Fix RTSP SETUP crash, HLS ctx check bypass, log fd and timer bugs; make core classes testable. v8.0.44

internal/version/version.go

@@ -15,7 +15,7 @@ func VersionMinor() int {
 }
 
 func VersionRevision() int {
-	return 41
+	return 44
 }
 
 func Version() string {

skills/srs-develop/references/integration-tests.md

@@ -24,15 +24,15 @@ Run focused and component-native tests first, then run every command below seque
    ```bash
    bash skills/srs-develop/scripts/proxy-e2e-redis-test.sh
    ```
-6. RTMP publish with RTMP, HTTP-FLV, and HLS playback verification; WHEP remains a placeholder:
+6. RTMP publish with RTMP, HTTP-FLV, HLS, and WHEP playback verification:
    ```bash
    bash skills/srs-develop/scripts/proxy-e2e-transmux-test.sh
    ```
-7. SRT publish with SRT, RTMP, HTTP-FLV, and HLS playback verification; WHEP remains a placeholder:
+7. SRT publish with SRT, RTMP, HTTP-FLV, HLS, and WHEP playback verification:
    ```bash
    bash skills/srs-develop/scripts/proxy-e2e-srt-test.sh
    ```
-8. WHIP publish with RTMP, HTTP-FLV, and HLS playback verification; WHEP remains a placeholder:
+8. WHIP publish with RTMP, HTTP-FLV, HLS, and WHEP playback verification:
    ```bash
    bash skills/srs-develop/scripts/proxy-e2e-whip-test.sh
    ```
@@ -41,7 +41,7 @@ Run focused and component-native tests first, then run every command below seque
    bash skills/srs-develop/scripts/proxy-e2e-bearer-auth-test.sh
    ```
 
-The SRT test requires an FFmpeg build with libsrt. The WHIP test requires the `whip` muxer and OpenSSL. Both scripts automatically run `skills/srs-develop/scripts/setup-ffmpeg-with-whip.sh` on macOS when no suitable FFmpeg is available. If an environmental dependency is unavailable, run the script, preserve its exact result, and report the blocked coverage instead of claiming full verification.
+The SRT test requires an FFmpeg build with libsrt. The WHIP test requires the `whip` muxer and OpenSSL. Both scripts automatically run `skills/srs-develop/scripts/setup-ffmpeg-with-whip.sh` on macOS when no suitable FFmpeg is available. If an environmental dependency is unavailable, run the script, preserve its exact result, and report the blocked coverage instead of claiming full verification. The transmux, SRT, and WHIP tests play WHEP with `tools/pion-whep`, because FFmpeg has no WHEP demuxer; they build it with `go` when the binary is missing or stale.
 
 Run feature-specific bundled tests in addition to this matrix when the routed workflow or the touched area requires them:
 

skills/srs-develop/scripts/proxy-e2e-srt-test.sh

@@ -6,7 +6,7 @@
 #   - RTMP play (via srt_to_rtmp on origin)
 #   - HTTP-FLV (HTTP remux of the bridged RTMP)
 #   - HLS (m3u8 + TS segments)
-#   - WebRTC WHEP (placeholder only, not actually verified here)
+#   - WebRTC WHEP (via rtmp_to_rtc on origin, played with tools/pion-whep)
 set -e
 
 SCRIPT_DIR="$(cd -P "$(dirname "$0")" && pwd)"
@@ -100,6 +100,50 @@ probe_has_audio_video() {
   fi
 }
 
+# Play over WHEP with tools/pion-whep, because FFmpeg has no WHEP demuxer, and
+# require both video and audio RTP packets. The tool is rebuilt when its binary
+# is missing or older than its sources.
+verify_whep_playback() {
+  local url="$1"
+  local log="$2"
+  local tool_dir="$WORKSPACE/tools/pion-whep"
+  local tool_bin="$tool_dir/objs/pion-whep"
+  local summary video audio
+
+  if [[ ! -x "$tool_bin" ]] || [[ -n "$(find "$tool_dir" -maxdepth 1 \( -name '*.go' -o -name 'go.mod' -o -name 'go.sum' \) -newer "$tool_bin")" ]]; then
+    echo "Building pion-whep: $tool_bin"
+    (cd "$tool_dir" && mkdir -p objs && go build -o objs/pion-whep .)
+  fi
+
+  echo "Verifying WHEP playback: $url"
+  if ! "$tool_bin" -hide_banner -f whep -i "$url" -t 5 -f null - >"$log" 2>&1; then
+    echo "FAIL: WHEP playback failed. pion-whep log:" >&2
+    cat "$log" >&2
+    exit 1
+  fi
+
+  summary="$(grep -a "^Received video=" "$log" | tail -1)"
+  video="$(echo "$summary" | sed -n 's/^Received video=\([0-9]*\),.*/\1/p')"
+  audio="$(echo "$summary" | sed -n 's/^Received video=[0-9]*, audio=\([0-9]*\) .*/\1/p')"
+  echo "pion-whep: $summary"
+
+  if [[ -n "$video" && "$video" -gt 0 ]]; then
+    echo "PASS: WHEP video packets received."
+  else
+    echo "FAIL: WHEP no video packets received." >&2
+    cat "$log" >&2
+    exit 1
+  fi
+
+  if [[ -n "$audio" && "$audio" -gt 0 ]]; then
+    echo "PASS: WHEP audio packets received."
+  else
+    echo "FAIL: WHEP no audio packets received." >&2
+    cat "$log" >&2
+    exit 1
+  fi
+}
+
 wait_for_hls_playlist() {
   local url="$1"
   local deadline=45
@@ -133,6 +177,10 @@ if ! command -v curl &>/dev/null; then
   echo "Error: curl not found in PATH" >&2
   exit 1
 fi
+if ! command -v go &>/dev/null; then
+  echo "Error: go not found in PATH" >&2
+  exit 1
+fi
 
 # SRT URLs need libsrt compiled into ffmpeg/ffprobe. The default Homebrew
 # ffmpeg formula does NOT include libsrt. Resolution order:
@@ -233,7 +281,10 @@ echo "Proxy started."
 echo "=== Step 4: Starting SRS origin ==="
 ulimit -n 10000 2>/dev/null || true
 cd "$WORKSPACE/trunk"
-./objs/srs -c conf/origin1-for-proxy.conf >/tmp/srs-origin-srt-e2e.log 2>&1 &
+# The proxy rewrites only the port of the WebRTC candidate, so the origin must
+# advertise an IP the WHEP player can reach.
+env CANDIDATE="127.0.0.1" \
+    ./objs/srs -c conf/origin1-for-proxy.conf >/tmp/srs-origin-srt-e2e.log 2>&1 &
 ORIGIN_PID=$!
 echo "SRS origin PID: $ORIGIN_PID"
 
@@ -285,11 +336,10 @@ HLS_URL="http://localhost:$PROXY_HTTP_SERVER_PORT/$STREAM_URL.m3u8"
 wait_for_hls_playlist "$HLS_URL"
 probe_has_audio_video "HLS" "$HLS_URL"
 
-# --- Step 10: WebRTC WHEP playback (placeholder) ---
-echo "=== Step 10: WebRTC WHEP playback (placeholder) ==="
-echo "SKIP: WebRTC WHEP playback is not verified by this script."
-echo "      The origin has rtmp_to_rtc enabled, so SRT->RTMP->RTC should work end-to-end,"
-echo "      but actual playback verification is intentionally left as a TODO here."
+# --- Step 10: Verify WebRTC WHEP playback (srt_to_rtmp + rtmp_to_rtc) ---
+echo "=== Step 10: Verifying WebRTC WHEP playback via proxy ==="
+verify_whep_playback "http://localhost:$PROXY_HTTP_API_PORT/rtc/v1/whep/?app=live&stream=${STREAM_URL#live/}" \
+  /tmp/srs-pion-whep-srt-e2e.log
 
 echo ""
 echo "=== E2E SRT Proxy Test PASSED ==="

skills/srs-develop/scripts/proxy-e2e-transmux-test.sh

@@ -1,8 +1,8 @@
 #!/bin/bash
 # E2E test for RTMP-to-multiple-protocol transmuxing through the proxy:
 # starts one proxy with memory load balancer + one SRS origin, publishes one
-# RTMP stream, then verifies RTMP, HTTP-FLV, and HLS playback through the
-# proxy. WebRTC WHEP verification is intentionally a placeholder (not run).
+# RTMP stream, then verifies RTMP, HTTP-FLV, HLS, and WebRTC WHEP playback
+# through the proxy. WHEP is played with tools/pion-whep.
 set -e
 
 SCRIPT_DIR="$(cd -P "$(dirname "$0")" && pwd)"
@@ -92,6 +92,50 @@ probe_has_audio_video() {
   fi
 }
 
+# Play over WHEP with tools/pion-whep, because FFmpeg has no WHEP demuxer, and
+# require both video and audio RTP packets. The tool is rebuilt when its binary
+# is missing or older than its sources.
+verify_whep_playback() {
+  local url="$1"
+  local log="$2"
+  local tool_dir="$WORKSPACE/tools/pion-whep"
+  local tool_bin="$tool_dir/objs/pion-whep"
+  local summary video audio
+
+  if [[ ! -x "$tool_bin" ]] || [[ -n "$(find "$tool_dir" -maxdepth 1 \( -name '*.go' -o -name 'go.mod' -o -name 'go.sum' \) -newer "$tool_bin")" ]]; then
+    echo "Building pion-whep: $tool_bin"
+    (cd "$tool_dir" && mkdir -p objs && go build -o objs/pion-whep .)
+  fi
+
+  echo "Verifying WHEP playback: $url"
+  if ! "$tool_bin" -hide_banner -f whep -i "$url" -t 5 -f null - >"$log" 2>&1; then
+    echo "FAIL: WHEP playback failed. pion-whep log:" >&2
+    cat "$log" >&2
+    exit 1
+  fi
+
+  summary="$(grep -a "^Received video=" "$log" | tail -1)"
+  video="$(echo "$summary" | sed -n 's/^Received video=\([0-9]*\),.*/\1/p')"
+  audio="$(echo "$summary" | sed -n 's/^Received video=[0-9]*, audio=\([0-9]*\) .*/\1/p')"
+  echo "pion-whep: $summary"
+
+  if [[ -n "$video" && "$video" -gt 0 ]]; then
+    echo "PASS: WHEP video packets received."
+  else
+    echo "FAIL: WHEP no video packets received." >&2
+    cat "$log" >&2
+    exit 1
+  fi
+
+  if [[ -n "$audio" && "$audio" -gt 0 ]]; then
+    echo "PASS: WHEP audio packets received."
+  else
+    echo "FAIL: WHEP no audio packets received." >&2
+    cat "$log" >&2
+    exit 1
+  fi
+}
+
 wait_for_hls_playlist() {
   local url="$1"
   local deadline=45
@@ -133,6 +177,10 @@ if ! command -v curl &>/dev/null; then
   echo "Error: curl not found in PATH" >&2
   exit 1
 fi
+if ! command -v go &>/dev/null; then
+  echo "Error: go not found in PATH" >&2
+  exit 1
+fi
 
 # --- Step 0: Clean up stale state ---
 rm -f "$WORKSPACE/trunk/objs/origin1.pid"
@@ -184,7 +232,10 @@ echo "Proxy started."
 echo "=== Step 4: Starting SRS origin ==="
 ulimit -n 10000 2>/dev/null || true
 cd "$WORKSPACE/trunk"
-./objs/srs -c conf/origin1-for-proxy.conf >/tmp/srs-origin-transmux-e2e.log 2>&1 &
+# The proxy rewrites only the port of the WebRTC candidate, so the origin must
+# advertise an IP the WHEP player can reach.
+env CANDIDATE="127.0.0.1" \
+    ./objs/srs -c conf/origin1-for-proxy.conf >/tmp/srs-origin-transmux-e2e.log 2>&1 &
 ORIGIN_PID=$!
 echo "SRS origin PID: $ORIGIN_PID"
 
@@ -230,11 +281,10 @@ HLS_URL="http://localhost:$PROXY_HTTP_SERVER_PORT/$STREAM_URL.m3u8"
 wait_for_hls_playlist "$HLS_URL"
 probe_has_audio_video "HLS" "$HLS_URL"
 
-# --- Step 9: WebRTC WHEP playback (placeholder) ---
-echo "=== Step 9: WebRTC WHEP playback (placeholder) ==="
-echo "SKIP: WebRTC WHEP playback is not verified by this script."
-echo "      The origin has rtmp_to_rtc enabled, so RTMP->RTC should work end-to-end,"
-echo "      but actual playback verification is intentionally left as a TODO here."
+# --- Step 9: Verify WebRTC WHEP playback (rtmp_to_rtc) ---
+echo "=== Step 9: Verifying WebRTC WHEP playback via proxy ==="
+verify_whep_playback "http://localhost:$PROXY_HTTP_API_PORT/rtc/v1/whep/?app=live&stream=${STREAM_URL#live/}" \
+  /tmp/srs-pion-whep-transmux-e2e.log
 
 echo ""
 echo "NOTE: RTSP is not tested here because the Go proxy currently has no RTSP listener."

skills/srs-develop/scripts/proxy-e2e-whip-test.sh

@@ -6,7 +6,7 @@
 #   - RTMP play (via rtc_to_rtmp on origin)
 #   - HTTP-FLV (HTTP remux of the bridged RTMP)
 #   - HLS (m3u8 + TS segments)
-#   - WebRTC WHEP (placeholder only, not actually verified here)
+#   - WebRTC WHEP (RTC passthrough, played with tools/pion-whep)
 set -e
 
 SCRIPT_DIR="$(cd -P "$(dirname "$0")" && pwd)"
@@ -107,6 +107,50 @@ probe_has_audio_video() {
   fi
 }
 
+# Play over WHEP with tools/pion-whep, because FFmpeg has no WHEP demuxer, and
+# require both video and audio RTP packets. The tool is rebuilt when its binary
+# is missing or older than its sources.
+verify_whep_playback() {
+  local url="$1"
+  local log="$2"
+  local tool_dir="$WORKSPACE/tools/pion-whep"
+  local tool_bin="$tool_dir/objs/pion-whep"
+  local summary video audio
+
+  if [[ ! -x "$tool_bin" ]] || [[ -n "$(find "$tool_dir" -maxdepth 1 \( -name '*.go' -o -name 'go.mod' -o -name 'go.sum' \) -newer "$tool_bin")" ]]; then
+    echo "Building pion-whep: $tool_bin"
+    (cd "$tool_dir" && mkdir -p objs && go build -o objs/pion-whep .)
+  fi
+
+  echo "Verifying WHEP playback: $url"
+  if ! "$tool_bin" -hide_banner -f whep -i "$url" -t 5 -f null - >"$log" 2>&1; then
+    echo "FAIL: WHEP playback failed. pion-whep log:" >&2
+    cat "$log" >&2
+    exit 1
+  fi
+
+  summary="$(grep -a "^Received video=" "$log" | tail -1)"
+  video="$(echo "$summary" | sed -n 's/^Received video=\([0-9]*\),.*/\1/p')"
+  audio="$(echo "$summary" | sed -n 's/^Received video=[0-9]*, audio=\([0-9]*\) .*/\1/p')"
+  echo "pion-whep: $summary"
+
+  if [[ -n "$video" && "$video" -gt 0 ]]; then
+    echo "PASS: WHEP video packets received."
+  else
+    echo "FAIL: WHEP no video packets received." >&2
+    cat "$log" >&2
+    exit 1
+  fi
+
+  if [[ -n "$audio" && "$audio" -gt 0 ]]; then
+    echo "PASS: WHEP audio packets received."
+  else
+    echo "FAIL: WHEP no audio packets received." >&2
+    cat "$log" >&2
+    exit 1
+  fi
+}
+
 wait_for_hls_playlist() {
   local url="$1"
   local deadline=60
@@ -183,6 +227,10 @@ if ! command -v curl &>/dev/null; then
   echo "Error: curl not found in PATH" >&2
   exit 1
 fi
+if ! command -v go &>/dev/null; then
+  echo "Error: go not found in PATH" >&2
+  exit 1
+fi
 
 # WHIP needs an ffmpeg with the `whip` muxer (added in ffmpeg 7.1, requires
 # --enable-openssl at build time for DTLS-SRTP). Neither vanilla brew nor the
@@ -340,11 +388,10 @@ wait_for_hls_playlist "$HLS_URL"
 wait_for_hls_to_skip_first_segment "$HLS_URL"
 probe_has_audio_video "HLS" "$HLS_URL"
 
-# --- Step 9: WebRTC WHEP playback (placeholder) ---
-echo "=== Step 9: WebRTC WHEP playback (placeholder) ==="
-echo "SKIP: WebRTC WHEP playback is not verified by this script."
-echo "      The origin has rtmp_to_rtc enabled, so WHIP->RTMP->RTC should work end-to-end,"
-echo "      but actual playback verification is intentionally left as a TODO here."
+# --- Step 9: Verify WebRTC WHEP playback (RTC passthrough) ---
+echo "=== Step 9: Verifying WebRTC WHEP playback via proxy ==="
+verify_whep_playback "http://localhost:$PROXY_HTTP_API_PORT/rtc/v1/whep/?app=live&stream=$STREAM_NAME" \
+  /tmp/srs-pion-whep-whip-e2e.log
 
 echo ""
 echo "=== E2E WHIP Proxy Test PASSED ==="

trunk/configure

@@ -388,7 +388,8 @@ if [[ $SRS_UTEST == YES ]]; then
     MODULE_FILES+=("srs_utest_ai01" "srs_utest_ai02" "srs_utest_ai03" "srs_utest_ai04" "srs_utest_ai05"
         "srs_utest_ai06" "srs_utest_ai07" "srs_utest_ai08" "srs_utest_ai09" "srs_utest_ai10" "srs_utest_ai11"
         "srs_utest_ai12" "srs_utest_ai13" "srs_utest_ai14" "srs_utest_ai15" "srs_utest_ai16" "srs_utest_ai17"
-        "srs_utest_ai18" "srs_utest_ai19" "srs_utest_ai20" "srs_utest_ai24" "srs_utest_ai25" "srs_utest_ai26" "srs_utest_ai27")
+        "srs_utest_ai18" "srs_utest_ai19" "srs_utest_ai20" "srs_utest_ai24" "srs_utest_ai25" "srs_utest_ai26" "srs_utest_ai27" 
+        "srs_utest_ai28" "srs_utest_ai29" "srs_utest_ai30" "srs_utest_ai31" "srs_utest_ai32")
     if [[ $SRS_GB28181 == YES ]]; then
         MODULE_FILES+=("srs_utest_manual_gb28181" "srs_utest_ai23")
     fi

trunk/doc/CHANGELOG.md

@@ -7,6 +7,9 @@ The changelog for SRS.
 <a name="v8-changes"></a>
 
 ## SRS 8.0 Changelog
+* v8.0, 2026-09-22, Timer: Notify every subscriber when one unsubscribes during the round. v8.0.44
+* v8.0, 2026-09-22, Log: Clear the log file descriptor when the logger reopens it. v8.0.43
+* v8.0, 2026-09-22, HTTP: Clamp the VOD MP4 range end to the last byte of the file. v8.0.42
 * v8.0, 2026-09-21, Merge [#4746](https://github.com/ossrs/srs/pull/4746): WebRTC: Support RFC 4588 RTX retransmission with SSRC multiplexing, preferred by nack_prefer_rtx. v8.0.41 (#4746)
 * v8.0, 2026-09-21, RTC: Reset the play track cache when the publisher republishes with new SSRCs. v8.0.40
 * v8.0, 2026-09-21, RTC: Resolve the publish and play tracks once per SSRC through the fast cache. v8.0.39

trunk/src/app/srs_app_config.hpp

@@ -352,6 +352,9 @@ class ISrsAppConfig : public ISrsConfig
     virtual std::string get_https_stream_ssl_cert() = 0;
     virtual std::string get_http_stream_dir() = 0;
     virtual bool get_http_stream_crossdomain() = 0;
+    virtual bool get_vhost_http_enabled(std::string vhost) = 0;
+    virtual std::string get_vhost_http_mount(std::string vhost) = 0;
+    virtual std::string get_vhost_http_dir(std::string vhost) = 0;
 
 public:
     // WebRTC config
@@ -377,8 +380,21 @@ class ISrsAppConfig : public ISrsConfig
 public:
     // SRT config
     virtual std::vector<std::string> get_srt_listens() = 0;
+    virtual int64_t get_srto_maxbw() = 0;
+    virtual int get_srto_mss() = 0;
+    virtual bool get_srto_tsbpdmode() = 0;
+    virtual int get_srto_latency() = 0;
+    virtual int get_srto_recv_latency() = 0;
+    virtual int get_srto_peer_latency() = 0;
+    virtual bool get_srto_tlpktdrop() = 0;
+    virtual srs_utime_t get_srto_conntimeout() = 0;
     // Get the srt SRTO_PEERIDLETIMEO, peer idle timeout, default is 10000ms.
     virtual srs_utime_t get_srto_peeridletimeout() = 0;
+    virtual int get_srto_sendbuf() = 0;
+    virtual int get_srto_recvbuf() = 0;
+    virtual int get_srto_payloadsize() = 0;
+    virtual std::string get_srto_passphrase() = 0;
+    virtual int get_srto_pbkeylen() = 0;
 
 public:
     // Stream caster config
@@ -640,6 +656,13 @@ class ISrsAppConfig : public ISrsConfig
     virtual std::string get_engine_output(SrsConfDirective *conf) = 0;
     virtual bool get_security_enabled(std::string vhost) = 0;
     virtual SrsConfDirective *get_security_rules(std::string vhost) = 0;
+    // Whether write log to file, otherwise to console.
+    virtual bool get_log_tank_file() = 0;
+    // The file to write log to, empty if not configured.
+    virtual std::string get_log_file() = 0;
+    virtual std::string get_log_level() = 0;
+    virtual std::string get_log_level_v2() = 0;
+    virtual bool get_utc_time() = 0;
 };
 
 // The config service provider.