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Haibo Chen(陈海博) ef048b0d65 RTC: Fix DVR missing first 4-6 seconds by initializing rate from SDP (#4541) 
for issue #4418, #4151, #4076 .DVR Missing First Few Seconds of
Audio/Video

### Root Cause
When recording WebRTC streams to FLV files using DVR, the first 4-6
seconds of audio/video are missing. This occurs because:

1. **Packets are discarded before A/V sync is available**: The
RTC-to-RTMP conversion pipeline actively discards all RTP packets when
avsync_time <= 0.
2. **Original algorithm requires 2 RTCP SR packets**: The previous
implementation needed to receive two RTCP Sender Report (SR) packets
before it could calculate the rate for audio/video synchronization
timestamp conversion.
3. **Delay causes packet loss**: Since RTCP SR packets typically arrive
every 2-3 seconds, waiting for 2 SRs means 4-6 seconds of packets are
discarded before A/V sync becomes available.
4. **Audio SR arrives slower than video SR**: As reported in the issue,
video RTCP SR packets arrive much faster than audio SR packets. This
asymmetry causes audio packets to be discarded for a longer period,
resulting in the audio loss observed in DVR recordings.

### Solution
1. **Initialize rate from SDP**: Use the sample rate from SDP (Session
Description Protocol) to calculate the initial rate immediately when the
track is created.
Audio (Opus): 48000 Hz → rate = 48 (RTP units per millisecond)
Video (H.264/H.265): 90000 Hz → rate = 90 (RTP units per millisecond)
2. **Enable immediate A/V sync:** With the SDP rate available,
cal_avsync_time() can calculate valid timestamps from the very first RTP
packet, eliminating packet loss.
3. **Smooth transition to precise rate**: After receiving the 2nd RTCP
SR, update to the precisely calculated rate based on actual RTP/NTP
timestamp mapping.

## Configuration

Added new configuration option `init_rate_from_sdp` in the RTC vhost
section:

```nginx
vhost rtc.vhost.srs.com {
    rtc {
        # Whether initialize RTP rate from SDP sample rate for immediate A/V sync.
        # When enabled, the RTP rate (units per millisecond) is initialized from the SDP
        # sample rate (e.g., 90 for video 90kHz, 48 for audio 48kHz) before receiving
        # 2 RTCP SR packets. This allows immediate audio/video synchronization.
        # The rate will be updated to a more precise value after receiving the 2nd SR.
        # Overwrite by env SRS_VHOST_RTC_INIT_RATE_FROM_SDP for all vhosts.
        # Default: off
        init_rate_from_sdp off;
    }
}
```

**⚠️ Important Note**: This config defaults to **off** because:
- ✅ When **enabled**: Fixes the audio loss problem (no missing first 4-6
seconds)
- ❌ When **enabled**: VLC on macOS cannot play the video properly
- ✅ Other platforms work fine (Windows, Linux)
- ✅ FFplay works fine on all platforms

Users experiencing audio loss in DVR recordings can enable this option
if they don't need VLC macOS compatibility. We're investigating the VLC
macOS issue to make this feature safe to enable by default in the
future.

---------

Co-authored-by: winlin <winlinvip@gmail.com>
Co-authored-by: OSSRS-AI <winlinam@gmail.com>
2025-10-28 09:33:40 -04:00
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AI: Add utest to cover encoder module.
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2024-09-09 10:37:41 +08:00
.vscode
Refactor: Rename ide/ directory to cmake/ for better clarity (#4539)
2025-10-23 20:38:48 -04:00
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Migrate proxy to ossrs/proxy-go repository.
2025-06-23 10:39:57 -04:00
trunk
RTC: Fix DVR missing first 4-6 seconds by initializing rate from SDP (#4541)
2025-10-28 09:33:40 -04:00
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Update guideline for AI about sanitizer.
2025-10-26 16:28:02 -04:00
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Refactor: Rename ide/ directory to cmake/ for better clarity (#4539)
2025-10-23 20:38:48 -04:00
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AI: Add utest to cover process module
2025-10-15 07:52:46 -04:00
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AI: Improve converage for app rtc module.
2025-09-27 09:40:57 -04:00
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CONTRIBUTING.md
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README.md
Update README for v6.0-b2
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SECURITY.md
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2023-01-30 14:20:37 +08:00

README.md

SRS(Simple Realtime Server)

SRS/7.0 (Kai) is a simple, high-efficiency, and real-time video server, supporting RTMP/WebRTC/HLS/HTTP-FLV/SRT/MPEG-DASH/GB28181, Linux/macOS, X86_64/ARMv7/AARCH64/M1/RISCV/LOONGARCH/MIPS, and essential features.

SRS Overview

Note: For more details on the single-node architecture for SRS, please visit the following link.

SRS is licenced under MIT, and some third-party libraries are distributed under their licenses.

Usage

Please check the Getting Started guide in English or Chinese. We highly recommend using SRS with docker:

docker run --rm -it -p 1935:1935 -p 1985:1985 -p 8080:8080 \
    -p 8000:8000/udp -p 10080:10080/udp ossrs/srs:6

Tips: If you're in China, use this image registry.cn-hangzhou.aliyuncs.com/ossrs/srs:6 for faster speed.

Open http://localhost:8080/ to verify, and then stream using the following FFmpeg command:

ffmpeg -re -i ./doc/source.flv -c copy -f flv -y rtmp://localhost/live/livestream

Alternatively, stream by OBS using the following configuration:

  • Service: Custom
  • Server: rtmp://localhost/live
  • Stream Key: livestream

Play the following streams using media players:

If you'd like to use WebRTC, convert RTMP to WebRTC, or convert WebRTC to RTMP, please check out the wiki documentation in either English or Chinese.

To learn more about RTMP, HLS, HTTP-FLV, SRT, MPEG-DASH, WebRTC protocols, clustering, HTTP API, DVR, and transcoding, please check the documents in English or Chinese.

If you want to use an IDE, VSCode is recommanded. VSCode supports macOS, and Linux platforms. The settings are ready. All you need to do is open the folder with VSCode and enjoy the efficiency brought by the IDE. See VSCode README for details.

Sponsor

Would you like additional assistance from us? By becoming a sponsor or backer of SRS, we can provide you with the support you need:

  • Backer: $5 per month, online text chat support through Discord.
  • Sponsor: $100 per month, online text chat plus online meeting support.

Please visit OpenCollective to become a backer or sponsor, and send us a direct message on Discord. We are currently providing support to the developers listed below:

At SRS, our goal is to create a free, open-source community that helps developers all over the world build high-quality streaming and RTC platforms for their businesses.

Contributing

The maintainers, and contributors are listed here. The maintainers who made significant contributions and maintained parts of SRS are listed below, ranked by the number of commits:

  • Winlin: Founder of the project, focusing on ST and Issues/PR. Responsible for architecture and maintenance.
  • XiaoZhihong: Concentrates on WebRTC/QUIC and SRT, with expertise in network QoS. Contributed to ARM on ST and was the original contributor for WebRTC.
  • ChenHaibo: Specializes in GB28181 and HTTP API, contributing to patches for FFmpeg with WHIP.
  • ZhangJunqin: Focused on H.265, Prometheus Exporter, and API module.
  • XiaLixin: Specializes in GB28181, with expertise in live streaming and WebRTC.
  • Jacob Su: Jacob Su has contributed to various modules of SRS.
  • ShiWei: Specializes in SRT and H.265, maintaining SRT and FLV patches for FFmpeg. An expert in codecs and FFmpeg.
  • ChenGuanghua: Focused on WebRTC/QoS and introduced the Asan toolchain to SRS.
  • LiPeng: Concentrates on WebRTC and contributes to memory management and smart pointers.
  • ZhaoWenjie: One of the earliest contributors, focusing on HDS. Has expertise in client technology.
  • WuPengqiang: Focused on H.265, initially contributed to the FFmpeg module in SRS for transcoding AAC with OPUS for WebRTC.

A huge THANK YOU goes out to:

We're really thankful to everyone in the community for helping us find bugs and improve the project. To stay in touch and keep helping our community, please check out this guide.

LICENSE

SRS is licenced under MIT, and some third-party libraries are distributed under their licenses.

Releases

Features

Please read FEATURES.

Changelog

Please read CHANGELOG.

Performance

Please read PERFORMANCE.

Architecture

Please read ARCHITECTURE.

Ports

Please read PORTS.

APIs

Please read APIS.

Mirrors

Please read MIRRORS.

Dockers

Please read DOCKERS.

Beijing, 2013.10
Winlin

Description
SRS is a simple, high-efficiency, real-time media server supporting RTMP, WebRTC, HLS, HTTP-FLV, HTTP-TS, SRT, MPEG-DASH, and GB28181.
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