SRT: Enable tlpktdrop by default to prevent 100% CPU usage. v7.0.133 (#4587) (#4601)

2026-01-12 00:05:45 +08:00 · 2025-12-04 09:47:01 -05:00
parent e970ad26fd
commit a83c9026f9
12 changed files with 100 additions and 26 deletions
--- a/.augment-guidelines
+++ b/.augment-guidelines
@@ -13,7 +13,7 @@ architecture:

  origin_cluster:
    v6_mesh: "DEPRECATED - C++ MESH mode, RTMP only"
-    v7_proxy: "RECOMMENDED - Go-based proxy (github.com/ossrs/proxy-go), all protocols, better scaling"
+    v7_proxy: "RECOMMENDED - Go-based proxy (https://github.com/ossrs/srsx/tree/main/proxy-go), all protocols, better scaling"

 codebase_structure:
  dirs: "trunk/src/{main,core,kernel,protocol,app}"
--- a/proxy/README.md
+++ b/proxy/README.md
@@ -2,5 +2,5 @@

 Migrated to below repositoties:

-* [proxy-go](https://github.com/ossrs/proxy-go) An common proxy server for any media servers with RTMP/SRT/HLS/HTTP-FLV and WebRTC/WHIP/WHEP protocols support.
+* [srsx/proxy-go](https://github.com/ossrs/srsx/tree/main/proxy-go) An common proxy server for any media servers with RTMP/SRT/HLS/HTTP-FLV and WebRTC/WHIP/WHEP protocols support.

--- a/trunk/3rdparty/srs-docs/doc/origin-cluster.md
+++ b/trunk/3rdparty/srs-docs/doc/origin-cluster.md
@@ -45,12 +45,12 @@ Please see [Protocols](#protocols) for details.

 ## Build

-To build the proxy server, you need to have Go 1.18+ installed. Then, you can build the proxy 
+To build the proxy server, you need to have Go 1.18+ installed. Then, you can build the proxy
 server by below command, and get the executable binary `./srs-proxy`:

 ```bash
-git clone https://github.com/ossrs/proxy-go.git
-cd proxy-go && make
+git clone https://github.com/ossrs/srsx.git
+cd srsx/proxy-go && make
 ```

 > Note: You can also download the dependencies by running `go mod download` before building.
--- a/trunk/3rdparty/srs-docs/doc/srt.md
+++ b/trunk/3rdparty/srs-docs/doc/srt.md
@@ -269,13 +269,37 @@ srt_server {
    peerlatency 0;
    recvlatency 0;
    latency 0;
-    tlpktdrop off;
+    tlpktdrop on;
    tsbpdmode off;
 }
 ```

 > Note: If you still experience screen glitches with the above settings, please refer to the [FFmpeg patch](https://github.com/FFmpeg/FFmpeg/commit/9099046cc76c9e3bf02f62a237b4d444cdaf5b20).

+## Too-Late Packet Drop
+
+The `tlpktdrop` (Too-Late Packet Drop) setting controls whether SRT discards packets that arrive later than the configured latency window.
+
+When `tlpktdrop off` is combined with `tsbpdmode on`, the SRT receiver buffer can fill up with packets that cannot be recovered. Over time, iterating through this buffer to request retransmissions consumes 100% CPU, causing the server to become completely unresponsive - no new streams can be published, existing streams stop working, and even the HTTP API becomes unavailable.
+
+**Recommended configuration:**
+
+```
+srt_server {
+    enabled on;
+    listen 10080;
+    tlpktdrop on;
+}
+```
+
+**Key points:**
+
+* `tlpktdrop on` does NOT disable retransmission entirely. Packets within the latency window will still be retransmitted.
+* With `tlpktdrop on`, if the network has severe jitter, some late packets may be dropped, causing minor video glitches.
+* The trade-off is: occasional glitches (with `tlpktdrop on`) vs. potential server hang (with `tlpktdrop off` + `tsbpdmode on`).
+
+For more details, see [#4587](https://github.com/ossrs/srs/issues/4587).
+
 ## Video codec

 Currently, H264 and HEVC encoding are supported. Since SRT protocol transfers media in MPEG-TS format, which already supports HEVC encoding (streamtype 0x24), SRT can naturally transmit HEVC encoded video without any modifications.
--- a/trunk/3rdparty/srs-docs/pages/product-en.md
+++ b/trunk/3rdparty/srs-docs/pages/product-en.md
@@ -91,27 +91,58 @@ For a detailed interpretation, please see Welcome to SRS: Mission, Vision, and V

 ## Release 7.0

-Code name: Kai. Named by TOC member [Haibo Chen](https://github.com/duiniuluantanqin). Expected to complete major development by the end of 2026 and officially release.
+Code name: Kai. Named by TOC member [Haibo Chen](https://github.com/duiniuluantanqin). Planned for release by the end of 2026.

 > Code Name Story: I am Haibo Chen, a core maintainer of SRS and a TOC member. The code name Kai is inspired by my son Chen Kaiqi's name. As a father, I aim to set a good example by doing meaningful and interesting work. I appreciate the support and collaboration from everyone in the community, making it more vibrant and warm. This upgrade aims to provide users with more powerful features and a smoother experience, laying a strong foundation for SRS's future.

- [x] Support for Proxy Cluster, allowing more stream paths. [#4158](https://github.com/ossrs/srs/pull/4158)
- [ ] WebRTC support for HEVC, recording HEVC to MP4 files, completing full HEVC support. [#4289](https://github.com/ossrs/srs/pull/4289), [#4349](https://github.com/ossrs/srs/pull/4349), [#4296](https://github.com/ossrs/srs/pull/4296)
- [ ] HLS protocol support for fMP4. [#4159](https://github.com/ossrs/srs/pull/4159)
- [ ] Support for RTSP protocol playback. [#4333](https://github.com/ossrs/srs/pull/4333)
+- [x] Proxy Cluster - Support for more stream paths. [#4158](https://github.com/ossrs/srs/pull/4158)
+- [ ] WebRTC HEVC - WebRTC support for HEVC, recording HEVC to MP4 files, completing full HEVC support. [#4289](https://github.com/ossrs/srs/pull/4289), [#4349](https://github.com/ossrs/srs/pull/4349), [#4296](https://github.com/ossrs/srs/pull/4296)
+- [ ] HLS fMP4 - HLS protocol support for fMP4. [#4159](https://github.com/ossrs/srs/pull/4159)
+- [ ] RTSP Playback - Support for RTSP protocol playback. [#4333](https://github.com/ossrs/srs/pull/4333)

 ## Release 6.0

-Development codename: Hang. It is planned for release by the end of 2025.
+Development codename: Hang. Named by TOC [John](https://github.com/xiaozhihong). Released on December 3, 2024, now the stable version replacing SRS 5.0.

-> Note: The development codename Hang is named by TOC [John](https://github.com/xiaozhihong), and the specific meaning 
+> Note: The development codename Hang is named by TOC [John](https://github.com/xiaozhihong), and the specific meaning
 > is left for everyone to appreciate and ponder.

- [x] Supports HEVC encoding format, including protocols like RTMP, HTTP-FLV, HTTP-TS, HLS, and SRT. [#465](https://github.com/ossrs/srs/issues/465)
- [x] Uses smart pointers (SrsUniquePtr and SrsSharedPtr) to improve memory management and fixes multiple memory leak issues. [#4089](https://github.com/ossrs/srs/pull/4089), [#4109](https://github.com/ossrs/srs/pull/4109)
- [x] Supports IP whitelisting for HTTP-FLV, HLS, WebRTC, and SRT. [#3902](https://github.com/ossrs/srs/pull/3902)
- [x] Added Basic Authentication feature for HTTP API. [#3458](https://github.com/ossrs/srs/pull/3458)
- [x] GB28181 protocol supports external SIP servers. [#4101](https://github.com/ossrs/srs/pull/4101), [srs-sip](https://github.com/ossrs/srs-sip)
+**HEVC/H.265 Support (Major Feature)**
+
+- [x] HEVC over RTMP/HTTP-FLV - Full H.265 support for RTMP and HTTP-FLV streaming. [#465](https://github.com/ossrs/srs/issues/465)
+- [x] HEVC over HLS - H.265 support in HLS delivery.
+- [x] HEVC over HTTP-TS - H.265 support for MPEG-TS streaming.
+- [x] HEVC over SRT - H.265 support for SRT protocol.
+- [x] HEVC DVR to MP4 - Record H.265 streams to MP4 format.
+- [x] HEVC DVR to FLV - Record H.265 streams to FLV fragments.
+
+**WebRTC Enhancements**
+
+- [x] WHEP Support - WebRTC-HTTP Egress Protocol for playback.
+- [x] OPUS Stereo SDP option - Support for stereo audio in WebRTC.
+- [x] Configurable AAC to Opus bitrate - Customize transcoding bitrate.
+
+**Security & Authentication**
+
+- [x] IP Whitelist - Security support for HTTP-FLV, HLS, WebRTC, and SRT. [#3902](https://github.com/ossrs/srs/pull/3902)
+- [x] Basic Authentication - HTTP API authentication feature. [#3458](https://github.com/ossrs/srs/pull/3458)
+
+**SRT Improvements**
+
+- [x] Upgraded libsrt to v1.5.3 - Latest SRT library version.
+- [x] Configurable default_streamid - Customizable SRT stream IDs.
+- [x] Reduced SRT-to-RTC latency - Down to 200ms.
+
+**HLS Features**
+
+- [x] Kick-off HLS clients - Ability to disconnect HLS clients.
+
+**Other Improvements**
+
+- [x] Smart pointers (SrsUniquePtr and SrsSharedPtr) - Improved memory management and fixed multiple memory leak issues. [#4089](https://github.com/ossrs/srs/pull/4089), [#4109](https://github.com/ossrs/srs/pull/4109)
+- [x] GB28181 protocol - Support for external SIP servers. [#4101](https://github.com/ossrs/srs/pull/4101), [srs-sip](https://github.com/ossrs/srs-sip)
+
+SRS 6.0 was released at 2024.12, see [6.0-r0](https://github.com/ossrs/srs/releases/tag/v6.0-r0).

 # Release 5.0

--- a/trunk/conf/full.conf
+++ b/trunk/conf/full.conf
@@ -723,13 +723,15 @@ srt_server {
    # Overwrite by env SRS_SRT_SERVER_TSBPDMODE
    # default: on
    tsbpdmode off;
-	# The tlpkdrop means too-late Packet Drop
-	# SRT sender side will pack timestamp in each packet, When the network is congested,
+	# The tlpkdrop means too-late Packet Drop.
+	# SRT sender side will pack timestamp in each packet. When the network is congested,
 	# the packet will drop if latency is bigger than the configuration in both sender side and receiver side.
 	# And on the sender side, it also will be dropped because latency is bigger than configuration.
+	# Note: It's strongly recommended to keep this on. When off with tsbpdmode on, it may cause
+	# buffer overflow and 100% CPU usage due to retransmission storms. See https://github.com/ossrs/srs/issues/4587
    # Overwrite by env SRS_SRT_SERVER_TLPKTDROP
    # default: on
-    tlpktdrop off;
+    tlpktdrop on;
 	# The send buffer size of SRT.
    # Overwrite by env SRS_SRT_SERVER_SENDBUF
    # default:  8192 * (1500-28)
--- a/trunk/conf/srt.conf
+++ b/trunk/conf/srt.conf
@@ -28,7 +28,11 @@ srt_server {
    recvlatency 0;
    latency 0;
    tsbpdmode off;
-    tlpktdrop off;
+    # Whether to enable too-late packet drop.
+    # When enabled, packets arriving later than the latency window will be dropped,
+    # which prevents buffer overflow and 100% CPU usage caused by retransmission storms.
+    # See https://github.com/ossrs/srs/issues/4587
+    tlpktdrop on;
    sendbuf 2000000;
    recvbuf 2000000;
    default_streamid "#!::r=live/livestream,m=publish";
--- a/trunk/conf/srt.shortstreamid.play.conf
+++ b/trunk/conf/srt.shortstreamid.play.conf
@@ -28,7 +28,11 @@ srt_server {
    recvlatency 0;
    latency 0;
    tsbpdmode off;
-    tlpktdrop off;
+    # Whether to enable too-late packet drop.
+    # When enabled, packets arriving later than the latency window will be dropped,
+    # which prevents buffer overflow and 100% CPU usage caused by retransmission storms.
+    # See https://github.com/ossrs/srs/issues/4587
+    tlpktdrop on;
    sendbuf 2000000;
    recvbuf 2000000;
    payloadsize 1316;
--- a/trunk/conf/srt.shortstreamid.publish.conf
+++ b/trunk/conf/srt.shortstreamid.publish.conf
@@ -28,7 +28,11 @@ srt_server {
    recvlatency 0;
    latency 0;
    tsbpdmode off;
-    tlpktdrop off;
+    # Whether to enable too-late packet drop.
+    # When enabled, packets arriving later than the latency window will be dropped,
+    # which prevents buffer overflow and 100% CPU usage caused by retransmission storms.
+    # See https://github.com/ossrs/srs/issues/4587
+    tlpktdrop on;
    sendbuf 2000000;
    recvbuf 2000000;
    payloadsize 1316;
--- a/trunk/conf/srt.vlc.conf
+++ b/trunk/conf/srt.vlc.conf
@@ -28,7 +28,11 @@ srt_server {
    recvlatency 0;
    latency 0;
    tsbpdmode off;
-    tlpktdrop off;
+    # Whether to enable too-late packet drop.
+    # When enabled, packets arriving later than the latency window will be dropped,
+    # which prevents buffer overflow and 100% CPU usage caused by retransmission storms.
+    # See https://github.com/ossrs/srs/issues/4587
+    tlpktdrop on;
    sendbuf 2000000;
    recvbuf 2000000;
    default_streamid "#!::r=live/livestream,m=request";
--- a/trunk/doc/CHANGELOG.md
+++ b/trunk/doc/CHANGELOG.md
@@ -7,6 +7,7 @@ The changelog for SRS.
 <a name="v7-changes"></a>

 ## SRS 7.0 Changelog
+* v7.0, 2025-12-04, SRT: Enable tlpktdrop by default to prevent 100% CPU usage. v7.0.133 (#4587)
 * v7.0, 2025-12-03, AI: WebRTC: Fix audio-only WHIP publish without SSRC. v7.0.132 (#4570)
 * v7.0, 2025-11-30, SRT: Support default_mode config for short streamid format. v7.0.131
 * v7.0, 2025-11-28, SRT: Fix player not exiting when publisher disconnects. v7.0.130 (#4591)
--- a/trunk/src/core/srs_core_version7.hpp
+++ b/trunk/src/core/srs_core_version7.hpp
@@ -9,6 +9,6 @@

 #define VERSION_MAJOR 7
 #define VERSION_MINOR 0
-#define VERSION_REVISION 132
+#define VERSION_REVISION 133

 #endif