RTSP -> LLHLS 404 on first request, second request is successful #2220

infernalxdrm · 2026-06-26T06:44:04Z

infernalxdrm
Jun 26, 2026

Hi, I ran into an issue with LLHLS giving 404 on the first request.
The second request is always successful.
I suspect that it might be happening because the stream initialization hasn't completed yet.
Is there a way to make the client wait until the initialization is done without giving an error?
Thank you in advance!

RTSP source is a Hikvision DVR.
OvenMediaEngine Docker Image Version: 0.20.5

Logs:

[2026-06-26 06:15:10.400] I [SPRtcSig-t3334:35] HTTP.Server | http_connection.cpp:147  | TLS connection accepted : Server Name(media-server) Alpn Protocol(h2) Client (<ClientSocket: 0x7cca83604010, #31, Connected, TCP, Nonblocking, myip:60036>)
[2026-06-26 06:15:10.422] I [SPRtcSig-t3334:35] Publisher | publisher.cpp:179  | Try to pull stream from local origin map: [#default#app/stream]
[2026-06-26 06:15:10.424] I [SPRtcSig-t3334:35] Orchestrator | orchestrator.cpp:612  | Trying to pull stream [#default#app/stream] from provider using origin map: PullProvider
[2026-06-26 06:15:10.429] I [SPRtcSig-t3334:35] RtspcStream | rtspc_stream.cpp:243  | Request Describe : Request: DESCRIBE rtsp:link 1.0
Accept: application/sdp
CSeq: 0
User-Agent: OvenMediaEngine

[2026-06-26 06:15:10.435] I [SPRtcSig-t3334:35] RtspcStream | rtspc_stream.cpp:320  | Response Describe : Response: RTSP/1.0 200 OK
Content-Length: 542
Content-Type: application/sdp
CSeq: 0
Date: Fri, 26 Jun 2026 06:15:09 GMT

[2026-06-26 06:15:10.435] W [SPRtcSig-t3334:35] SDP | common_attr.cpp:211  | Unknown attribute type: a=Media_header:MEDIAINFO=494D4B48020100000400000100000000000000000000000000000000000000000000000000000000;
[2026-06-26 06:15:10.435] W [SPRtcSig-t3334:35] SDP | common_attr.cpp:211  | Unknown attribute type: a=appversion:1.0
[2026-06-26 06:15:10.435] I [SPRtcSig-t3334:35] RtspcStream | rtspc_stream.cpp:364  | SDP : v=0
o=- 1109162014219182 0 IN IP4 0.0.0.0
s=HIK Media Server V4.32.116
i=HIK Media Server Session Description : standard
e=NONE
c=IN IP4 0.0.0.0
t=0 0
a=control:*
b=AS:2058
a=range:npt=now-
m=video 0 RTP/AVP 96
i=Video Media
a=rtpmap:96 H264/90000
a=fmtp:96 profile-level-id=4D0014;packetization-mode=0;sprop-parameter-sets=Z00AH5W4FAFum4CAgKAAAOEAACvyAIA=,aO48gA==
a=control:trackID=video
b=AS:2048
a=Media_header:MEDIAINFO=494D4B48020100000400000100000000000000000000000000000000000000000000000000000000;
a=appversion:1.0


[2026-06-26 06:15:10.436] I [SPRtcSig-t3334:35] RtspcStream | rtspc_stream.cpp:463  | Request SETUP : Request: SETUP rtsp:link 1.0
Authorization: Digest username="admin", realm="2e69984a458ecd56190ead68", nonce="02922eb70", uri="rtsp:link", response="b978964635fd9e160d18f654352671d9"
CSeq: 1
Transport: RTP/AVP/TCP;unicast;interleaved=0-1;ssrc=F10EE538
User-Agent: OvenMediaEngine

[2026-06-26 06:15:10.440] I [SPRtcSig-t3334:35] RtspcStream | rtspc_stream.cpp:479  | Response SETUP : Response: RTSP/1.0 200 OK
Accept-Ranges: NPT
CSeq: 1
Date: Fri, 26 Jun 2026 06:15:09 GMT
Media-Properties: No-Seeking, Time-Progressing, Time-Duration=0
Session: 574387674;timeout=60
Transport: RTP/AVP/TCP;unicast;interleaved=0-1;ssrc=223C75DA

[2026-06-26 06:15:10.440] I [SPRtcSig-t3334:35] RtspcStream | rtspc_stream.cpp:677  | Request PLAY : Request: PLAY rtsp://link 1.0
Authorization: Digest username="admin", realm="2e69984a458ecd56190ead68", nonce="02922eb70", uri="rtsp://link", response="9541b4fee306eb16b9fa9dade9fa8ebf"
CSeq: 2
Session: 574387674
User-Agent: OvenMediaEngine

[2026-06-26 06:15:10.481] I [SPRtcSig-t3334:35] RtspcStream | rtspc_stream.cpp:694  | Response PLAY : Response: RTSP/1.0 200 OK
CSeq: 2
Date: Fri, 26 Jun 2026 06:15:09 GMT
Session: 574387674;timeout=0

[2026-06-26 06:15:10.481] I [SPRtcSig-t3334:35] Provider | stream.cpp:65   | RTSPCProvider Application has started to play [stream(100)] stream : rtsp://link
[2026-06-26 06:15:10.481] I [SPRtcSig-t3334:35] Provider | stream.cpp:51   | #default#app/stream(100) has been started stream
[2026-06-26 06:15:10.482] I [SPRtcSig-t3334:35] MediaRouter | mediarouter_application.cpp:342  | [#default#app/stream(100)] Trying to create a stream
[2026-06-26 06:15:10.482] I [SPRtcSig-t3334:35] Monitor | application_metrics.cpp:58   | Create StreamMetrics(stream/b01cd20d-b438-47e6-826a-e2ad6a12ae19/#default#app/stream/i) for monitoring
[2026-06-26 06:15:10.482] I [SPRtcSig-t3334:35] MediaRouter | mediarouter_application.cpp:458  | [#default#app/stream(100)] Stream has been created
[2026-06-26 06:15:10.501] I [SPRtcSig-t3334:35] Transcoder | transcoder_stream.cpp:75   | [#default#app/stream(100)] stream has been started
[2026-06-26 06:15:10.502] I [SPRtcSig-t3334:35] Provider | application.cpp:201  | #default#app application has created 0 StreamMotor
[2026-06-26 06:15:10.502] I [SPRtcSig-t3334:35] Provider | stream_motor.cpp:137  | #default#app/stream(100) stream has added to 0 StreamMotor
[2026-06-26 06:15:10.502] I [SPRtcSig-t3334:35] Provider | application.cpp:209  | #default#app/stream(100) stream has been added to StreamMotor 0
[2026-06-26 06:15:10.502] E [SPRtcSig-t3334:35] LLHLS Publisher | llhls_publisher.cpp:380  | Cannot find stream (#default#app/stream)
[2026-06-26 06:15:10.503] I [StreamMotor:67] RtpRtcp | rtp_rtcp.cpp:350  | Connect SSRC(574387674) to track ID(0)
[2026-06-26 06:15:10.503] E [SPRtcSig-t3334:35] HTTP.Server | http_exchange.cpp:45   |
[Client] HttpConnection(0x7cca83602210) : HTTP/2 <ClientSocket: 0x7cca83604010, #31, Connected, TCP, Nonblocking, myip:60036> TLS(Enabled) (Elapsed: 101)
[Request] GET https://media-server (HTTP/2, Request Time: 2026-06-26T06:15:10.401+00:00)
[Response] 404 Not Found (124 bytes sent, Response Time: 2026-06-26T06:15:10.502+00:00)
[2026-06-26 06:15:10.505] I [StreamMotor:67] Provider | stream.cpp:437  | Since this stream has a single track, it computes PTS alone without RTCP SR.
[2026-06-26 06:15:10.578] I [SPRtcSig-t3334:35] HTTP.Server | http_server.cpp:223  | Client(<ClientSocket: 0x7cca83604010, #31, Disconnected, TCP, Nonblocking, myip:60036>) has disconnected from *:3334
[2026-06-26 06:15:11.493] I [InboundWorker:59] MediaRouter | mediarouter_application.cpp:497  | [#default#app/stream(100)] Stream has been prepared
[Stream Info]
id(100), msid(0), output(stream), SourceType(RtspPull), RepresentationType(Source), Created Time (Fri Jun 26 06:15:10 2026) UUID(b01cd20d-b438-47e6-826a-e2ad6a12ae19/#default#app/stream/i)

        Video Track #0: Public Name(Video_0) Variant Name(Video) Bitrate(1.64Mb) Codec(H264,none:0) BSF(H264_RTP_RFC_6184) Resolution(1280x720) MaxResolution(1280x720) Framerate(25.99) MaxFramerate(25.99) KeyInterval(0.00/frame) SkipFrames(-1) BFrames(0) timebase(1/90000)
        Data  Track #1: Public Name(Data_1) Variant Name(Data) Codec(None,none) BSF(Unknown) timebase(1/1000)
[2026-06-26 06:15:11.494] I [InboundWorker:68] Transcoder | transcoder_stream.cpp:236  | [#default#app/stream(100)] Using local output profiles by webhook. Response time: 0 ms
[2026-06-26 06:15:11.495] I [InboundWorker:68] Transcoder | transcoder_stream.cpp:1004 | [[#default#app/stream(100)]] Playlist(default) : master
        Default : false
        WebRTC Auto ABR : true
        HLS Chunklist Path Depth : 0
        TS Packaging : true
        720p
                Video Variant : bypass_video
                Video Index Hint : 0
                Audio Variant :
                Audio Index Hint : -1
        1080p
                Video Variant : video_1280
                Video Index Hint : 0
                Audio Variant :
                Audio Index Hint : -1
        720p_1
                Video Variant : video_720
                Video Index Hint : 0
                Audio Variant :
                Audio Index Hint : -1

[2026-06-26 06:15:11.495] I [InboundWorker:68] Transcoder | transcoder_stream.cpp:1004 | [[#default#app/stream(100)]] Playlist(For LLHLS) : stream
        Default : false
        WebRTC Auto ABR : true
        HLS Chunklist Path Depth : -1
        TS Packaging : true
        Bypass
                Video Variant : bypass_video
                Video Index Hint : -1
                Audio Variant : bypass_audio
                Audio Index Hint : -1
        FHD
                Video Variant : video_1280
                Video Index Hint : -1
                Audio Variant : bypass_audio
                Audio Index Hint : -1
        HD
                Video Variant : video_720
                Video Index Hint : -1
                Audio Variant : bypass_audio
                Audio Index Hint : -1

[2026-06-26 06:15:11.496] I [InboundWorker:68] Transcoder | transcoder_stream.cpp:653  | [#default#app/stream(100)] Output stream has been created. [#default#app/stream(1124356501)]
[2026-06-26 06:15:11.496] I [InboundWorker:68] MediaRouter | mediarouter_application.cpp:342  | [#default#app/stream(1124356501)] Trying to create a stream
[2026-06-26 06:15:11.496] I [InboundWorker:68] Monitor | application_metrics.cpp:58   | Create StreamMetrics(stream/b01cd20d-b438-47e6-826a-e2ad6a12ae19/#default#app/stream/o) for monitoring
[2026-06-26 06:15:11.496] I [InboundWorker:68] MediaRouter | mediarouter_application.cpp:458  | [#default#app/stream(1124356501)] Stream has been created
[2026-06-26 06:15:11.496] I [InboundWorker:68] Publisher | application.cpp:69   | Stream(stream/1124356501) created on AppWorker (WebRTC / 0)
[2026-06-26 06:15:11.496] I [InboundWorker:68] Publisher | application.cpp:69   | Stream(stream/1124356501) created on AppWorker (LLHLS / 0)
[2026-06-26 06:15:11.496] I [InboundWorker:68] Publisher | application.cpp:69   | Stream(stream/1124356501) created on AppWorker (OVT / 0)
[2026-06-26 06:15:11.589] I [InboundWorker:68] Transcoder | transcoder_stream.cpp:154  | [#default#app/stream(100)] stream has been prepared
[2026-06-26 06:15:11.674] W [OutboundWorker:60] MediaRouter | mediarouter_stream.cpp:272  | [#default#app/stream] Detected out of order DTS of packet. track_id:0 dts:0->0
[2026-06-26 06:15:11.674] E [dec_H264_t0:69] FFmpeg | third_parties.cpp:113  | [AVCodecContext: 0x7cca66201000] missing picture in access unit with size 41
[2026-06-26 06:15:11.676] E [dec_H264_t0:70] FFmpeg | third_parties.cpp:113  | [AVCodecContext: 0x7cca66201400] no frame!
[2026-06-26 06:15:11.691] E [dec_H264_t0:70] FFmpeg | third_parties.cpp:113  | [AVCodecContext: 0x7cca66201400] reference picture missing during reorder
[2026-06-26 06:15:11.692] E [dec_H264_t0:70] FFmpeg | third_parties.cpp:113  | [AVCodecContext: 0x7cca66201400] Missing reference picture, default is 2
[2026-06-26 06:15:11.801] W [ENC-H264-t2:72] FFmpeg | third_parties.cpp:117  | [AVCodecContext: 0x7cca63204000] [OpenH264] this = 0x0x7cca63208030, Warning:layerId(0) doesn't support profile(578), change to UNSPECIFIC profile
[2026-06-26 06:15:11.810] W [ENC-H264-t2:72] FFmpeg | third_parties.cpp:117  | [AVCodecContext: 0x7cca63204000] [OpenH264] this = 0x0x7cca63208030, Warning:bEnableFrameSkip = 0,bitrate can't be controlled for RC_QUALITY_MODE,RC_BITRATE_MODE and RC_TIMESTAMP_MODE without enabling skip frame.
[2026-06-26 06:15:11.838] W [ENC-H264-t1:78] FFmpeg | third_parties.cpp:117  | [AVCodecContext: 0x7cca5e204000] [OpenH264] this = 0x0x7cca5e20a000, Warning:layerId(0) doesn't support profile(578), change to UNSPECIFIC profile
[2026-06-26 06:15:11.838] W [ENC-H264-t1:78] FFmpeg | third_parties.cpp:117  | [AVCodecContext: 0x7cca5e204000] [OpenH264] this = 0x0x7cca5e20a000, Warning:bEnableFrameSkip = 0,bitrate can't be controlled for RC_QUALITY_MODE,RC_BITRATE_MODE and RC_TIMESTAMP_MODE without enabling skip frame.
[2026-06-26 06:15:11.932] W [FLT-rscl-t2:79] FFmpeg | third_parties.cpp:117  | [SWScaler: 0x7cca59a56c80] deprecated pixel format used, make sure you did set range correctly
[2026-06-26 06:15:11.936] W [FLT-rscl-t2:79] FFmpeg | third_parties.cpp:117  | [SWScaler: 0x7cca59aefa00] deprecated pixel format used, make sure you did set range correctly
[2026-06-26 06:15:11.936] W [FLT-rscl-t2:79] FFmpeg | third_parties.cpp:117  | [SWScaler: 0x7cca59b2a000] deprecated pixel format used, make sure you did set range correctly
[2026-06-26 06:15:11.937] W [FLT-rscl-t2:79] FFmpeg | third_parties.cpp:117  | [SWScaler: 0x7cca59b5f100] deprecated pixel format used, make sure you did set range correctly
[2026-06-26 06:15:11.940] W [FLT-rscl-t1:86] FFmpeg | third_parties.cpp:117  | [SWScaler: 0x7cca55c567c0] deprecated pixel format used, make sure you did set range correctly
[2026-06-26 06:15:11.943] W [FLT-rscl-t1:86] FFmpeg | third_parties.cpp:117  | [SWScaler: 0x7cca55d36080] deprecated pixel format used, make sure you did set range correctly
[2026-06-26 06:15:11.945] W [FLT-rscl-t1:86] FFmpeg | third_parties.cpp:117  | [SWScaler: 0x7cca55da0f40] deprecated pixel format used, make sure you did set range correctly
[2026-06-26 06:15:11.947] W [FLT-rscl-t1:86] FFmpeg | third_parties.cpp:117  | [SWScaler: 0x7cca5298e040] deprecated pixel format used, make sure you did set range correctly
[2026-06-26 06:15:11.951] W [FLT-rscl-t1:86] FFmpeg | third_parties.cpp:117  | [SWScaler: 0x7cca55c56140] deprecated pixel format used, make sure you did set range correctly
[2026-06-26 06:15:11.953] W [FLT-rscl-t1:86] FFmpeg | third_parties.cpp:117  | [SWScaler: 0x7cca55cc3300] deprecated pixel format used, make sure you did set range correctly
[2026-06-26 06:15:11.954] W [FLT-rscl-t1:86] FFmpeg | third_parties.cpp:117  | [SWScaler: 0x7cca55d5e200] deprecated pixel format used, make sure you did set range correctly
[2026-06-26 06:15:11.955] W [FLT-rscl-t1:86] FFmpeg | third_parties.cpp:117  | [SWScaler: 0x7cca52989340] deprecated pixel format used, make sure you did set range correctly
[2026-06-26 06:15:11.957] W [FLT-rscl-t1:86] FFmpeg | third_parties.cpp:117  | [SWScaler: 0x7cca55c4e640] deprecated pixel format used, make sure you did set range correctly
[2026-06-26 06:15:11.958] W [FLT-rscl-t1:86] FFmpeg | third_parties.cpp:117  | [SWScaler: 0x7cca52989700] deprecated pixel format used, make sure you did set range correctly
[2026-06-26 06:15:11.959] W [FLT-rscl-t1:86] FFmpeg | third_parties.cpp:117  | [SWScaler: 0x7cca55d1f780] deprecated pixel format used, make sure you did set range correctly
[2026-06-26 06:15:11.960] W [FLT-rscl-t1:86] FFmpeg | third_parties.cpp:117  | [SWScaler: 0x7cca52998400] deprecated pixel format used, make sure you did set range correctly
[2026-06-26 06:15:11.962] W [FLT-rscl-t1:86] FFmpeg | third_parties.cpp:117  | [SWScaler: 0x7cca55c4e5c0] deprecated pixel format used, make sure you did set range correctly
[2026-06-26 06:15:11.963] W [FLT-rscl-t1:86] FFmpeg | third_parties.cpp:117  | [SWScaler: 0x7cca52989100] deprecated pixel format used, make sure you did set range correctly
[2026-06-26 06:15:11.964] W [FLT-rscl-t1:86] FFmpeg | third_parties.cpp:117  | [SWScaler: 0x7cca529e45c0] deprecated pixel format used, make sure you did set range correctly
[2026-06-26 06:15:11.965] W [FLT-rscl-t1:86] FFmpeg | third_parties.cpp:117  | [SWScaler: 0x7cca55d59cc0] deprecated pixel format used, make sure you did set range correctly
[2026-06-26 06:15:11.977] W [FLT-rscl-t2:79] FFmpeg | third_parties.cpp:117  | [SWScaler: 0x7cca59a56ec0] deprecated pixel format used, make sure you did set range correctly
[2026-06-26 06:15:11.977] W [FLT-rscl-t2:79] FFmpeg | third_parties.cpp:117  | [SWScaler: 0x7cca59aef680] deprecated pixel format used, make sure you did set range correctly
[2026-06-26 06:15:11.978] W [FLT-rscl-t2:79] FFmpeg | third_parties.cpp:117  | [SWScaler: 0x7cca59b5fa40] deprecated pixel format used, make sure you did set range correctly
[2026-06-26 06:15:11.979] W [FLT-rscl-t2:79] FFmpeg | third_parties.cpp:117  | [SWScaler: 0x7cca59b94b80] deprecated pixel format used, make sure you did set range correctly
[2026-06-26 06:15:11.980] W [FLT-rscl-t2:79] FFmpeg | third_parties.cpp:117  | [SWScaler: 0x7cca59a56b00] deprecated pixel format used, make sure you did set range correctly
[2026-06-26 06:15:11.980] W [FLT-rscl-t2:79] FFmpeg | third_parties.cpp:117  | [SWScaler: 0x7cca59aef680] deprecated pixel format used, make sure you did set range correctly
[2026-06-26 06:15:11.981] W [FLT-rscl-t2:79] FFmpeg | third_parties.cpp:117  | [SWScaler: 0x7cca59b53840] deprecated pixel format used, make sure you did set range correctly
[2026-06-26 06:15:11.982] W [FLT-rscl-t2:79] FFmpeg | third_parties.cpp:117  | [SWScaler: 0x7cca59b74900] deprecated pixel format used, make sure you did set range correctly
[2026-06-26 06:15:11.983] W [FLT-rscl-t2:79] FFmpeg | third_parties.cpp:117  | [SWScaler: 0x7cca59a56b40] deprecated pixel format used, make sure you did set range correctly
[2026-06-26 06:15:11.984] W [FLT-rscl-t2:79] FFmpeg | third_parties.cpp:117  | [SWScaler: 0x7cca59aef500] deprecated pixel format used, make sure you did set range correctly
[2026-06-26 06:15:11.984] W [FLT-rscl-t2:79] FFmpeg | third_parties.cpp:117  | [SWScaler: 0x7cca59b42ec0] deprecated pixel format used, make sure you did set range correctly
[2026-06-26 06:15:11.985] W [FLT-rscl-t2:79] FFmpeg | third_parties.cpp:117  | [SWScaler: 0x7cca59b51bc0] deprecated pixel format used, make sure you did set range correctly
[2026-06-26 06:15:12.040] I [OutboundWorker:60] MediaRouter | mediarouter_application.cpp:497  | [#default#app/stream(1124356501)] Stream has been prepared
[Stream Info]
id(1124356501), msid(0), output(stream), SourceType(Transcoder), RepresentationType(Source), Created Time (Fri Jun 26 06:15:11 2026) UUID(b01cd20d-b438-47e6-826a-e2ad6a12ae19/#default#app/stream/o)
        >> Origin Stream Info
        id(100), output(stream), SourceType(RtspPull), Created Time (Fri Jun 26 06:15:10 2026)

        Video Track #0: Public Name(Video_0) Variant Name(bypass_video) Bitrate(1.64Mb) Codec(H264,Passthrough:0) BSF(H264_RTP_RFC_6184) Resolution(1280x720) MaxResolution(1280x720) Framerate(25.99) MaxFramerate(25.99) KeyInterval(0.00/frame) SkipFrames(-1) BFrames(0) timebase(1/90000)
        Video Track #1: Public Name(Video_0) Variant Name(video_1280) Bitrate(5.02Mb) Codec(H264,openh264:0) BSF(H264_ANNEXB) Resolution(1920x1080) MaxResolution(1920x1080) Framerate(30.00) MaxFramerate(30.00) KeyInterval(30.00/frame) SkipFrames(-1) BFrames(0) timebase(1/90000)
        Video Track #2: Public Name(Video_0) Variant Name(video_720) Bitrate(2.02Mb) Codec(H264,openh264:0) BSF(H264_ANNEXB) Resolution(1280x720) MaxResolution(1280x720) Framerate(30.00) MaxFramerate(30.00) KeyInterval(30.00/frame) SkipFrames(-1) BFrames(0) timebase(1/90000)
        Data  Track #3: Public Name(Data_1) Variant Name(Data) Codec(None,Passthrough) BSF(Unknown) timebase(1/1000)
[2026-06-26 06:15:12.042] I [OutboundWorker:60] WebRTC Publisher | rtc_stream.cpp:240  | WebRTC Stream has been created : stream/1124356501
Rtx(false) Ulpfec(false) JitterBuffer(false) PlayoutDelay(false min:0 max: 0)
[2026-06-26 06:15:12.042] I [OutboundWorker:60] Publisher | stream.cpp:267  | WebRTC Publisher Application has started [stream(1124356501)] stream (MSID : 0)
[2026-06-26 06:15:12.044] I [OutboundWorker:60] LLHLS Publisher | llhls_stream.cpp:1409 | LLHlsStream::ComputeOptimalPartDuration() - Video track(0) FrameRate(25.986526) frame_duration_ms(38.481481) optimal_frame_count(52.000000) part_target(2000.000000) optimal_part_target(2001.037037)
[2026-06-26 06:15:12.044] I [OutboundWorker:60] LLHLS Publisher | llhls_stream.cpp:1434 | LLHlsStream::AddPackager() - Track(0) ChunkDuration(2001.000000)
[2026-06-26 06:15:12.044] I [OutboundWorker:60] LLHLS Publisher | llhls_stream.cpp:1409 | LLHlsStream::ComputeOptimalPartDuration() - Video track(1) FrameRate(30.000000) frame_duration_ms(33.333333) optimal_frame_count(60.000000) part_target(2000.000000) optimal_part_target(2000.000000)
[2026-06-26 06:15:12.044] I [OutboundWorker:60] LLHLS Publisher | llhls_stream.cpp:1434 | LLHlsStream::AddPackager() - Track(1) ChunkDuration(2000.000000)
[2026-06-26 06:15:12.044] I [OutboundWorker:60] LLHLS Publisher | llhls_stream.cpp:1409 | LLHlsStream::ComputeOptimalPartDuration() - Video track(2) FrameRate(30.000000) frame_duration_ms(33.333333) optimal_frame_count(60.000000) part_target(2000.000000) optimal_part_target(2000.000000)
[2026-06-26 06:15:12.044] I [OutboundWorker:60] LLHLS Publisher | llhls_stream.cpp:1434 | LLHlsStream::AddPackager() - Track(2) ChunkDuration(2000.000000)
[2026-06-26 06:15:12.045] I [OutboundWorker:60] LLHLS Publisher | llhls_stream.cpp:291  | LLHlsStream has been created : stream/1124356501
OriginMode(false) Chunk Duration(2.00) Segment Duration(6.00) Segment Count(10) DRM(none)
[2026-06-26 06:15:12.045] I [OutboundWorker:60] Publisher | stream.cpp:267  | LLHLS Publisher Application has started [stream(1124356501)] stream (MSID : 0)
[2026-06-26 06:15:12.046] I [OutboundWorker:60] Publisher | stream.cpp:267  | OVTPublisher Application has started [stream(1124356501)] stream (MSID : 0)
[2026-06-26 06:15:15.379] I [SPRtcSig-t3334:35] HTTP.Server | http_server.cpp:149  | Client(<ClientSocket: 0x7cca83604010, #31, Connected, TCP, Nonblocking, myip:60037>) is connected on *:3334
[2026-06-26 06:15:15.458] I [SPRtcSig-t3334:35] HTTP.Server | http_connection.cpp:147  | TLS connection accepted : Server Name(media-server) Alpn Protocol(h2) Client (<ClientSocket: 0x7cca83604010, #31, Connected, TCP, Nonblocking, myip:60037>)
[2026-06-26 06:15:16.658] I [AW-LLHLS0:62] LLHLS Publisher | llhls_stream.cpp:1768 | LLHlsStream(#default#app/stream) - Ready to play : Part Hold Back = 6.000000
[2026-06-26 06:15:18.565] W [SW-LLHLS:126] LLHLS Publisher | llhls_stream.cpp:626  | LLHlsStream(#default#app/stream) - bypass_audio audio is excluded from the Bypass rendition in stream playlist because there is no audio track.
[2026-06-26 06:15:18.566] W [SW-LLHLS:126] LLHLS Publisher | llhls_stream.cpp:626  | LLHlsStream(#default#app/stream) - bypass_audio audio is excluded from the FHD rendition in stream playlist because there is no audio track.
[2026-06-26 06:15:18.566] W [SW-LLHLS:126] LLHLS Publisher | llhls_stream.cpp:626  | LLHlsStream(#default#app/stream) - bypass_audio audio is excluded from the HD rendition in stream playlist because there is no audio track.
[2026-06-26 06:15:18.567] I [SW-LLHLS:126] Monitor | stream_metrics.cpp:187  | A new session has started playing #default#app/stream on the LLHLS publisher. LLHLS(1)/Stream total(1)/App total(1)
[2026-06-26 06:15:18.829] I [SPRtcSig-t3334:35] HTTP.Server | http_server.cpp:149  | Client(<ClientSocket: 0x7cca83604510, #35, Connected, TCP, Nonblocking, myip:60038>) is connected on *:3334
[2026-06-26 06:15:18.906] I [SPRtcSig-t3334:35] HTTP.Server | http_connection.cpp:147  | TLS connection accepted : Server Name(media-server) Alpn Protocol(h2) Client (<ClientSocket: 0x7cca83604510, #35, Connected, TCP, Nonblocking, myip:60038>)

Server.xml


<Server version="8">
	<Name>OvenMediaEngine</Name>
	<!-- Host type (origin/edge) -->
	<Type>origin</Type>
	<!-- Specify IP address to bind ("*" means all IPv4 IPs, "::" means all IPv6 IPs) -->
	<!-- Uncomment the line below to enable IPv6 -->
	<!-- <IP>::</IP> -->
	<IP>*</IP>
	<PrivacyProtection>false</PrivacyProtection>

	<!-- 
	To get the public IP address(mapped address of stun) of the local server. 
	This is useful when OME cannot obtain a public IP from an interface, such as AWS or docker environment. 
	If this is successful, you can use ${PublicIP} in your settings.
	-->
	<StunServer>stun.ovenmediaengine.com:13478</StunServer>

	<Modules>
		<!-- 
		Currently OME only supports h2 like all browsers do. Therefore, HTTP/2 only works on TLS ports.			
		-->
		<HTTP2>
			<Enable>true</Enable>
		</HTTP2>

		<LLHLS>
			<Enable>true</Enable>
		</LLHLS>

		<!-- P2P works only in WebRTC and is experiment feature -->
		<P2P>
			<!-- disabled by default -->
			<Enable>false</Enable>
			<MaxClientPeersPerHostPeer>2</MaxClientPeersPerHostPeer>
		</P2P>
	</Modules>

	<!-- Settings for the ports to bind -->
	<Bind>
		<!-- Enable this configuration if you want to use API Server -->
		<!--
		<Managers>
			<API>
				<Port>8081</Port>
				<TLSPort>8082</TLSPort>
				<WorkerCount>1</WorkerCount>
			</API>
		</Managers>
		-->

		<Providers>
			<!-- Pull providers -->
			<RTSPC>
				<WorkerCount>1</WorkerCount>
			</RTSPC>
			<OVT>
				<WorkerCount>1</WorkerCount>
			</OVT>
			<!-- Push providers -->
			<RTMP>
				<Port>1936</Port>
				<WorkerCount>1</WorkerCount>
			</RTMP>
			<SRT>
				<Port>9996</Port>
				<WorkerCount>1</WorkerCount>
			</SRT>
			<MPEGTS>
				<!--
					Below is an example of listening on port 4000-4005 (<Port>4000-4004,4005/udp</Port>)
					This is just a demonstration to show that you can configure the port in several ways
				
					<Port>4000-4004,4005/udp</Port>
				-->
				<Port>4000/udp</Port>
			</MPEGTS>
			<WebRTC>
				<Signalling>
					<Port>3333</Port>
					<TLSPort>3334</TLSPort>
					<WorkerCount>1</WorkerCount>
				</Signalling>

				<IceCandidates>
					<!-- Uncomment the line below to use IPv6 ICE Candidate -->
					<!-- <IceCandidate>[::]:10000-10004/udp</IceCandidate> -->
					<IceCandidate>*:10000-10004/udp</IceCandidate>

					<!-- Uncomment the line below to use a link local address when specifying an address with IPv6 wildcard (::) -->
					<!-- <EnableLinkLocalAddress>true</EnableLinkLocalAddress> -->

					<!-- 
						If you want to stream WebRTC over TCP, specify IP:Port for TURN server.
						This uses the TURN protocol, which delivers the stream from the built-in TURN server to the player's TURN client over TCP. 
					-->
					<TcpRelay>*:3478</TcpRelay>
					<!-- TcpForce is an option to force the use of TCP rather than UDP in WebRTC streaming. (You can omit ?transport=tcp accordingly.) If <TcpRelay> is not set, playback may fail. -->
					<TcpForce>true</TcpForce>
					<TcpRelayWorkerCount>1</TcpRelayWorkerCount>
				</IceCandidates>
			</WebRTC>
		</Providers>

		<Publishers>
			<OVT>
				<Port>9001</Port>
				<WorkerCount>1</WorkerCount>
			</OVT>
			<LLHLS>
				<!-- 
				OME only supports h2, so LLHLS works over HTTP/1.1 on non-TLS ports. 
				LLHLS works with higher performance over HTTP/2, 
				so it is recommended to use a TLS port.
				-->
				<Port>3333</Port>
				<!-- If you want to use TLS, specify the TLS port -->
				<TLSPort>3334</TLSPort>
				<WorkerCount>1</WorkerCount>
			</LLHLS>
			<WebRTC>
				<Signalling>
					<Port>3333</Port>
					<TLSPort>3334</TLSPort>
					<WorkerCount>1</WorkerCount>
				</Signalling>
				<IceCandidates>
					<!-- Uncomment the line below to use IPv6 ICE Candidate -->
					<!-- <IceCandidate>[::]:10000-10004/udp</IceCandidate> -->
					<IceCandidate>*:10000-10004/udp</IceCandidate>
					
					<!-- Uncomment the line below to use a link local address when specifying an address with IPv6 wildcard (::) -->
					<!-- <EnableLinkLocalAddress>true</EnableLinkLocalAddress> -->

					<!-- 
						If you want to stream WebRTC over TCP, specify IP:Port for TURN server.
						This uses the TURN protocol, which delivers the stream from the built-in TURN server to the player's TURN client over TCP. 
					-->
					<TcpRelay>*:3478</TcpRelay>
					<!-- TcpForce is an option to force the use of TCP rather than UDP in WebRTC streaming. (You can omit ?transport=tcp accordingly.) If <TcpRelay> is not set, playback may fail. -->
					<TcpForce>true</TcpForce>
					<TcpRelayWorkerCount>1</TcpRelayWorkerCount>
				</IceCandidates>
			</WebRTC>
			<SRT>
				<Port>9998</Port>
				<WorkerCount>1</WorkerCount>
			</SRT>
		</Publishers>
	</Bind>

	<!--
		Enable this configuration if you want to use API Server
		
		<AccessToken> is a token for authentication, and when you invoke the API, you must put "Basic base64encode(<AccessToken>)" in the "Authorization" header of HTTP request.
		For example, if you set <AccessToken> to "ome-access-token", you must set "Basic b21lLWFjY2Vzcy10b2tlbg==" in the "Authorization" header.
	-->
	<!--
	<Managers>
		<Host>
			<Names>
				<Name>*</Name>
			</Names>
			<TLS>
				<CertPath>path/to/file.crt</CertPath>
				<KeyPath>path/to/file.key</KeyPath>
				<ChainCertPath>path/to/file.crt</ChainCertPath>
			</TLS>
		</Host>
		<API>
			<AccessToken>ome-access-token</AccessToken>

			<CrossDomains>
				<Url>*.airensoft.com</Url>
				<Url>http://*.sub-domain.airensoft.com</Url>
				<Url>http?://airensoft.*</Url>
			</CrossDomains>
		</API>
	</Managers>
	-->

	<!--
		Refer https://airensoft.gitbook.io/ovenmediaengine/alert
	<Alert>
		<Url></Url>
		<SecretKey></SecretKey>
		<Timeout>3000</Timeout>
		<Rules>
			<Ingress>
				<MinBitrate></MinBitrate>
				<MaxBitrate></MaxBitrate>
				<MinFramerate></MinFramerate>
				<MaxFramerate></MaxFramerate>
				<MinWidth></MinWidth>
				<MinHeight></MinHeight>
				<MaxWidth></MaxWidth>
				<MaxHeight></MaxHeight>
				<MinSamplerate></MinSamplerate>
				<MaxSamplerate></MaxSamplerate>
				<LongKeyFrameInterval />
				<HasBFrames />
			</Ingress>
		</Rules>
	</Alert>
	-->

	<!-- Default Settings for all Virtual Hosts -->
	<Defaults>
		<CrossDomains>
			<Url>*</Url>
		</CrossDomains>
	</Defaults>

	<VirtualHosts>
		<!-- You can use wildcard like this to include multiple XMLs -->
		<VirtualHost include="VHost*.xml" />
		<VirtualHost>
			<Name>default</Name>
			<!--Distribution is a value that can be used when grouping the same vhost distributed across multiple servers. This value is output to the events log, so you can use it to aggregate statistics. -->
			<Distribution>media-server</Distribution>

			<!-- Settings for multi ip/domain and TLS -->
			<Host>
				<Names>
					<!-- Host names
						<Name>stream1.airensoft.com</Name>
						<Name>stream2.airensoft.com</Name>
						<Name>*.sub.airensoft.com</Name>
						<Name>192.168.0.1</Name>
					-->
					<Name>media-server.domain.name</Name>
					<Name>media-server.ip.address</Name>
				</Names>
				<TLS>
					<CertPath>/opt/ovenmediaengine/bin/certs/cert.crt</CertPath>
					<KeyPath>/opt/ovenmediaengine/bin/certs/cert.key</KeyPath>
					<ChainCertPath>/opt/ovenmediaengine/bin/certs/cert.crt</ChainCertPath>
				</TLS>
			</Host>

			<!-- 	
			Refer https://airensoft.gitbook.io/ovenmediaengine/signedpolicy
			<SignedPolicy>
				<PolicyQueryKeyName>policy</PolicyQueryKeyName>
				<SignatureQueryKeyName>signature</SignatureQueryKeyName>
				<SecretKey>aKq#1kj</SecretKey>

				<Enables>
					<Providers>rtmp,webrtc,srt</Providers>
					<Publishers>webrtc,llhls</Publishers>
				</Enables>
			</SignedPolicy>
			-->

			<!--
			<AdmissionWebhooks>
				<ControlServerUrl></ControlServerUrl>
				<SecretKey></SecretKey>
				<Timeout>3000</Timeout>
				<Enables>
					<Providers>rtmp,webrtc,srt</Providers>
					<Publishers>webrtc,llhls</Publishers>
				</Enables>
			</AdmissionWebhooks>
			-->

			<Origins>
				<Properties>
					<NoInputFailoverTimeout>3000</NoInputFailoverTimeout>
					<UnusedStreamDeletionTimeout>40000</UnusedStreamDeletionTimeout>
				</Properties>
				<Origin>
					<Location>/app/stream</Location>
					<Pass>
						<Scheme>rtsp</Scheme>
						<Urls><Url>link</Url></Urls>
					</Pass>
				</Origin>
				<!--
				<Origin>
					<Location>/app/</Location>
					<Pass>
						<Scheme>ovt</Scheme>
						<Urls><Url>origin.com:9000/app/</Url></Urls>
					</Pass>
				</Origin>
				<Origin>
					<Location>/edge/</Location>
					<Pass>
						<Scheme>ovt</Scheme>
						<Urls><Url>origin.com:9000/app/</Url></Urls>
					</Pass>
				</Origin>
				-->
			</Origins>

			<!--
			<OriginMapStore>
				In order to use OriginMap, you must enable OVT Publisher in Origin and OVT Provider in Edge.
				<RedisServer>
					<Host>192.168.0.160:6379</Host>
					<Auth>!@#ovenmediaengine</Auth>
				</RedisServer>
				
				This is only needed for the origin server and used to register the ovt address of the stream. 
				<OriginHostName>ome-dev.airensoft.com</OriginHostName>
			</OriginMapStore>
			-->

			<!-- Settings for applications -->
			<Applications>
				<Application>
					<Name>app</Name>
					<!-- Application type (live/vod) -->
					<Type>live</Type>
					<OutputProfiles>
						<!-- Common setting for decoders. Decodes is optional. -->
						<Decodes>
							<!-- Number of threads for the decoder. -->
							<ThreadCount>2</ThreadCount>
							<!-- 
							By default, OME decodes all video frames. 
							With OnlyKeyframes, only keyframes are decoded, massively improving performance.
							Thumbnails are generated only on keyframes, they may not generate at your requested fps!
							-->
							<OnlyKeyframes>false</OnlyKeyframes>
						</Decodes>

						<!-- Enable this configuration if you want to hardware acceleration using GPU -->
						<HWAccels>
							<Decoder>
									<Enable>false</Enable>
									<!-- 
										Setting for Hardware Modules.
										 - xma :Xilinx Media Accelerator
										 - nv : Nvidia Video Codec SDK
										 - nilogan: Netint VPU

										You can use multiple modules by separating them with commas.
										For example, if you want to use xma and nv, you can set it as follows.

										<Modules>[ModuleName]:[DeviceId],[ModuleName]:[DeviceId],...</Modules>
										<Modules>xma:0,nv:0</Modules>
									-->
									<!-- <Modules>nv</Modules> -->
							</Decoder>
							<Encoder>
									<Enable>false</Enable>
									<!-- <Modules>nv</Modules> -->
							</Encoder>
						</HWAccels>

						<OutputProfile>
							<Name>bypass_stream</Name>
							<OutputStreamName>${OriginStreamName}</OutputStreamName>

							<Playlist>
								<Name>default</Name>
								<FileName>master</FileName>
								<Options>
									<WebRtcAutoAbr>true</WebRtcAutoAbr>
									<HLSChunklistPathDepth>0</HLSChunklistPathDepth>
									<EnableTsPackaging>true</EnableTsPackaging>
								</Options>
								<RenditionTemplate>
									<Name>${Height}p</Name>
									<VideoTemplate> <!-- Filter by video properties -->
										<EncodingType>all</EncodingType> <!-- all, bypassed, encoded -->
										<!-- <VariantName>bypass_video</VariantName>
										<VideoIndexHint>0</VideoIndexHint>
										<MaxWidth>1080</MaxWidth>
										<MinWidth>240</MinWidth>
										<MaxHeight>720</MaxHeight>
										<MinHeight>240</MinHeight>
										<MaxFPS>30</MaxFPS>
										<MinFPS>30</MinFPS>
										<MaxBitrate>2000000</MaxBitrate>
										<MinBitrate>500000</MinBitrate> -->
									</VideoTemplate>
									<AudioTemplate> <!-- Filter by audio properties -->
										<EncodingType>all</EncodingType> <!-- all, bypassed, encoded -->
										<!-- <VariantName>aac_audio</VariantName>
										<MaxBitrate>128000</MaxBitrate>
										<MinBitrate>128000</MinBitrate>
										<MaxSamplerate>48000</MaxSamplerate>
										<MinSamplerate>48000</MinSamplerate>
										<MaxChannel>2</MaxChannel>
										<MinChannel>2</MinChannel>
										<AudioIndexHint>0</AudioIndexHint> -->
									</AudioTemplate>
								</RenditionTemplate>
							</Playlist>
							<Playlist>
								<Name>For LLHLS</Name>
								<FileName>stream</FileName>
								<Options> <!-- Optional -->
									<!-- 
									Automatically switch rendition in WebRTC ABR 
									[Default] : true
									-->
									<WebRtcAutoAbr>true</WebRtcAutoAbr> 
									<EnableTsPackaging>true</EnableTsPackaging>
								</Options>
								<Rendition>
									<Name>Bypass</Name>
									<Video>bypass_video</Video>
									<Audio>bypass_audio</Audio>
								</Rendition>
								<Rendition>
									<Name>FHD</Name>
									<Video>video_1280</Video>
									<Audio>bypass_audio</Audio>
								</Rendition>
								<Rendition>
									<Name>HD</Name>
									<Video>video_720</Video>
									<Audio>bypass_audio</Audio>
								</Rendition>
							</Playlist>

							<Encodes>
								<Video>
									<Name>bypass_video</Name>
									<Bypass>true</Bypass>
								</Video>
								<Audio>
									<Name>aac_audio</Name>
									<Codec>aac</Codec>
									<Bitrate>128000</Bitrate>
									<Samplerate>48000</Samplerate>
									<Channel>2</Channel>
									<BypassIfMatch>
										<Codec>eq</Codec>
									</BypassIfMatch>
								</Audio>
								<Audio>
									<Name>opus_audio</Name>
									<Codec>opus</Codec>
									<Bitrate>128000</Bitrate>
									<Samplerate>48000</Samplerate>
									<Channel>2</Channel>
									<BypassIfMatch>
										<Codec>eq</Codec>
									</BypassIfMatch>
								</Audio>
								<Video>
									<Name>video_1280</Name>
									<Codec>h264</Codec>
									<Bitrate>5024000</Bitrate>
									<Width>1920</Width>
									<Height>1080</Height>
									<Framerate>30</Framerate>
									<KeyFrameInterval>30</KeyFrameInterval>
									<BFrames>0</BFrames>
									<Preset>faster</Preset>
								</Video>
								<Video>
									<Name>video_720</Name>
									<Codec>h264</Codec>
									<Bitrate>2024000</Bitrate>
									<Width>1280</Width>
									<Height>720</Height>
									<Framerate>30</Framerate>
									<KeyFrameInterval>30</KeyFrameInterval>
									<BFrames>0</BFrames>									
									<Preset>faster</Preset>
								</Video>	
								<Audio>
									<Name>bypass_audio</Name>
									<Bypass>true</Bypass>
								</Audio>
								<!-- 
									 If all conditions of BypassIfMatch are matched, bypass is used without encoding.
									 [Options] 
									   eq: euqal to
									   lte: less than or equal to
									   gte: greater than or equal to

								<Video>
									<Name>conditional_video_encoding</Name>
									<Codec>h264</Codec>
									<Bitrate>2024000</Bitrate>
									<Width>1280</Width>
									<Height>720</Height>
									<Framerate>30</Framerate>
									<BypassIfMatch>
										<Codec>eq</Codec>
										<Framerate>lte</Framerate>
										<Width>lte</Width>
										<Height>lte</Height>
										<SAR>eq</SAR>
									</BypassIfMatch>									
								</Video>
								<Audio>
									<Name>conditional_audio_encoding</Name>
									<Codec>aac</Codec>
									<Bitrate>128000</Bitrate>
									<Samplerate>48000</Samplerate>
									<Channel>2</Channel>
									<BypassIfMatch>
										<Codec>eq</Codec>
										<Samplerate>lte</Samplerate>
										<Channel>eq</Channel>
									</BypassIfMatch>
								</Audio>																			
								-->
							</Encodes>
						</OutputProfile>
					</OutputProfiles>
					<Providers>
						<OVT />
						<RTMP />
						<SRT />
						<MPEGTS>
							<StreamMap>
								<!--
									Set the stream name of the client connected to the port to "stream_${Port}"
									For example, if a client connects to port 4000, OME creates a "stream_4000" stream
									<Stream>
										<Name>stream_${Port}</Name>
										<Port>4000,4001-4004</Port>
									</Stream>
									<Stream>
										<Name>stream_4005</Name>
										<Port>4005</Port>
									</Stream>
								-->
								<Stream>
									<Name>stream_${Port}</Name>
									<Port>4000</Port>
								</Stream>
							</StreamMap>
						</MPEGTS>
						<RTSPPull />
						<WebRTC>
							<Timeout>30000</Timeout>
							<CrossDomains>
								<Url>*</Url>
							</CrossDomains>
						</WebRTC>

						<Schedule>
							<MediaRootDir>/opt/ovenmediaengine/media</MediaRootDir>
							<ScheduleFilesDir>/opt/ovenmediaengine/media</ScheduleFilesDir>
						</Schedule>
					</Providers>
					<Publishers>
						<AppWorkerCount>1</AppWorkerCount>
						<StreamWorkerCount>8</StreamWorkerCount>
						<OVT />
						<WebRTC>
							<Timeout>30000</Timeout>
							<Rtx>false</Rtx>
							<Ulpfec>false</Ulpfec>
							<JitterBuffer>false</JitterBuffer>
							<CreateDefaultPlaylist>true</CreateDefaultPlaylist>
						</WebRTC>
						<LLHLS>
							<ChunkDuration>2</ChunkDuration>
							<!-- <PartHoldBack> SHOULD be at least three times the <ChunkDuration> -->
							<PartHoldBack>6</PartHoldBack>
							<SegmentDuration>6</SegmentDuration>
							<SegmentCount>10</SegmentCount>
							<DVR>
								<Enable>false</Enable>
								<TempStoragePath>/tmp/ome_dvr/</TempStoragePath>
								<MaxDuration>3600</MaxDuration>
							</DVR>
							<DRM>
								<Enable>false</Enable>
								<InfoFile>path/to/file.xml</InfoFile>
							</DRM>
							<CrossDomains>
								<Url>*</Url>
							</CrossDomains>
							<CreateDefaultPlaylist>true</CreateDefaultPlaylist> <!-- llhls.m3u8 -->
						</LLHLS>
						<HLS>
							<SegmentCount>4</SegmentCount>
							<SegmentDuration>4</SegmentDuration>
							<!-- <DVR>
								<Enable>true</Enable>
								<EventPlaylistType>false</EventPlaylistType>
								<TempStoragePath>/tmp/ome_dvr/</TempStoragePath>
								<MaxDuration>600</MaxDuration>
							</DVR> -->
							<CrossDomains>
								<Url>*</Url>
							</CrossDomains>
							<CreateDefaultPlaylist>true</CreateDefaultPlaylist> <!-- ts:playlist.m3u8 -->
						</HLS>

						<!--
						<Push>
							<StreamMap>
								<Enable>false</Enable>
								<Path>path/to/file.xml</Path>
							</StreamMap>
						</Push>
						<FILE>
							<RootPath>/prefix/path/to</RootPath>
							<StreamMap>
								<Enable>false</Enable>
								<Path>path/to/file.xml</Path>
							</StreamMap>
						</FILE>
						<Thumbnail>
							<CrossDomains>
								<Url>*</Url>
							</CrossDomains>
						</Thumbnail>						
						-->
					</Publishers>
				</Application>
			</Applications>
		</VirtualHost>
	</VirtualHosts>
</Server>

Answered by getroot

Aug 18, 2026

Initially, it worked by blocking playback requests until the segment was created, but since this occupied all of the HLS socket pool, we've now changed it to return a 404 response. There is a way to solve this, but it requires a very large structural change, so it's on hold for now. This will be addressed in the future.

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getroot · 2026-08-18T03:23:45Z

getroot
Aug 18, 2026
Maintainer

Initially, it worked by blocking playback requests until the segment was created, but since this occupied all of the HLS socket pool, we've now changed it to return a 404 response. There is a way to solve this, but it requires a very large structural change, so it's on hold for now. This will be addressed in the future.

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RTSP -> LLHLS 404 on first request, second request is successful #2220

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RTSP -> LLHLS 404 on first request, second request is successful #2220

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infernalxdrm Jun 26, 2026

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getroot Aug 18, 2026 Maintainer

infernalxdrm
Jun 26, 2026

getroot
Aug 18, 2026
Maintainer