feat: add WebRTC streaming via str0m + portal session persistence

- Add src/webrtc.rs: HTTP signaling server + str0m Sans-IO WebRTC transport
  with H.264 Annex-B → RTP packetization and key-frame request handling
- avhw: introduce FrameOutput enum (Muxer | Channel) so SwEncState can
  output to either MP4 muxer or crossbeam channel for WebRTC
- cap_portal: support portal session restore tokens (PersistMode::ExplicitlyRevoked)
  to skip re-authorization dialog; add --no-persist flag to force fresh dialog
- args: make --output optional when --port is used for WebRTC mode
- state_portal: integrate WebRTC pipeline (encoder channel → RTP forwarding)
  with shorter GOP for WebRTC (fps/2, min 10)
- main: redirect tracing to stderr; validate --output or --port required
- Add dependencies: str0m 0.20, serde_json 1, dirs 6

Cargo.toml

@@ -25,3 +25,6 @@ tokio = { version = "1", features = ["rt"] }
 pipewire = { version = "0.9", features = ["v0_3_45"] }
 libspa = "0.9"
 crossbeam-channel = "0.5"
+str0m = "0.20"
+serde_json = "1"
+dirs = "6"

src/args.rs

@@ -3,9 +3,9 @@ use clap::Parser;
 #[derive(Parser, Debug, Clone)]
 #[command(name = "wl-webrtc", about = "Wayland screen capture and encoding tool")]
 pub struct Args {
-    /// Output file path (e.g., output.mp4, output.mkv)
+    /// Output file path (e.g., output.mp4, output.mkv). Optional when using --port for WebRTC mode
     #[arg(short, long)]
-    pub output: String,
+    pub output: Option<String>,
 
     /// Wayland output name to capture
     #[arg(long)]
@@ -43,7 +43,11 @@ pub struct Args {
     #[arg(long)]
     pub backend: Option<String>,
 
-    /// Port for WebTransport server (Phase 2, unused in MVP)
+    /// Port for WebRTC HTTP signaling server; 0 keeps MP4 file output mode
     #[arg(long, default_value_t = 0)]
     pub port: u16,
+
+    /// Force re-authorization dialog (ignore saved portal restore token)
+    #[arg(long)]
+    pub no_persist: bool,
 }

src/avhw.rs

@@ -596,13 +596,18 @@ impl EncState {
 // SwEncState - VAAPI GPU downscale + software H.264 encode
 // ---------------------------------------------------------------------------
 
+pub enum FrameOutput {
+    Muxer(ff::format::context::Output),
+    Channel(crossbeam_channel::Sender<Vec<u8>>),
+}
+
 pub struct SwEncState {
     hw_dev: AvHwDevCtx,
     frames_rgb: AvHwFrameCtx,
     filter_graph: ff::filter::Graph,
     sws_ctx: *mut ffi::SwsContext,
     enc_video: ff::codec::encoder::video::Video,
-    octx: ff::format::context::Output,
+    output: Option<FrameOutput>,
     yuv_frame: *mut ffi::AVFrame,
     starting_timestamp: Option<i64>,
     frames_written: bool,
@@ -651,7 +656,52 @@ impl SwEncState {
             filter_graph,
             sws_ctx,
             enc_video,
-            octx,
+            output: Some(FrameOutput::Muxer(octx)),
+            yuv_frame,
+            starting_timestamp: None,
+            frames_written: false,
+        })
+    }
+
+    #[allow(clippy::too_many_arguments)]
+    pub fn new_webrtc(
+        drm_device: &Path,
+        width: u32,
+        height: u32,
+        enc_width: u32,
+        enc_height: u32,
+        fps: u32,
+        bitrate: u64,
+        gop_size: u32,
+        tx: crossbeam_channel::Sender<Vec<u8>>,
+    ) -> Result<Self> {
+        tracing::info!(
+            "SwEncState::new_webrtc: GPU downscale {width}x{height} BGRA -> {enc_width}x{enc_height} NV12, software H.264 -> WebRTC"
+        );
+
+        let hw_dev = AvHwDevCtx::new_vaapi(drm_device)?;
+        let frames_rgb =
+            AvHwFrameCtx::for_capture(&hw_dev, width, height, ff::format::Pixel::BGRA)?;
+        let filter_graph = build_swenc_filter_graph(
+            &hw_dev,
+            &frames_rgb,
+            width,
+            height,
+            enc_width,
+            enc_height,
+            fps,
+        )?;
+        let sws_ctx = create_nv12_to_yuv420p_sws(enc_width, enc_height)?;
+        let enc_video = create_software_h264_encoder(enc_width, enc_height, fps, bitrate, gop_size)?;
+        let yuv_frame = alloc_yuv420p_frame(enc_width, enc_height)?;
+
+        Ok(Self {
+            hw_dev,
+            frames_rgb,
+            filter_graph,
+            sws_ctx,
+            enc_video,
+            output: Some(FrameOutput::Channel(tx)),
             yuv_frame,
             starting_timestamp: None,
             frames_written: false,
@@ -704,7 +754,6 @@ impl SwEncState {
             }
         }
 
-        // SAFETY: Sending a null frame flushes the encoder without transferring ownership.
         unsafe {
             let ret = ffi::avcodec_send_frame(self.enc_video.as_mut_ptr(), ptr::null());
             if ret < 0 && ret != ffi::AVERROR_EOF {
@@ -715,9 +764,10 @@ impl SwEncState {
         self.drain_encoder(start_ts)?;
 
         if self.frames_written {
-            self.octx
-                .write_trailer()
-                .map_err(|e| anyhow::anyhow!("Failed to write trailer: {e}"))?;
+            if let Some(FrameOutput::Muxer(ref mut octx)) = self.output {
+                octx.write_trailer()
+                    .map_err(|e| anyhow::anyhow!("Failed to write trailer: {e}"))?;
+            }
         }
 
         Ok(())
@@ -793,25 +843,39 @@ impl SwEncState {
                 bail!("avcodec_receive_packet failed: error {ret}");
             }
 
-            let enc_tb = self.enc_video.time_base();
-            let stream_tb = unsafe {
-                let streams = (*self.octx.as_ptr()).streams;
-                let st = *streams.add(0);
-                ff::Rational::from((*st).time_base)
-            };
-            pkt.rescale_ts(enc_tb, stream_tb);
+            match self.output {
+                Some(FrameOutput::Muxer(ref mut octx)) => {
+                    let enc_tb = self.enc_video.time_base();
+                    let stream_tb = unsafe {
+                        let streams = (*octx.as_ptr()).streams;
+                        let st = *streams.add(0);
+                        ff::Rational::from((*st).time_base)
+                    };
+                    pkt.rescale_ts(enc_tb, stream_tb);
 
-            if let Some(pts) = pkt.pts() {
-                pkt.set_pts(Some(pts - start_ts));
-            }
-            if let Some(dts) = pkt.dts() {
-                pkt.set_dts(Some(dts - start_ts));
-            }
+                    if let Some(pts) = pkt.pts() {
+                        pkt.set_pts(Some(pts - start_ts));
+                    }
+                    if let Some(dts) = pkt.dts() {
+                        pkt.set_dts(Some(dts - start_ts));
+                    }
 
-            pkt.set_stream(0);
-            pkt.write_interleaved(&mut self.octx)
-                .map_err(|e| anyhow::anyhow!("Failed to write packet: {e}"))?;
-            self.frames_written = true;
+                    pkt.set_stream(0);
+                    pkt.write_interleaved(octx)
+                        .map_err(|e| anyhow::anyhow!("Failed to write packet: {e}"))?;
+                    self.frames_written = true;
+                }
+                Some(FrameOutput::Channel(ref tx)) => {
+                    let data: &[u8] = unsafe {
+                        std::slice::from_raw_parts(
+                            (*pkt.as_mut_ptr()).data,
+                            (*pkt.as_mut_ptr()).size as usize,
+                        )
+                    };
+                    let _ = tx.send(data.to_vec());
+                }
+                None => {}
+            }
         }
         Ok(())
     }
@@ -1115,6 +1179,54 @@ fn create_software_h264_muxer(
     Ok((enc_video, octx))
 }
 
+fn create_software_h264_encoder(
+    width: u32,
+    height: u32,
+    fps: u32,
+    bitrate: u64,
+    gop_size: u32,
+) -> Result<ff::codec::encoder::video::Video> {
+    let codec = ff::encoder::find_by_name("libx264")
+        .or_else(|| ff::encoder::find_by_name("libopenh264"))
+        .ok_or_else(|| anyhow::anyhow!("No H.264 software encoder found"))?;
+    let codec_name = codec.name().to_string();
+
+    let mut enc = {
+        let ctx = ff::codec::Context::new_with_codec(codec);
+        ctx.encoder().video()?
+    };
+    enc.set_width(width);
+    enc.set_height(height);
+    enc.set_format(ff::format::Pixel::YUV420P);
+    enc.set_bit_rate(bitrate as usize);
+    enc.set_gop(gop_size);
+    enc.set_time_base(ff::Rational::new(1, fps as i32));
+    enc.set_max_b_frames(0);
+
+    if codec_name == "libx264" {
+        unsafe {
+            let key = CString::new("preset").unwrap();
+            let val = CString::new("ultrafast").unwrap();
+            ffi::av_opt_set((*enc.as_mut_ptr()).priv_data, key.as_ptr(), val.as_ptr(), 0);
+            let key = CString::new("tune").unwrap();
+            let val = CString::new("zerolatency").unwrap();
+            ffi::av_opt_set((*enc.as_mut_ptr()).priv_data, key.as_ptr(), val.as_ptr(), 0);
+            let key = CString::new("threads").unwrap();
+            let val = CString::new("6").unwrap();
+            ffi::av_opt_set((*enc.as_mut_ptr()).priv_data, key.as_ptr(), val.as_ptr(), 0);
+            let key = CString::new("x264opts").unwrap();
+            let val = CString::new("repeat_headers=1").unwrap();
+            ffi::av_opt_set((*enc.as_mut_ptr()).priv_data, key.as_ptr(), val.as_ptr(), 0);
+        }
+    }
+
+    let opened = enc
+        .open()
+        .map_err(|e| anyhow::anyhow!("Failed to open {codec_name} encoder: {e}"))?;
+    tracing::info!("WebRTC encoder: {codec_name} {width}x{height} @ {fps}fps {bitrate}bps");
+    Ok(opened.0)
+}
+
 // ---------------------------------------------------------------------------
 // Filter graph (inline)
 // ---------------------------------------------------------------------------

src/cap_portal.rs

@@ -12,6 +12,7 @@
 //   - crossbeam-channel: 高性能有界通道，用于线程间帧传递
 
 use std::os::fd::{AsRawFd, FromRawFd, OwnedFd};
+use std::path::PathBuf;
 use std::sync::atomic::{AtomicU64, Ordering};
 use std::sync::Arc;
 use std::thread::{self, JoinHandle};
@@ -98,12 +99,10 @@ impl CapPortal {
     ///   4. 创建 eventfd 对，用于线程安全的关闭信号传递
     ///   5. 启动 PipeWire 捕获线程
     pub fn new(args: &Args) -> Result<Self> {
-        // 创建独立的 Tokio 运行时，仅用于 setup_portal 中的异步 Portal D-Bus 调用
         let rt = Runtime::new()?;
 
-        // 通过 Portal 获取 PipeWire 连接 fd 和节点 ID
-        // block_on 在此处同步等待异步 Portal 调用完成
-        let (pw_fd, node_id) = rt.block_on(async { Self::setup_portal().await })?;
+        let no_persist = args.no_persist;
+        let (pw_fd, node_id) = rt.block_on(async { Self::setup_portal(no_persist).await })?;
 
         let (frame_tx, frame_rx) = bounded(16);
         let (event_tx, event_rx) = bounded(8);
@@ -172,44 +171,50 @@ impl CapPortal {
     ///   5. 打开 PipeWire 远程连接，获取文件描述符
     ///
     /// 返回 (PipeWire fd, node_id)，供 PipeWire 线程连接使用
-    async fn setup_portal() -> Result<(OwnedFd, u32)> {
+    async fn setup_portal(no_persist: bool) -> Result<(OwnedFd, u32)> {
         use ashpd::desktop::screencast::{
             CursorMode, Screencast, SelectSourcesOptions, SourceType,
         };
         use ashpd::desktop::PersistMode;
 
-        // 创建 Screencast D-Bus 代理，与桌面环境的 Portal 服务通信
         let proxy = Screencast::new()
             .await
             .map_err(|e| anyhow::anyhow!("Failed to create Screencast proxy: {e}"))?;
 
-        // 创建 ScreenCast 会话（每个会话对应一次屏幕录制请求）
         let session = proxy
             .create_session(Default::default())
             .await
             .map_err(|e| anyhow::anyhow!("Failed to create ScreenCast session: {e}"))?;
 
-        // 配置录制源选择参数:
-        //   - CursorMode::Embedded: 光标嵌入到帧数据中（而非单独的元数据）
-        //   - SourceType::Monitor: 仅捕获显示器（不捕获窗口）
-        //   - multiple: false: 不允许多源选择
-        //   - PersistMode::DoNot: 不持久化会话（每次需要重新授权）
+        let version_supported = proxy.version() >= 4;
+
+        let (persist_mode, saved_token) = if !no_persist && version_supported {
+            let token = load_restore_token();
+            if token.is_some() {
+                tracing::info!("Attempting to restore portal session with saved token");
+            }
+            (PersistMode::ExplicitlyRevoked, token)
+        } else {
+            (PersistMode::DoNot, None)
+        };
+
+        let mut options = SelectSourcesOptions::default()
+            .set_cursor_mode(CursorMode::Embedded)
+            .set_sources(ashpd::enumflags2::BitFlags::from(SourceType::Monitor))
+            .set_multiple(false)
+            .set_persist_mode(persist_mode);
+
+        if let Some(ref token) = saved_token {
+            options = options.set_restore_token(token.as_str());
+        }
+
         proxy
-            .select_sources(
-                &session,
-                SelectSourcesOptions::default()
-                    .set_cursor_mode(CursorMode::Embedded)
-                    .set_sources(ashpd::enumflags2::BitFlags::from(SourceType::Monitor))
-                    .set_multiple(false)
-                    .set_persist_mode(PersistMode::DoNot),
-            )
+            .select_sources(&session, options)
             .await
             .map_err(|e| {
-                anyhow::anyhow!("屏幕共享权限被拒绝 / Screen sharing permission denied: {e}")
+                anyhow::anyhow!("Screen sharing permission denied: {e}")
             })?;
 
-        // 启动录制会话，此时桌面环境会弹出权限确认对话框
-        // 用户确认后返回包含 PipeWire 流信息的响应
         let response = proxy
             .start(&session, None, Default::default())
             .await
@@ -217,18 +222,19 @@ impl CapPortal {
             .response()
             .map_err(|e| anyhow::anyhow!("ScreenCast response error: {e}"))?;
 
-        // 获取返回的第一个（也是唯一的）视频流
-        // 每个流对应一个 PipeWire 节点
+        if !no_persist && version_supported {
+            if let Some(new_token) = response.restore_token() {
+                save_restore_token(new_token);
+            }
+        }
+
         let stream = response
             .streams()
             .first()
             .ok_or_else(|| anyhow::anyhow!("No streams returned from ScreenCast"))?;
 
-        // 提取 PipeWire 节点 ID，用于后续连接到该节点的视频流
         let node_id = stream.pipe_wire_node_id();
 
-        // 打开 PipeWire 远程连接，获取文件描述符
-        // 这个 fd 允许直接与 PipeWire 守护进程通信
         let fd = proxy
             .open_pipe_wire_remote(&session, Default::default())
             .await
@@ -240,6 +246,30 @@ impl CapPortal {
     }
 }
 
+fn token_path() -> PathBuf {
+    let base = dirs::cache_dir()
+        .unwrap_or_else(|| PathBuf::from("/tmp"));
+    base.join("wl-webrtc").join("portal-restore-token")
+}
+
+fn load_restore_token() -> Option<String> {
+    let path = token_path();
+    let token = std::fs::read_to_string(&path).ok()?;
+    let trimmed = token.trim().to_string();
+    if trimmed.is_empty() { None } else { Some(trimmed) }
+}
+
+fn save_restore_token(token: &str) {
+    let path = token_path();
+    if let Some(parent) = path.parent() {
+        let _ = std::fs::create_dir_all(parent);
+    }
+    match std::fs::write(&path, token) {
+        Ok(()) => tracing::info!("Saved portal restore token"),
+        Err(e) => tracing::warn!("Failed to save restore token: {e}"),
+    }
+}
+
 impl Drop for CapPortal {
     /// 析构时安全关闭 PipeWire 线程
     ///

src/lib.rs

@@ -7,3 +7,4 @@ pub mod fps_limit;
 pub mod state;
 pub mod state_portal;
 pub mod transform;
+pub mod webrtc;

src/main.rs

@@ -18,6 +18,7 @@ mod fps_limit; // 帧率限制器
 mod state; // wlr-screencopy 后端的主状态机
 mod state_portal; // Portal/PipeWire 后端的主状态机
 mod transform; // 图像变换（旋转/翻转）
+mod webrtc; // WebRTC 传输（str0m Sans-IO）
 
 use crate::args::Args;
 use crate::cap_wlr_screencopy::CapWlrScreencopy;
@@ -49,6 +50,7 @@ fn main() -> Result<()> {
         } else {
             tracing::Level::INFO
         })
+        .with_writer(std::io::stderr)
         .init();
 
     tracing::info!("wl-webrtc starting");
@@ -59,6 +61,10 @@ fn main() -> Result<()> {
         anyhow::bail!("HEVC not supported in MVP. Use --codec h264");
     }
 
+    if args.output.is_none() && args.port == 0 {
+        anyhow::bail!("Either --output or --port is required");
+    }
+
     // 自动检测当前桌面环境可用的截屏后端
     // 会尝试列举 Wayland 全局对象，判断合成器是否支持 wlr-screencopy 协议
     let backend = crate::backend_detect::detect_backend(&args)?;

src/state.rs

@@ -613,7 +613,7 @@ impl<S: CaptureSource> State<S> {
             .unwrap_or_else(|| 2 * (width as u64) * (height as u64) * (fps as u64) / 100);
         let enc = match crate::avhw::create_encoder(
             &drm_path,
-            Path::new(&self.args.output),
+            Path::new(self.args.output.as_deref().expect("output required for MP4 mode")),
             width,
             height,
             fps,

src/state_portal.rs

@@ -8,6 +8,7 @@ use anyhow::{bail, Result};
 use crate::args::Args;
 use crate::avhw::{self, SwEncState};
 use crate::cap_portal::{CapPortal, PwCtrlEvent, PwDmaBufFrame};
+use crate::webrtc::WebRtcState;
 
 /// 门户采集的阶段状态
 /// - WaitingForFormat: 等待接收到第一帧 DMA-BUF 以确定视频格式参数
@@ -32,6 +33,10 @@ pub struct StatePortal {
     start_time: Option<Instant>,
     last_stats_time: Option<Instant>,
     last_stats_frames: u64,
+    webrtc: Option<WebRtcState>,
+    webrtc_tx: Option<crossbeam_channel::Sender<Vec<u8>>>,
+    webrtc_rx: Option<crossbeam_channel::Receiver<Vec<u8>>>,
+    webrtc_frames_sent: u64,
 }
 
 impl StatePortal {
@@ -48,6 +53,14 @@ impl StatePortal {
 
         let cap = CapPortal::new(&args)?;
 
+        let (webrtc, webrtc_tx, webrtc_rx) = if args.port > 0 {
+            let (tx, rx) = crossbeam_channel::bounded(32);
+            let wrtc = WebRtcState::new(args.port, args.fps)?;
+            (Some(wrtc), Some(tx), Some(rx))
+        } else {
+            (None, None, None)
+        };
+
         Ok(Self {
             stage: PortalStage::WaitingForFormat,
             enc: None,
@@ -59,6 +72,10 @@ impl StatePortal {
             start_time: None,
             last_stats_time: None,
             last_stats_frames: 0,
+            webrtc,
+            webrtc_tx,
+            webrtc_rx,
+            webrtc_frames_sent: 0,
         })
     }
 
@@ -68,6 +85,9 @@ impl StatePortal {
     /// `block=false` 时使用 try_recv 非阻塞检查。
     /// 返回 `Ok(true)` 表示已处理事件，`Ok(false)` 表示暂无数据。
     pub fn poll_and_encode(&mut self, block: bool) -> Result<bool> {
+        // WebRTC: process signaling, network, and forward encoded frames
+        self.poll_webrtc()?;
+
         if let Ok(ctrl) = self.cap.event_receiver().try_recv() {
             match ctrl {
                 PwCtrlEvent::StreamEnded => {
@@ -119,19 +139,39 @@ impl StatePortal {
                 let actual_bitrate = self.args.bitrate.unwrap_or_else(|| {
                     2 * (enc_width as u64) * (enc_height as u64) * (self.args.fps as u64) / 100
                 });
-                let actual_gop_size = self.args.gop_size.unwrap_or(self.args.fps);
+                let actual_gop_size = self.args.gop_size.unwrap_or_else(|| {
+                    if self.webrtc_tx.is_some() {
+                        (self.args.fps / 2).max(10)
+                    } else {
+                        self.args.fps
+                    }
+                });
 
-                let enc = avhw::SwEncState::new(
-                    &drm_path,
-                    self.args.output.as_ref(),
-                    frame.width,
-                    frame.height,
-                    enc_width,
-                    enc_height,
-                    self.args.fps,
-                    actual_bitrate,
-                    actual_gop_size,
-                )?;
+                let enc = if let Some(ref tx) = self.webrtc_tx {
+                    avhw::SwEncState::new_webrtc(
+                        &drm_path,
+                        frame.width,
+                        frame.height,
+                        enc_width,
+                        enc_height,
+                        self.args.fps,
+                        actual_bitrate,
+                        actual_gop_size,
+                        tx.clone(),
+                    )?
+                } else {
+                    avhw::SwEncState::new(
+                        &drm_path,
+                        std::path::Path::new(self.args.output.as_deref().expect("output required for MP4 mode")),
+                        frame.width,
+                        frame.height,
+                        enc_width,
+                        enc_height,
+                        self.args.fps,
+                        actual_bitrate,
+                        actual_gop_size,
+                    )?
+                };
 
                 self.enc = Some(enc);
                 self.stage = PortalStage::Streaming;
@@ -145,6 +185,9 @@ impl StatePortal {
             }
         }
 
+        // WebRTC: drain encoded frames produced by this poll before returning.
+        self.poll_webrtc()?;
+
         Ok(true)
     }
 
@@ -266,6 +309,29 @@ impl StatePortal {
     pub fn is_errored(&self) -> bool {
         self.errored
     }
+
+    fn poll_webrtc(&mut self) -> Result<()> {
+        let Some(ref mut wrtc) = self.webrtc else { return Ok(()); };
+
+        wrtc.handle_signaling()?;
+        wrtc.poll_and_feed()?;
+
+        if let Some(ref rx) = self.webrtc_rx {
+            let mut count = 0u32;
+            while let Ok(data) = rx.try_recv() {
+                count += 1;
+                if let Err(e) = wrtc.write_h264_frame(&data, self.webrtc_frames_sent, self.args.fps) {
+                    tracing::debug!("WebRTC write frame error: {e}");
+                }
+                self.webrtc_frames_sent = self.webrtc_frames_sent.saturating_add(1);
+            }
+            if count > 0 {
+                tracing::info!("WebRTC forwarded {count} frames from channel");
+            }
+        }
+
+        Ok(())
+    }
 }
 
 impl Drop for StatePortal {

src/webrtc.rs

@@ -0,0 +1,531 @@
+// WebRTC 传输模块 — 使用 str0m (Sans-IO) 将 H.264 编码帧推送到浏览器
+use std::io::{Read, Write};
+use std::net::{SocketAddr, TcpListener, UdpSocket};
+use std::time::Instant;
+
+use anyhow::{bail, Result};
+use str0m::change::SdpOffer;
+use str0m::format::Codec;
+use str0m::media::{Frequency, MediaKind, MediaTime, Mid, Pt};
+use str0m::net::{Protocol, Receive};
+use str0m::{Candidate, Event, IceConnectionState, Input, Output, Rtc, RtcConfig};
+
+// ── 嵌入式 HTML 测试页面 ──────────────────────────────────────────────────
+
+const HTML_PAGE: &str = r#"<!DOCTYPE html>
+<html>
+<head><title>wl-webrtc P0</title>
+<style>body{background:#000;color:#fff;font-family:monospace;display:flex;flex-direction:column;align-items:center;justify-content:center;height:100vh;margin:0}
+video{max-width:90vw;max-height:80vh;border:1px solid #333}
+#status{margin:12px;font-size:14px;color:#aaa}
+#debug{position:fixed;bottom:8px;left:8px;font-size:11px;color:#666;max-width:90vw;white-space:pre-wrap}
+</style></head>
+<body>
+<div id="status">Connecting...</div>
+<video id="video" autoplay playsinline muted></video>
+<pre id="debug"></pre>
+<script>
+const status = document.getElementById('status');
+const video  = document.getElementById('video');
+const debug  = document.getElementById('debug');
+let pc       = null;
+
+const log = msg => { debug.textContent += msg + '\n'; console.log(msg); };
+
+function preferH264(sdp) {
+  const lines = sdp.split('\r\n');
+  const h264Pts = lines
+    .filter(line => line.startsWith('a=rtpmap:') && line.toUpperCase().includes('H264/90000'))
+    .map(line => line.match(/^a=rtpmap:(\d+)/)?.[1])
+    .filter(Boolean);
+  if (h264Pts.length === 0) return sdp;
+  return lines.map(line => {
+    if (!line.startsWith('m=video ')) return line;
+    const parts = line.split(' ');
+    const header = parts.slice(0, 3);
+    const pts = parts.slice(3);
+    const preferred = h264Pts.filter(pt => pts.includes(pt));
+    const rest = pts.filter(pt => !preferred.includes(pt));
+    return [...header, ...preferred, ...rest].join(' ');
+  }).join('\r\n');
+}
+
+function installStatsLogger(peer) {
+  setInterval(() => {
+    if (peer !== pc) return;
+    const v = video;
+    log(`video: readyState=${v.readyState} currentTime=${v.currentTime.toFixed(2)} ` +
+        `paused=${v.paused} width=${v.videoWidth} height=${v.videoHeight} ` +
+        `srcObject=${v.srcObject ? 'yes' : 'no'}`);
+    peer.getStats().then(stats => {
+      stats.forEach(report => {
+        if (report.type === 'inbound-rtp' && report.kind === 'video') {
+          log(`RTP-in: packetsReceived=${report.packetsReceived} packetsLost=${report.packetsLost} ` +
+              `bytesReceived=${report.bytesReceived} framesDecoded=${report.framesDecoded} ` +
+              `framesDropped=${report.framesDropped} codecId=${report.codecId}`);
+        }
+        if (report.type === 'codec' && report.mimeType && report.mimeType.includes('H264')) {
+          log(`Codec: ${report.mimeType} ${report.payloadType} sdpFmtpLine=${report.sdpFmtpLine}`);
+        }
+      });
+    }).catch(() => {});
+  }, 2000);
+}
+
+function connect() {
+  if (pc) pc.close();
+  pc = new RTCPeerConnection();
+  const peer = pc;
+
+  peer.ontrack = e => {
+    log('ontrack: streams=' + e.streams.length + ' kind=' + e.track.kind);
+    video.srcObject = e.streams[0];
+    status.textContent = 'Track received';
+  };
+  peer.oniceconnectionstatechange = () => {
+    log('ICE: ' + peer.iceConnectionState);
+    status.textContent = 'ICE: ' + peer.iceConnectionState;
+  };
+
+  peer.addTransceiver('video', { direction: 'recvonly' });
+  installStatsLogger(peer);
+
+  peer.createOffer().then(offer => {
+      offer.sdp = preferH264(offer.sdp);
+      return peer.setLocalDescription(offer);
+    })
+    .then(() => new Promise(resolve => {
+      if (peer.iceGatheringState === 'complete') resolve();
+      else peer.onicegatheringstatechange = () => { if (peer.iceGatheringState === 'complete') resolve(); };
+    }))
+    .then(() => fetch('/sdp', { method: 'POST', body: JSON.stringify(peer.localDescription) }))
+    .then(r => { if (!r.ok) throw new Error('SDP exchange failed: ' + r.status); return r.json(); })
+    .then(answer => { if (answer.error) throw new Error(answer.error); return peer.setRemoteDescription(answer); })
+    .then(() => log('SDP answer set'))
+    .catch(e => {
+      status.textContent = 'Error: ' + e.message;
+      log('ERROR: ' + e.message + ' — retrying in 2s...');
+      console.error(e);
+      setTimeout(connect, 2000);
+    });
+}
+
+connect();
+</script>
+</body></html>"#;
+
+// ── WebRTC 状态 ───────────────────────────────────────────────────────────
+
+pub struct WebRtcState {
+    signal_listener: TcpListener,
+    inner: Option<WebRtcInner>,
+    fps: u32,
+}
+
+struct WebRtcInner {
+    rtc: Rtc,
+    socket: UdpSocket,
+    udp_addr: SocketAddr,
+    video_mid: Option<Mid>,
+    video_pt: Option<Pt>,
+    connected: bool,
+    need_keyframe: bool,
+    rtp_clock: u32,
+    buf: Vec<u8>,
+}
+
+impl WebRtcState {
+    pub fn new(port: u16, fps: u32) -> Result<Self> {
+        let signal_listener = TcpListener::bind(format!("0.0.0.0:{port}"))?;
+        signal_listener.set_nonblocking(true)?;
+        tracing::info!("WebRTC signaling on http://0.0.0.0:{port}/");
+
+        Ok(Self {
+            signal_listener,
+            inner: None,
+            fps,
+        })
+    }
+
+    pub fn handle_signaling(&mut self) -> Result<bool> {
+        let mut handled = false;
+        loop {
+            let (mut stream, _addr) = match self.signal_listener.accept() {
+                Ok(s) => s,
+                Err(ref e) if e.kind() == std::io::ErrorKind::WouldBlock => break,
+                Err(e) => bail!("TCP accept error: {e}"),
+            };
+            handled = true;
+            stream.set_nonblocking(true)?;
+
+            let mut req = vec![0u8; 65536];
+            let n = match stream.read(&mut req) {
+                Ok(n) => n,
+                Err(ref e) if e.kind() == std::io::ErrorKind::WouldBlock => continue,
+                Err(e) => {
+                    tracing::warn!("TCP read error: {e}");
+                    continue;
+                }
+            };
+            let req_str = String::from_utf8_lossy(&req[..n]);
+
+            if req_str.starts_with("GET / ")
+                || req_str.starts_with("GET /sdp ")
+                    && !req_str.contains("Content-Type: application/json")
+            {
+                let resp = format!(
+                    "HTTP/1.1 200 OK\r\nContent-Type: text/html\r\nContent-Length: {}\r\nConnection: close\r\n\r\n{}",
+                    HTML_PAGE.len(),
+                    HTML_PAGE
+                );
+                let _ = stream.write_all(resp.as_bytes());
+            } else if req_str.starts_with("POST /sdp") {
+                let body = extract_body(&req_str);
+                if body.is_empty() {
+                    let resp = "HTTP/1.1 400 Bad Request\r\nConnection: close\r\n\r\nempty body";
+                    let _ = stream.write_all(resp.as_bytes());
+                    continue;
+                }
+
+                match WebRtcInner::new(self.fps)
+                    .and_then(|mut new_inner| {
+                        let answer_json = new_inner.handle_sdp_offer(body.as_bytes())?;
+                        Ok((new_inner, answer_json))
+                    }) {
+                    Ok((new_inner, answer_json)) => {
+                        let replacing = self.inner.is_some();
+                        self.inner = Some(new_inner);
+                        if replacing {
+                            tracing::info!("Replaced WebRTC connection (old dropped)");
+                        } else {
+                            tracing::info!("New WebRTC connection");
+                        }
+
+                        let resp = format!(
+                            "HTTP/1.1 200 OK\r\nContent-Type: application/json\r\nContent-Length: {}\r\nConnection: close\r\n\r\n{}",
+                            answer_json.len(),
+                            answer_json
+                        );
+                        let _ = stream.write_all(resp.as_bytes());
+                    }
+                    Err(e) => {
+                        tracing::error!("SDP offer handling failed: {e}");
+                        let resp = format!("HTTP/1.1 500 Error\r\nConnection: close\r\n\r\n{e}");
+                        let _ = stream.write_all(resp.as_bytes());
+                    }
+                }
+            } else {
+                let resp = "HTTP/1.1 404 Not Found\r\nConnection: close\r\n\r\n";
+                let _ = stream.write_all(resp.as_bytes());
+            }
+        }
+        Ok(handled)
+    }
+
+    pub fn poll_rtc(&mut self) -> Result<()> {
+        if let Some(inner) = self.inner.as_mut() {
+            if inner.poll_rtc()? {
+                tracing::warn!("WebRTC connection closed/failed; clearing connection state");
+                self.inner = None;
+            }
+        }
+        Ok(())
+    }
+
+    pub fn feed_network(&mut self) -> Result<()> {
+        if let Some(inner) = self.inner.as_mut() {
+            inner.feed_network()?;
+        }
+        Ok(())
+    }
+
+    pub fn poll_and_feed(&mut self) -> Result<()> {
+        self.poll_rtc()?;
+        self.feed_network()?;
+        self.poll_rtc()
+    }
+
+    pub fn write_h264_frame(&mut self, data: &[u8], frame_number: u64, fps: u32) -> Result<()> {
+        if let Some(inner) = self.inner.as_mut() {
+            inner.write_h264_frame(data, frame_number, fps)?;
+        }
+        Ok(())
+    }
+
+    pub fn is_connected(&self) -> bool {
+        self.inner.as_ref().is_some_and(WebRtcInner::is_connected)
+    }
+}
+
+impl WebRtcInner {
+    fn new(fps: u32) -> Result<Self> {
+        let _ = fps;
+        let mut rtc = RtcConfig::new().build(Instant::now());
+
+        let socket = UdpSocket::bind("0.0.0.0:0")?;
+        socket.set_nonblocking(true)?;
+        let local_addr = socket.local_addr()?;
+
+        let lan_ip = local_ip().unwrap_or_else(|| {
+            tracing::warn!("Failed to detect LAN IP, falling back to 127.0.0.1");
+            "127.0.0.1".to_string()
+        });
+        let candidate_addr: SocketAddr = format!("{lan_ip}:{}", local_addr.port()).parse()?;
+        let candidate = Candidate::host(candidate_addr, "udp")
+            .map_err(|e| anyhow::anyhow!("candidate: {e}"))?;
+        rtc.add_local_candidate(candidate);
+        tracing::info!("WebRTC UDP: {candidate_addr} (bound 0.0.0.0)");
+
+        Ok(Self {
+            rtc,
+            socket,
+            udp_addr: candidate_addr,
+            video_mid: None,
+            video_pt: None,
+            connected: false,
+            need_keyframe: false,
+            rtp_clock: 0,
+            buf: vec![0u8; 65535],
+        })
+    }
+
+    fn handle_sdp_offer(&mut self, body: &[u8]) -> Result<String> {
+        let offer: SdpOffer = serde_json::from_slice(body)
+            .map_err(|e| anyhow::anyhow!("parse SDP offer: {e}"))?;
+
+        let answer = self
+            .rtc
+            .sdp_api()
+            .accept_offer(offer)
+            .map_err(|e| anyhow::anyhow!("accept_offer: {e}"))?;
+
+        self.need_keyframe = true;
+        tracing::info!("SDP exchange complete, waiting for ICE/DTLS...");
+
+        self.discover_video_params();
+
+        let answer_json =
+            serde_json::to_vec(&answer).map_err(|e| anyhow::anyhow!("serialize answer: {e}"))?;
+
+        String::from_utf8(answer_json).map_err(|e| anyhow::anyhow!("answer utf8: {e}"))
+    }
+
+    fn discover_video_params(&mut self) {
+        for s in ["0", "1", "2", "3"] {
+            let mid: Mid = s.into();
+            if let Some(media) = self.rtc.media(mid) {
+                if media.kind() == MediaKind::Video {
+                    tracing::info!("Found video media: mid={mid}");
+                    self.video_mid = Some(mid);
+                    break;
+                }
+            }
+        }
+
+        if let Some(mid) = self.video_mid {
+            if let Some(writer) = self.rtc.writer(mid) {
+                for pp in writer.payload_params() {
+                    tracing::debug!("Codec: pt={:?} spec={:?}", pp.pt(), pp.spec());
+                    if pp.spec().codec.is_video() && pp.spec().codec == Codec::H264 {
+                        self.video_pt = Some(pp.pt());
+                        tracing::info!("H.264 payload type: {:?}", pp.pt());
+                        break;
+                    }
+                }
+            }
+        }
+    }
+
+    fn poll_rtc(&mut self) -> Result<bool> {
+        loop {
+            match self.rtc.poll_output() {
+                Ok(Output::Transmit(t)) => {
+                    tracing::info!("TX {} bytes -> {}", t.contents.len(), t.destination);
+                    if let Err(e) = self.socket.send_to(&t.contents, t.destination) {
+                        tracing::warn!("UDP send error: {e}");
+                    }
+                }
+                Ok(Output::Event(e)) => {
+                    tracing::info!("RTC event: {e:?}");
+                    match &e {
+                        Event::Connected => {
+                            tracing::info!("WebRTC connected!");
+                            self.connected = true;
+                            self.need_keyframe = true;
+                            self.discover_video_params();
+                        }
+                        Event::IceConnectionStateChange(IceConnectionState::Disconnected) => {
+                            tracing::warn!("WebRTC disconnected");
+                            self.connected = false;
+                        }
+                        Event::MediaAdded(ma) => {
+                            tracing::info!("Media added: mid={:?}", ma.mid);
+                        }
+                        _ => {
+                            tracing::debug!("WebRTC event: {:?}", e);
+                        }
+                    }
+                }
+                Ok(Output::Timeout(_t)) => break,
+                Err(e) => {
+                    tracing::error!("rtc.poll_output error: {e}");
+                    break;
+                }
+            }
+        }
+        Ok(false)
+    }
+
+    fn feed_network(&mut self) -> Result<()> {
+        let mut recv_count = 0u32;
+        loop {
+            match self.socket.recv_from(&mut self.buf) {
+                Ok((n, source)) => {
+                    recv_count += 1;
+                    if recv_count <= 5 {
+                        tracing::info!("UDP recv {} bytes from {}", n, source);
+                    }
+                    let input = Input::Receive(
+                        Instant::now(),
+                        Receive {
+                            proto: Protocol::Udp,
+                            source,
+                            destination: self.udp_addr,
+                            contents: self.buf[..n]
+                                .try_into()
+                                .map_err(|e| anyhow::anyhow!("receive contents: {e}"))?,
+                        },
+                    );
+                    self.rtc
+                        .handle_input(input)
+                        .map_err(|e| anyhow::anyhow!("handle_input({n} bytes from {source}): {e}"))?;
+                }
+                Err(ref e) if e.kind() == std::io::ErrorKind::WouldBlock => break,
+                Err(ref e) if e.kind() == std::io::ErrorKind::Interrupted => continue,
+                Err(e) => bail!("UDP recv error: {e}"),
+            }
+        }
+
+        self.rtc
+            .handle_input(Input::Timeout(Instant::now()))
+            .map_err(|e| anyhow::anyhow!("handle timeout: {e}"))?;
+
+        Ok(())
+    }
+
+    fn write_h264_frame(&mut self, data: &[u8], frame_number: u64, fps: u32) -> Result<()> {
+        if !self.connected {
+            return Ok(());
+        }
+
+        let mid = match self.video_mid {
+            Some(m) => m,
+            None => {
+                tracing::warn!("write_h264: no video_mid");
+                return Ok(());
+            }
+        };
+        let pt = match self.video_pt {
+            Some(p) => p,
+            None => {
+                tracing::warn!("write_h264: no video_pt");
+                return Ok(());
+            }
+        };
+
+        if self.need_keyframe {
+            if !is_idr_nalu(data) {
+                tracing::debug!(
+                    "write_h264: skipping non-IDR frame ({} bytes), waiting for keyframe",
+                    data.len()
+                );
+                return Ok(());
+            }
+            tracing::info!(
+                "write_h264: got IDR keyframe ({} bytes), starting playback",
+                data.len()
+            );
+            self.need_keyframe = false;
+        }
+
+        let ticks_per_second = 90_000u64;
+        let fps = fps.max(1) as u64;
+        let rtp_timestamp = frame_number.saturating_mul(ticks_per_second) / fps;
+        self.rtp_clock = rtp_timestamp as u32;
+        let rtp_time = MediaTime::new(rtp_timestamp, Frequency::NINETY_KHZ);
+
+        let writer = match self.rtc.writer(mid) {
+            Some(w) => w,
+            None => {
+                tracing::warn!("write_h264: no writer for mid={mid}");
+                return Ok(());
+            }
+        };
+
+        tracing::debug!(
+            "write_h264: {} bytes, pt={:?}, rtp={}",
+            data.len(),
+            pt,
+            self.rtp_clock
+        );
+        writer
+            .write(pt, Instant::now(), rtp_time, data)
+            .map_err(|e| anyhow::anyhow!("writer.write: {e}"))?;
+
+        self.poll_rtc()?;
+
+        Ok(())
+    }
+
+    fn is_connected(&self) -> bool {
+        self.connected
+    }
+}
+
+// ── 工具函数 ──────────────────────────────────────────────────────────────
+
+/// 从 HTTP 请求中提取 body（在 \r\n\r\n 之后）
+fn extract_body(req: &str) -> &str {
+    if let Some(idx) = req.find("\r\n\r\n") {
+        req.get(idx + 4..).unwrap_or("")
+    } else {
+        ""
+    }
+}
+
+fn local_ip() -> Option<String> {
+    std::net::UdpSocket::bind("0.0.0.0:0")
+        .ok()
+        .and_then(|s| {
+            s.connect("1.1.1.1:80").ok()?;
+            let addr = s.local_addr().ok()?;
+            drop(s);
+            let ip = addr.ip().to_string();
+            if ip == "0.0.0.0" || ip.starts_with("127.") {
+                return None;
+            }
+            Some(ip)
+        })
+}
+
+fn is_idr_nalu(data: &[u8]) -> bool {
+    let mut i = 0;
+    while i + 4 < data.len() {
+        if data[i..i + 4] == [0, 0, 0, 1] {
+            let nal_type = data[i + 4] & 0x1F;
+            if nal_type == 5 {
+                return true;
+            }
+            i += 5;
+        } else if i + 3 < data.len() && data[i..i + 3] == [0, 0, 1] {
+            let nal_type = data[i + 3] & 0x1F;
+            if nal_type == 5 {
+                return true;
+            }
+            i += 4;
+        } else {
+            i += 1;
+        }
+    }
+    false
+}