SRT egress to lalmax (gosrt) fails: connection established but readBytes=0, no data transmitted

[124111294]

Describe the bug

When using SRT stream output to push to a lalmax SRT server (https://github.com/q191201771/lalmax, which uses the gosrt library), the SRT connection is established successfully but no data is transmitted (readBytes=0 on the server side). The egress then keeps reconnecting in an infinite loop.

The same SRT server works perfectly with ffmpeg, which strongly suggests a compatibility issue between GStreamer's srtsink element and the gosrt library's SRT implementation.

To Reproduce

1. Start a lalmax SRT server on port 6001
2. Start a LiveKit room with a participant publishing audio + video
3. Start a room composite egress with SRT output:

{
  "room_composite": {
    "room_name": "test-room",
    "layout": "grid",
    "stream_outputs": [
      {
        "protocol": 2,
        "urls": ["srt://127.0.0.1:6001?streamid=#!::h=live,m=publish"]
      }
    ]
  }
}

4. Check lalmax server logs — connection is accepted but readBytes=0
5. Egress keeps reconnecting repeatedly

Expected behavior

SRT stream should transmit data successfully, same as ffmpeg does.

Actual behavior

• ✅ Connection established successfully (lalmax sees the incoming connection)
• ❌ No data transmitted (lalmax reports readBytes=0)
• ❌ Egress reconnects in an infinite loop
• ❌ GStreamer logs: Socket is broken or closed. Trying to reconnect

Version info

Component	Version
LiveKit Server	v1.13.6
LiveKit Egress	v1.14.1
GStreamer (inside egress container)	1.24.x
lalmax SRT server	latest (uses gosrt)
OS	Ubuntu 22.04

Additional context

1. ffmpeg works fine with the same server:

ffmpeg -re -i test.mp4 -c copy -f mpegts 'srt://host:6001?streamid=#!::h=live,m=publish'

This pushes successfully, and the stream can be played back via ffplay/VLC.

2. Direct GStreamer srtsink test also fails:

gst-launch-1.0 videotestsrc ! x264enc ! mpegtsmux ! srtsink uri="srt://host:6001?streamid=#!::h=live,m=publish"

Error: Socket is broken or closed. Trying to reconnect

This confirms the issue is between GStreamer srtsink and gosrt, not specific to LiveKit's egress pipeline.

3. streamid not passed via URL query parameter:

GStreamer srtsink does not pass the streamid query parameter from the URL to the SRT server. The server receives an empty streamid. Setting streamid as a GStreamer property (streamid="#!::h=live,m=publish") does pass the ID correctly, but data transmission still fails.

4. RTMP works fine:

Pushing via RTMP (rtmp://host:1936/live/stream) to the same lalmax server works correctly.

Possible cause

Compatibility issue between GStreamer's srtsink (which uses libsrt) and the gosrt library's SRT implementation. The SRT handshake completes successfully, but data transmission fails. Possible reasons:

• Different default latency settings (GStreamer default 120ms vs gosrt default 300ms)
• Different TSBPD (Timestamp-Based Packet Delivery) mode handling
• SRT protocol version mismatch
• Stream ID format/handling differences

Server-side logs (lalmax)

INFO srt connection request: remoteAddr=127.0.0.1:xxxxx, rawStreamId=""
INFO srt streamid is empty, use config default streamName=live
INFO srt AddCustomizePubSession success: streamName=live, sessionKey=CUSTOMIZEPUBxxxx
INFO srt publisher finished normally: streamName=live, remoteAddr=127.0.0.1:xxxxx, readBytes=0  ← no data!

Egress-side logs (GStreamer)

[srtobject warning] gst_srt_object_write_one: warning: Socket is broken or closed. Trying to reconnect
[basesink warning] gst_base_sink_chain_main: Push on pad srtsink_xxx:sink, but it was not activated in push mode

References

• lalmax SRT server: https://github.com/q191201771/lalmax
• gosrt library: https://github.com/datarhei/gosrt
• GStreamer srtsink docs: https://gstreamer.freedesktop.org/documentation/srt/srtsink.html
• SRT protocol: https://github.com/Haivision/srt

[124111294]

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📌 关键信息点

在 issue 中一定要强调这几点：

1. ✅ ffmpeg 可以，GStreamer 不行 — 说明不是 lalmax 的问题
2. ✅ 连接建立了，但没有数据 — readBytes=0
3. ✅ streamid 没有通过 URL 参数传递 — 这是 GStreamer srtsink 的 bug
4. ✅ RTMP 正常 — 说明 Egress 本身没问题
5. ✅ 直接用 gst-launch 测试也失败 — 说明不是 LiveKit 的问题，是 GStreamer srtsink 和 gosrt 的兼容性问题

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🔗 相关链接

可以在 issue 中引用：

• lalmax 项目：https://github.com/q191201771/lalmax
• gosrt 库：https://github.com/datarhei/gosrt
• GStreamer srtsink 文档：https://gstreamer.freedesktop.org/documentation/srt/srtsink.html

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需要我帮你把这个 issue 内容整理成可以直接复制粘贴的格式吗？