_unified(), ruc->audio_before_video_)) != srs_success) {
         return srs_error_wrap(err, "generate local sdp");
     }
 
@@ -2047,7 +2058,7 @@ srs_error_t SrsRtcConnection::add_player(SrsRtcUserConfig* ruc, SrsSdp& local_sd
         ++it;
     }
 
-    if ((err = generate_play_local_sdp(req, local_sdp, stream_desc, ruc->remote_sdp_.is_unified())) != srs_success) {
+    if ((err = generate_play_local_sdp(req, local_sdp, stream_desc, ruc->remote_sdp_.is_unified(), ruc->audio_before_video_)) != srs_success) {
         return srs_error_wrap(err, "generate local sdp");
     }
 
@@ -2842,9 +2853,14 @@ srs_error_t SrsRtcConnection::negotiate_publish_capability(SrsRtcUserConfig* ruc
     // TODO: FIME: Should check packetization-mode=1 also.
     bool has_42e01f = srs_sdp_has_h264_profile(remote_sdp, "42e01f");
 
+    // How many video descriptions we have parsed.
+    int nn_any_video_parsed = 0;
+
     for (int i = 0; i < (int)remote_sdp.media_descs_.size(); ++i) {
         const SrsMediaDesc& remote_media_desc = remote_sdp.media_descs_.at(i);
 
+        if (remote_media_desc.is_video()) nn_any_video_parsed++;
+
         SrsRtcTrackDescription* track_desc = new SrsRtcTrackDescription();
         SrsAutoFree(SrsRtcTrackDescription, track_desc);
 
@@ -2867,6 +2883,9 @@ srs_error_t SrsRtcConnection::negotiate_publish_capability(SrsRtcUserConfig* ruc
         }
 
         if (remote_media_desc.is_audio()) {
+            // Update the ruc, which is about user specified configuration.
+            ruc->audio_before_video_ = !nn_any_video_parsed;
+
             // TODO: check opus format specific param
             std::vector<SrsMediaPayloadType> payloads = remote_media_desc.find_media_with_encoding_name("opus");
             if (payloads.empty()) {
@@ -2966,6 +2985,9 @@ srs_error_t SrsRtcConnection::negotiate_publish_capability(SrsRtcUserConfig* ruc
                     SrsVideoPayload* video_payload = new SrsVideoPayload(payload.payload_type_, payload.encoding_name_, payload.clock_rate_);
                     video_payload->set_h264_param_desc(payload.format_specific_param_);
 
+                    // Set the codec parameter for H.264, to make Unity happy.
+                    video_payload->h264_param_ = h264_param;
+
                     // TODO: FIXME: Only support some transport algorithms.
                     for (int k = 0; k < (int)payload.rtcp_fb_.size(); ++k) {
                         const string& rtcp_fb = payload.rtcp_fb_.at(k);
@@ -3015,6 +3037,7 @@ srs_error_t SrsRtcConnection::negotiate_publish_capability(SrsRtcUserConfig* ruc
                     }
                 }
 
+                track_desc->type_ = "video";
                 track_desc->set_codec_payload((SrsCodecPayload*)video_payload);
                 srs_warn("choose backup H.264 payload type=%d", payload.payload_type_);
             }
@@ -3023,6 +3046,11 @@ srs_error_t SrsRtcConnection::negotiate_publish_capability(SrsRtcUserConfig* ruc
             //local_media_desc.payload_types_.back().rtcp_fb_.push_back("rrtr");
         }
 
+        // Error if track desc is invalid, that is failed to match SDP, for example, we require H264 but no H264 found.
+        if (track_desc->type_.empty() || !track_desc->media_) {
+            return srs_error_new(ERROR_RTC_SDP_EXCHANGE, "no match for track=%s, mid=%s, tracker=%s", remote_media_desc.type_.c_str(), remote_media_desc.mid_.c_str(), remote_media_desc.msid_tracker_.c_str());
+        }
+
         // TODO: FIXME: use one parse payload from sdp.
 
         track_desc->create_auxiliary_payload(remote_media_desc.find_media_with_encoding_name("red"));
@@ -3044,6 +3072,8 @@ srs_error_t SrsRtcConnection::negotiate_publish_capability(SrsRtcUserConfig* ruc
                     stream_desc->audio_track_desc_ = track_desc_copy;
                 } else if (remote_media_desc.is_video()) {
                     stream_desc->video_track_descs_.push_back(track_desc_copy);
+                } else {
+                    srs_freep(track_desc_copy);
                 }
             }
             track_id = ssrc_info.msid_tracker_;
@@ -3069,7 +3099,7 @@ srs_error_t SrsRtcConnection::negotiate_publish_capability(SrsRtcUserConfig* ruc
     return err;
 }
 
-srs_error_t SrsRtcConnection::generate_publish_local_sdp(SrsRequest* req, SrsSdp& local_sdp, SrsRtcSourceDescription* stream_desc, bool unified_plan)
+srs_error_t SrsRtcConnection::generate_publish_local_sdp(SrsRequest* req, SrsSdp& local_sdp, SrsRtcSourceDescription* stream_desc, bool unified_plan, bool audio_before_video)
 {
     srs_error_t err = srs_success;
 
@@ -3094,6 +3124,29 @@ srs_error_t SrsRtcConnection::generate_publish_local_sdp(SrsRequest* req, SrsSdp
 
     local_sdp.group_policy_ = "BUNDLE";
 
+    if (audio_before_video) {
+        if ((err = generate_publish_local_sdp_for_audio(local_sdp, stream_desc)) != srs_success) {
+            return srs_error_wrap(err, "audio");
+        }
+        if ((err = generate_publish_local_sdp_for_video(local_sdp, stream_desc, unified_plan)) != srs_success) {
+            return srs_error_wrap(err, "video");
+        }
+    } else {
+        if ((err = generate_publish_local_sdp_for_video(local_sdp, stream_desc, unified_plan)) != srs_success) {
+            return srs_error_wrap(err, "video");
+        }
+        if ((err = generate_publish_local_sdp_for_audio(local_sdp, stream_desc)) != srs_success) {
+            return srs_error_wrap(err, "audio");
+        }
+    }
+
+    return err;
+}
+
+srs_error_t SrsRtcConnection::generate_publish_local_sdp_for_audio(SrsSdp& local_sdp, SrsRtcSourceDescription* stream_desc)
+{
+    srs_error_t err = srs_success;
+
     // generate audio media desc
     if (stream_desc->audio_track_desc_) {
         SrsRtcTrackDescription* audio_track = stream_desc->audio_track_desc_;
@@ -3128,6 +3181,13 @@ srs_error_t SrsRtcConnection::generate_publish_local_sdp(SrsRequest* req, SrsSdp
         local_media_desc.payload_types_.push_back(payload->generate_media_payload_type());
     }
 
+    return err;
+}
+
+srs_error_t SrsRtcConnection::generate_publish_local_sdp_for_video(SrsSdp& local_sdp, SrsRtcSourceDescription* stream_desc, bool unified_plan)
+{
+    srs_error_t err = srs_success;
+
     for (int i = 0;  i < (int)stream_desc->video_track_descs_.size(); ++i) {
         SrsRtcTrackDescription* video_track = stream_desc->video_track_descs_.at(i);
 
@@ -3191,9 +3251,14 @@ srs_error_t SrsRtcConnection::negotiate_play_capability(SrsRtcUserConfig* ruc, s
         return srs_error_wrap(err, "fetch rtc source");
     }
 
+    // How many video descriptions we have parsed.
+    int nn_any_video_parsed = 0;
+
     for (int i = 0; i < (int)remote_sdp.media_descs_.size(); ++i) {
         const SrsMediaDesc& remote_media_desc = remote_sdp.media_descs_.at(i);
 
+        if (remote_media_desc.is_video()) nn_any_video_parsed++;
+
         // Whether feature enabled in remote extmap.
         int remote_twcc_id = 0;
         if (true) {
@@ -3209,6 +3274,9 @@ srs_error_t SrsRtcConnection::negotiate_play_capability(SrsRtcUserConfig* ruc, s
         std::vector<SrsRtcTrackDescription*> track_descs;
         SrsMediaPayloadType remote_payload(0);
         if (remote_media_desc.is_audio()) {
+            // Update the ruc, which is about user specified configuration.
+            ruc->audio_before_video_ = !nn_any_video_parsed;
+
             // TODO: check opus format specific param
             vector<SrsMediaPayloadType> payloads = remote_media_desc.find_media_with_encoding_name("opus");
             if (payloads.empty()) {
@@ -3368,7 +3436,7 @@ void video_track_generate_play_offer(SrsRtcTrackDescription* track, string mid,
     }
 }
 
-srs_error_t SrsRtcConnection::generate_play_local_sdp(SrsRequest* req, SrsSdp& local_sdp, SrsRtcSourceDescription* stream_desc, bool unified_plan)
+srs_error_t SrsRtcConnection::generate_play_local_sdp(SrsRequest* req, SrsSdp& local_sdp, SrsRtcSourceDescription* stream_desc, bool unified_plan, bool audio_before_video)
 {
     srs_error_t err = srs_success;
 
@@ -3395,6 +3463,29 @@ srs_error_t SrsRtcConnection::generate_play_local_sdp(SrsRequest* req, SrsSdp& l
 
     std::string cname = srs_random_str(16);
 
+    if (audio_before_video) {
+        if ((err = generate_play_local_sdp_for_audio(local_sdp, stream_desc, cname)) != srs_success) {
+            return srs_error_wrap(err, "audio");
+        }
+        if ((err = generate_play_local_sdp_for_video(local_sdp, stream_desc, unified_plan, cname)) != srs_success) {
+            return srs_error_wrap(err, "video");
+        }
+    } else {
+        if ((err = generate_play_local_sdp_for_video(local_sdp, stream_desc, unified_plan, cname)) != srs_success) {
+            return srs_error_wrap(err, "video");
+        }
+        if ((err = generate_play_local_sdp_for_audio(local_sdp, stream_desc, cname)) != srs_success) {
+            return srs_error_wrap(err, "audio");
+        }
+    }
+
+    return err;
+}
+
+srs_error_t SrsRtcConnection::generate_play_local_sdp_for_audio(SrsSdp& local_sdp, SrsRtcSourceDescription* stream_desc, std::string cname)
+{
+    srs_error_t err = srs_success;
+
     // generate audio media desc
     if (stream_desc->audio_track_desc_) {
         SrsRtcTrackDescription* audio_track = stream_desc->audio_track_desc_;
@@ -3454,6 +3545,13 @@ srs_error_t SrsRtcConnection::generate_play_local_sdp(SrsRequest* req, SrsSdp& l
         }
     }
 
+    return err;
+}
+
+srs_error_t SrsRtcConnection::generate_play_local_sdp_for_video(SrsSdp& local_sdp, SrsRtcSourceDescription* stream_desc, bool unified_plan, std::string cname)
+{
+    srs_error_t err = srs_success;
+
     for (int i = 0;  i < (int)stream_desc->video_track_descs_.size(); ++i) {
         SrsRtcTrackDescription* track = stream_desc->video_track_descs_[i];
 

trunk/src/app/srs_app_rtc_conn.hpp

@@ -568,11 +568,15 @@ class SrsRtcConnection : public ISrsResource, public ISrsDisposingHandler, publi
     srs_error_t on_binding_request(SrsStunPacket* r);
     // publish media capabilitiy negotiate
     srs_error_t negotiate_publish_capability(SrsRtcUserConfig* ruc, SrsRtcSourceDescription* stream_desc);
-    srs_error_t generate_publish_local_sdp(SrsRequest* req, SrsSdp& local_sdp, SrsRtcSourceDescription* stream_desc, bool unified_plan);
+    srs_error_t generate_publish_local_sdp(SrsRequest* req, SrsSdp& local_sdp, SrsRtcSourceDescription* stream_desc, bool unified_plan, bool audio_before_video);
+    srs_error_t generate_publish_local_sdp_for_audio(SrsSdp& local_sdp, SrsRtcSourceDescription* stream_desc);
+    srs_error_t generate_publish_local_sdp_for_video(SrsSdp& local_sdp, SrsRtcSourceDescription* stream_desc, bool unified_plan);
     // play media capabilitiy negotiate
     //TODO: Use StreamDescription to negotiate and remove first negotiate_play_capability function
     srs_error_t negotiate_play_capability(SrsRtcUserConfig* ruc, std::map<uint32_t, SrsRtcTrackDescription*>& sub_relations);
-    srs_error_t generate_play_local_sdp(SrsRequest* req, SrsSdp& local_sdp, SrsRtcSourceDescription* stream_desc, bool unified_plan);
+    srs_error_t generate_play_local_sdp(SrsRequest* req, SrsSdp& local_sdp, SrsRtcSourceDescription* stream_desc, bool unified_plan, bool audio_before_video);
+    srs_error_t generate_play_local_sdp_for_audio(SrsSdp& local_sdp, SrsRtcSourceDescription* stream_desc, std::string cname);
+    srs_error_t generate_play_local_sdp_for_video(SrsSdp& local_sdp, SrsRtcSourceDescription* stream_desc, bool unified_plan, std::string cname);
     srs_error_t create_player(SrsRequest* request, std::map<uint32_t, SrsRtcTrackDescription*> sub_relations);
     srs_error_t create_publisher(SrsRequest* request, SrsRtcSourceDescription* stream_desc);
 };

trunk/src/app/srs_app_rtc_sdp.cpp

@@ -56,32 +56,39 @@ static void skip_first_spaces(std::string& str)
 srs_error_t srs_parse_h264_fmtp(const std::string& fmtp, H264SpecificParam& h264_param)
 {
     srs_error_t err = srs_success;
-    std::vector<std::string> vec = split_str(fmtp, ";");
+
+    std::vector<std::string> vec = srs_string_split(fmtp, ";");
     for (size_t i = 0; i < vec.size(); ++i) {
-        std::vector<std::string> kv = split_str(vec[i], "=");
-        if (kv.size() == 2) {
-            if (kv[0] == "profile-level-id") {
-                h264_param.profile_level_id = kv[1];
-            } else if (kv[0] == "packetization-mode") {
-                // 6.3.  Non-Interleaved Mode
-                // This mode is in use when the value of the OPTIONAL packetization-mode
-                // media type parameter is equal to 1.  This mode SHOULD be supported.
-                // It is primarily intended for low-delay applications.  Only single NAL
-                // unit packets, STAP-As, and FU-As MAY be used in this mode.  STAP-Bs,
-                // MTAPs, and FU-Bs MUST NOT be used.  The transmission order of NAL
-                // units MUST comply with the NAL unit decoding order.
-                // @see https://tools.ietf.org/html/rfc6184#section-6.3
-                h264_param.packetization_mode = kv[1];
-            } else if (kv[0] == "level-asymmetry-allowed") {
-                h264_param.level_asymmerty_allow = kv[1];
-            } else {
-                return srs_error_new(ERROR_RTC_SDP_DECODE, "invalid h264 param=%s", kv[0].c_str());
-            }
-        } else {
-            return srs_error_new(ERROR_RTC_SDP_DECODE, "invalid h264 param=%s", vec[i].c_str());
+        std::vector<std::string> kv = srs_string_split(vec[i], "=");
+        if (kv.size() != 2) continue;
+
+        if (kv[0] == "profile-level-id") {
+            h264_param.profile_level_id = kv[1];
+        } else if (kv[0] == "packetization-mode") {
+            // 6.3.  Non-Interleaved Mode
+            // This mode is in use when the value of the OPTIONAL packetization-mode
+            // media type parameter is equal to 1.  This mode SHOULD be supported.
+            // It is primarily intended for low-delay applications.  Only single NAL
+            // unit packets, STAP-As, and FU-As MAY be used in this mode.  STAP-Bs,
+            // MTAPs, and FU-Bs MUST NOT be used.  The transmission order of NAL
+            // units MUST comply with the NAL unit decoding order.
+            // @see https://tools.ietf.org/html/rfc6184#section-6.3
+            h264_param.packetization_mode = kv[1];
+        } else if (kv[0] == "level-asymmetry-allowed") {
+            h264_param.level_asymmerty_allow = kv[1];
         }
     }
 
+    if (h264_param.profile_level_id.empty()) {
+        return srs_error_new(ERROR_RTC_SDP_DECODE, "no h264 param: profile-level-id");
+    }
+    if (h264_param.packetization_mode.empty()) {
+        return srs_error_new(ERROR_RTC_SDP_DECODE, "no h264 param: packetization-mode");
+    }
+    if (h264_param.level_asymmerty_allow.empty()) {
+        return srs_error_new(ERROR_RTC_SDP_DECODE, "no h264 param: level-asymmetry-allowed");
+    }
+
     return err;
 }
 

trunk/src/app/srs_app_rtc_server.cpp

@@ -158,28 +158,35 @@ srs_error_t api_server_as_candidates(string api, set<string>& candidate_ips)
         return err;
     }
 
-    SrsHttpUri uri;
-    if ((err = uri.initialize(api)) != srs_success) {
-        return srs_error_wrap(err, "parse %s", api.c_str());
-    }
-
-    string hostname = uri.get_host();
+    string hostname = api;
     if (hostname.empty() || hostname == SRS_CONSTS_LOCALHOST_NAME) {
         return err;
     }
     if (hostname == SRS_CONSTS_LOCALHOST || hostname == SRS_CONSTS_LOOPBACK || hostname == SRS_CONSTS_LOOPBACK6) {
         return err;
     }
 
+    // Whether add domain name.
+    if (!srs_is_ipv4(hostname) && _srs_config->get_keep_api_domain()) {
+        candidate_ips.insert(hostname);
+    }
+
     // Try to parse the domain name if not IP.
-    int family = 0;
-    string ip = srs_dns_resolve(hostname, family);
-    if (ip.empty() || ip == SRS_CONSTS_LOCALHOST || ip == SRS_CONSTS_LOOPBACK || ip == SRS_CONSTS_LOOPBACK6) {
-        return err;
+    if (!srs_is_ipv4(hostname) && _srs_config->get_resolve_api_domain()) {
+        int family = 0;
+        string ip = srs_dns_resolve(hostname, family);
+        if (ip.empty() || ip == SRS_CONSTS_LOCALHOST || ip == SRS_CONSTS_LOOPBACK || ip == SRS_CONSTS_LOOPBACK6) {
+            return err;
+        }
+
+        // Try to add the API server ip as candidates.
+        candidate_ips.insert(ip);
     }
 
-    // Try to add the API server ip as candidates.
-    candidate_ips.insert(ip);
+    // If hostname is IP, use it.
+    if (srs_is_ipv4(hostname)) {
+        candidate_ips.insert(hostname);
+    }
 
     return err;
 }
@@ -195,8 +202,8 @@ static set<string> discover_candidates(SrsRtcUserConfig* ruc)
     }
 
     // Try to discover from api of request, if api_as_candidates enabled.
-    if ((err = api_server_as_candidates(ruc->api_, candidate_ips)) != srs_success) {
-        srs_warn("ignore discovering ip from api %s, err %s", ruc->api_.c_str(), srs_error_summary(err).c_str());
+    if ((err = api_server_as_candidates(ruc->req_->host, candidate_ips)) != srs_success) {
+        srs_warn("ignore discovering ip from api %s, err %s", ruc->req_->host.c_str(), srs_error_summary(err).c_str());
         srs_freep(err);
     }
 
@@ -207,29 +214,32 @@ static set<string> discover_candidates(SrsRtcUserConfig* ruc)
         return candidate_ips;
     }
 
-    // Discover from local network interface addresses.
+    // All automatically detected IP list.
     vector<SrsIPAddress*>& ips = srs_get_local_ips();
     if (ips.empty()) {
         return candidate_ips;
     }
 
-    // We try to find the best match candidates, no loopback.
-    string family = _srs_config->get_rtc_server_ip_family();
-    for (int i = 0; i < (int)ips.size(); ++i) {
-        SrsIPAddress* ip = ips[i];
-        if (ip->is_loopback) {
-            continue;
-        }
-
-        if (family == "ipv4" && !ip->is_ipv4) {
-            continue;
-        }
-        if (family == "ipv6" && ip->is_ipv4) {
-            continue;
+    // Discover from local network interface addresses.
+    if (_srs_config->get_use_auto_detect_network_ip()) {
+        // We try to find the best match candidates, no loopback.
+        string family = _srs_config->get_rtc_server_ip_family();
+        for (int i = 0; i < (int) ips.size(); ++i) {
+            SrsIPAddress* ip = ips[i];
+            if (ip->is_loopback) {
+                continue;
+            }
+
+            if (family == "ipv4" && !ip->is_ipv4) {
+                continue;
+            }
+            if (family == "ipv6" && ip->is_ipv4) {
+                continue;
+            }
+
+            candidate_ips.insert(ip->ip);
+            srs_trace("Best matched ip=%s, ifname=%s", ip->ip.c_str(), ip->ifname.c_str());
         }
-
-        candidate_ips.insert(ip->ip);
-        srs_trace("Best matched ip=%s, ifname=%s", ip->ip.c_str(), ip->ifname.c_str());
     }
 
     if (!candidate_ips.empty()) {
@@ -248,7 +258,7 @@ static set<string> discover_candidates(SrsRtcUserConfig* ruc)
         return candidate_ips;
     }
 
-    // We use the first one.
+    // We use the first one, to make sure there will be at least one CANDIDATE.
     if (candidate_ips.empty()) {
         SrsIPAddress* ip = ips[0];
         candidate_ips.insert(ip->ip);
@@ -280,6 +290,7 @@ SrsRtcUserConfig::SrsRtcUserConfig()
     req_ = new SrsRequest();
     publish_ = false;
     dtls_ = srtp_ = true;
+    audio_before_video_ = false;
 }
 
 SrsRtcUserConfig::~SrsRtcUserConfig()
@@ -495,6 +506,16 @@ srs_error_t SrsRtcServer::listen_api()
         return srs_error_wrap(err, "handle publish");
     }
 
+    // Generally, WHIP is a publishing protocol, but it can be also used as playing.
+    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.
+    if ((err = http_api_mux->handle("/rtc/v1/whip-play/", new SrsGoApiRtcWhip(this))) != srs_success) {
+        return srs_error_wrap(err, "handle whip play");
+    }
+
 #ifdef SRS_SIMULATOR
     if ((err = http_api_mux->handle("/rtc/v1/nack/", new SrsGoApiRtcNACK(this))) != srs_success) {
         return srs_error_wrap(err, "handle nack");
@@ -577,7 +598,7 @@ srs_error_t SrsRtcServer::do_create_session(SrsRtcUserConfig* ruc, SrsSdp& local
         set<string> candidates = discover_candidates(ruc);
         for (set<string>::iterator it = candidates.begin(); it != candidates.end(); ++it) {
             string hostname; int port = listen_port;
-            srs_parse_hostport(*it, hostname,port);
+            srs_parse_hostport(*it, hostname, port);
             local_sdp.add_candidate(hostname, port, "host");
         }
 

trunk/src/app/srs_app_rtc_server.hpp

@@ -75,16 +75,25 @@ class SrsRtcUserConfig
 {
 public:
     // Original variables from API.
+    std::string remote_sdp_str_;
     SrsSdp remote_sdp_;
     std::string eip_;
     std::string codec_;
     std::string api_;
 
+    // Session data.
+    std::string local_sdp_str_;
+    std::string session_id_;
+
     // Generated data.
     SrsRequest* req_;
     bool publish_;
     bool dtls_;
     bool srtp_;
+
+    // The order of audio and video, or whether audio is before video. Please make sure the order is match for offer and
+    // answer, or client might fail at setRemoveDescription(answer). See https://github.com/ossrs/srs/issues/3179
+    bool audio_before_video_;
 public:
     SrsRtcUserConfig();
     virtual ~SrsRtcUserConfig();