Compatibility issue with rtmp2src / rtmp2sink

[usama-liaqat]

Which version are you using?

v0.21.2

Which operating system are you using?

• Linux amd64 standard
• Linux amd64 Docker
• Linux arm64 standard
• Linux arm64 Docker
• Linux arm7 standard
• Linux arm7 Docker
• Linux arm6 standard
• Linux arm6 Docker
• Windows amd64 standard
• Windows amd64 Docker (WSL backend)
• macOS amd64 standard
• macOS amd64 Docker
• Other (please describe)

Describe the issue

i am unable to play rtmp stream using gstreamer it gives error

gst-launch-1.0 playbin uri=rtmp://127.0.0.1:1935/test

Setting pipeline to PAUSED ...
ERROR: from element /GstPlayBin:playbin0/GstURIDecodeBin:uridecodebin0: URI lacks stream: rtmp://127.0.0.1:1935/test
Additional debug info:
../gst/playback/gsturidecodebin.c(1472): gen_source_element (): /GstPlayBin:playbin0/GstURIDecodeBin:uridecodebin0:
No element accepted URI 'rtmp://127.0.0.1:1935/test'
ERROR: pipeline doesn't want to preroll.
Failed to set pipeline to PAUSED.
Setting pipeline to NULL ...
Freeing pipeline ...

this stream is playable with ffmpeg but gives this error using gstreamer
same stream is playing good if i use rtsp url with gstreamer

Describe how to replicate the issue

1. start the server
2. publish with ffmpeg -stream_loop -1 -r 30 -f lavfi -re -i \"movie=cctv-demo.mp4\" -an -c:v libx264 -profile:v baseline -pix_fmt yuv420p -strict -2 -f flv rtmp://localhost:1935/test
3. read with gst-launch-1.0 playbin uri=rtmp://127.0.0.1:1935/test

Did you attach the server logs?

no

Did you attach a network dump?

no

[aler9]

Hello, this is a compatibility issue with the rtmp2src element of GStreamer, which is used by default by playbin.

The rtmp2src element provides an alternative rtmp implementation (the main one is rtmpsrc), it requires a path with two folders (i.e. /mystream/mysubpath) and there are other issues that makes it incompatible with the server.

While this issue is open, use rtmpsrc instead of playbin:

gst-launch-1.0 rtmpsrc location=rtmp://127.0.0.1:1935/test/stream ! decodebin ! autovideosink

[usama-liaqat]

Still Same Error i have tested with path with two folders.

gst-launch-1.0 playbin uri=rtmp://127.0.0.1:1935/live/fox-and-bird

Setting pipeline to PAUSED ... Pipeline is PREROLLING ... ERROR: from element /GstPlayBin:playbin0/GstURIDecodeBin:uridecodebin0/GstRtmp2Src:source: Failed to connect Additional debug info: ../gst/rtmp2/gstrtmp2src.c(974): send_connect_error (): /GstPlayBin:playbin0/GstURIDecodeBin:uridecodebin0/GstRtmp2Src:source: error g-io-error-quark:0: createStream failed: <cancelled> ERROR: pipeline doesn't want to preroll. ERROR: from element /GstPlayBin:playbin0/GstURIDecodeBin:uridecodebin0/GstDecodeBin:decodebin0/GstFlvDemux:flvdemux0: Internal data stream error. Additional debug info: ../gst/flv/gstflvdemux.c(3352): gst_flv_demux_sink_event (): /GstPlayBin:playbin0/GstURIDecodeBin:uridecodebin0/GstDecodeBin:decodebin0/GstFlvDemux:flvdemux0: Got EOS before any data ERROR: pipeline doesn't want to preroll. Setting pipeline to NULL ... Freeing pipeline ...

[gdesmott]

Problem is still present using GStreamer main. It seems that's mediamtx which is closing the RTMP connection:

$ GST_DEBUG="*rtmp*:6" gst-launch-1.0 rtmp2src location=rtmp://localhost/live/myStream ! fakesink
0:00:00.012389921 113743      0x1b1b460 DEBUG   rtmp2locationhandler gstrtmp2locationhandler.c:158:uri_handler_set_uri:<rtmp2src0> setting URI from rtmp://localhost/live/myStream
0:00:00.012406242 113743      0x1b1b460 DEBUG   rtmp2locationhandler gstrtmp2locationhandler.c:219:uri_handler_set_uri:<rtmp2src0> setting location to rtmp://localhost:1935/live stream myStream
Setting pipeline to PAUSED ...
0:00:00.012686722 113743      0x1b1b460 INFO                rtmp2src gstrtmp2src.c:463:gst_rtmp2_src_start:<rtmp2src0> Starting (async)
Pipeline is PREROLLING ...
0:00:00.012816257 113743      0x1d50400 DEBUG               rtmp2src gstrtmp2src.c:756:gst_rtmp2_src_task_func:<rtmp2src0> gst_rtmp2_src_task starting
0:00:00.013027747 113743      0x1d50400 DEBUG             rtmpclient rtmpclient.c:445:socket_connect: Starting socket connection
0:00:00.015060032 113743      0x1d50460 LOG                 rtmp2src gstrtmp2src.c:582:gst_rtmp2_src_create:<rtmp2src0> create
0:00:00.015248849 113743      0x1d50400 DEBUG             rtmpclient rtmpclient.c:478:socket_connect_done: Socket connection established
0:00:00.015255081 113743      0x1d50400 INFO           rtmphandshake rtmphandshake.c:152:gst_rtmp_client_handshake: Starting client handshake
0:00:00.015280109 113743      0x1d50400 DEBUG          rtmphandshake rtmphandshake.c:134:create_c0c1: Sending C0+C1
0:00:00.015304475 113743      0x1d50400 DEBUG          rtmphandshake rtmphandshake.c:189:client_handshake1_done: Sent C0+C1, waiting for S0+S1+S2
0:00:00.015385037 113743      0x1d50400 DEBUG          rtmphandshake rtmphandshake.c:247:client_handshake2_done: Got S0+S1+S2
0:00:00.015388173 113743      0x1d50400 DEBUG          rtmphandshake rtmphandshake.c:253:client_handshake2_done: S2 random data matches C1
0:00:00.015389626 113743      0x1d50400 DEBUG          rtmphandshake rtmphandshake.c:210:create_c2: Sending C2
0:00:00.015401118 113743      0x1d50400 DEBUG          rtmphandshake rtmphandshake.c:298:client_handshake3_done: Sent C2
0:00:00.015403071 113743      0x1d50400 INFO           rtmphandshake rtmphandshake.c:299:client_handshake3_done: Client handshake finished
0:00:00.015410876 113743      0x1d50400 LOG                  rtmpamf amf.c:1150:gst_amf_node_serialize: Serializing value: "@setDataFrame"
0:00:00.015439861 113743      0x1d50400 DEBUG         rtmpconnection rtmpconnection.c:1155:gst_rtmp_connection_send_command:<GstRtmpConnection@0x1d5e5a0> Sending command 'connect' on stream id 0
0:00:00.015445892 113743      0x1d50400 LOG           rtmpconnection rtmpconnection.c:1164:gst_rtmp_connection_send_command:<GstRtmpConnection@0x1d5e5a0> Registering send_connect_done for transid 1
0:00:00.015449559 113743      0x1d50400 LOG                  rtmpamf amf.c:1188:gst_amf_serialize_command_valist: Serializing command 'connect', transid 1
0:00:00.015455861 113743      0x1d50400 LOG                  rtmpamf amf.c:609:dump_argument: Argument #0: { "app": "live", "flashVer": "LNX 10,0,32,18", "tcUrl": "rtmp://localhost/live", "fpad": False, "capabilities": 15, "audioCodecs": 4071, "videoCodecs": 252, "videoFunction": 1 }
0:00:00.015467914 113743      0x1d50400 DEBUG        rtmpchunkstream rtmpchunkstream.c:729:gst_rtmp_chunk_streams_get: Allocating chunk stream 3
0:00:00.015472743 113743      0x1d50400 LOG              rtmpmessage rtmpmessage.c:269:gst_rtmp_buffer_dump: >>> message buffer: 0x1d590b0, pts 99:99:99.999999999, dts 99:99:99.999999999, dur 99:99:99.999999999, size 192, offset none, offset_end none, flags 0x4000
0:00:00.015476801 113743      0x1d50400 LOG              rtmpmessage rtmpmessage.c:273:gst_rtmp_buffer_dump: >>> message cstream:3    mstream:0    ts:0        len:192    type:command-amf0
0:00:00.015483123 113743      0x1d50400 LOG              rtmpmessage rtmpmessage.c:269:gst_rtmp_buffer_dump: >>> chunk buffer: 0x7fc71402c800, pts 99:99:99.999999999, dts 99:99:99.999999999, dur 99:99:99.999999999, size 140, offset 0, offset_end 128, flags 0x4000
0:00:00.015486750 113743      0x1d50400 LOG              rtmpmessage rtmpmessage.c:269:gst_rtmp_buffer_dump: >>> chunk buffer: 0x1ad81c0, pts 99:99:99.999999999, dts 99:99:99.999999999, dur 99:99:99.999999999, size 65, offset 128, offset_end 192, flags 0x4000
0:00:00.015498452 113743      0x1d50400 LOG           rtmpconnection rtmpconnection.c:624:gst_rtmp_connection_write_buffer_done:<GstRtmpConnection@0x1d5e5a0> write completed; wrote 205 bytes
0:00:00.056120694 113743      0x1d50400 DEBUG        rtmpchunkstream rtmpchunkstream.c:729:gst_rtmp_chunk_streams_get: Allocating chunk stream 2
0:00:00.056155380 113743      0x1d50400 DEBUG        rtmpchunkstream rtmpchunkstream.c:452:gst_rtmp_chunk_stream_parse_header: Starting parse with new buffer: 0x7fc71402c800, pts 99:99:99.999999999, dts 0:00:00.000000000, dur 99:99:99.999999999, size 0, offset 0, offset_end none, flags 0x40
0:00:00.056169897 113743      0x1d50400 LOG              rtmpmessage rtmpmessage.c:269:gst_rtmp_buffer_dump: <<< message buffer: 0x7fc71402c800, pts 99:99:99.999999999, dts 0:00:00.000000000, dur 99:99:99.999999999, size 4, offset 0, offset_end 4, flags 0x4040
0:00:00.056178023 113743      0x1d50400 LOG              rtmpmessage rtmpmessage.c:273:gst_rtmp_buffer_dump: <<< message cstream:2    mstream:0    ts:0        len:4      type:window-ack-size
0:00:00.056188733 113743      0x1d50400 LOG           rtmpconnection rtmpconnection.c:840:gst_rtmp_connection_handle_protocol_control:<GstRtmpConnection@0x1d5e5a0> got protocol control message 5:window-ack-size
0:00:00.056193642 113743      0x1d50400 INFO          rtmpconnection rtmpconnection.c:862:gst_rtmp_connection_handle_protocol_control:<GstRtmpConnection@0x1d5e5a0> incoming window ack size: 2500000
0:00:00.056201116 113743      0x1d50400 LOG              rtmpmessage rtmpmessage.c:269:gst_rtmp_buffer_dump: <<< message buffer: 0x1ad81c0, pts 99:99:99.999999999, dts 0:00:00.000000000, dur 99:99:99.999999999, size 5, offset 4, offset_end 9, flags 0x4000
0:00:00.056207488 113743      0x1d50400 LOG              rtmpmessage rtmpmessage.c:273:gst_rtmp_buffer_dump: <<< message cstream:2    mstream:0    ts:0        len:5      type:set-peer-bandwidth
0:00:00.056212628 113743      0x1d50400 LOG           rtmpconnection rtmpconnection.c:840:gst_rtmp_connection_handle_protocol_control:<GstRtmpConnection@0x1d5e5a0> got protocol control message 6:set-peer-bandwidth
0:00:00.056217026 113743      0x1d50400 FIXME         rtmpconnection rtmpconnection.c:868:gst_rtmp_connection_handle_protocol_control:<GstRtmpConnection@0x1d5e5a0> set peer bandwidth: 2500000, 2
0:00:00.056224581 113743      0x1d50400 INFO          rtmpconnection rtmpconnection.c:1322:gst_rtmp_connection_prepare_protocol_control:<GstRtmpConnection@0x1d5e5a0> pending window ack size: 2500000
0:00:00.056229089 113743      0x1d50400 DEBUG        rtmpchunkstream rtmpchunkstream.c:729:gst_rtmp_chunk_streams_get: Allocating chunk stream 2
0:00:00.056235962 113743      0x1d50400 LOG              rtmpmessage rtmpmessage.c:269:gst_rtmp_buffer_dump: >>> message buffer: 0x7fc71400fe50, pts 99:99:99.999999999, dts 99:99:99.999999999, dur 99:99:99.999999999, size 4, offset none, offset_end none, flags 0x4000
0:00:00.056248646 113743      0x1d50400 LOG              rtmpmessage rtmpmessage.c:273:gst_rtmp_buffer_dump: >>> message cstream:2    mstream:0    ts:0        len:4      type:window-ack-size
0:00:00.056257112 113743      0x1d50400 LOG              rtmpmessage rtmpmessage.c:269:gst_rtmp_buffer_dump: >>> chunk buffer: 0x7fc71403c8f0, pts 99:99:99.999999999, dts 99:99:99.999999999, dur 99:99:99.999999999, size 16, offset 0, offset_end 4, flags 0x4000
0:00:00.056289263 113743      0x1d50400 LOG              rtmpmessage rtmpmessage.c:269:gst_rtmp_buffer_dump: <<< message buffer: 0x7fc71402c800, pts 99:99:99.999999999, dts 0:00:00.000000000, dur 99:99:99.999999999, size 4, offset 9, offset_end 13, flags 0x4000
0:00:00.056295635 113743      0x1d50400 LOG              rtmpmessage rtmpmessage.c:273:gst_rtmp_buffer_dump: <<< message cstream:2    mstream:0    ts:0        len:4      type:set-chunk-size
0:00:00.056301186 113743      0x1d50400 LOG           rtmpconnection rtmpconnection.c:840:gst_rtmp_connection_handle_protocol_control:<GstRtmpConnection@0x1d5e5a0> got protocol control message 1:set-chunk-size
0:00:00.056305384 113743      0x1d50400 INFO          rtmpconnection rtmpconnection.c:845:gst_rtmp_connection_handle_protocol_control:<GstRtmpConnection@0x1d5e5a0> incoming chunk size 65536
0:00:00.056308950 113743      0x1d50400 DEBUG        rtmpchunkstream rtmpchunkstream.c:729:gst_rtmp_chunk_streams_get: Allocating chunk stream 3
0:00:00.056314751 113743      0x1d50400 DEBUG        rtmpchunkstream rtmpchunkstream.c:452:gst_rtmp_chunk_stream_parse_header: Starting parse with new buffer: 0x7fc71402c800, pts 99:99:99.999999999, dts 0:00:00.000000000, dur 99:99:99.999999999, size 0, offset 0, offset_end none, flags 0x40
0:00:00.056322556 113743      0x1d50400 LOG              rtmpmessage rtmpmessage.c:269:gst_rtmp_buffer_dump: <<< message buffer: 0x7fc71402c800, pts 99:99:99.999999999, dts 0:00:00.000000000, dur 99:99:99.999999999, size 190, offset 0, offset_end 190, flags 0x4040
0:00:00.056352523 113743      0x1d50400 LOG              rtmpmessage rtmpmessage.c:273:gst_rtmp_buffer_dump: <<< message cstream:3    mstream:0    ts:0        len:190    type:command-amf0
0:00:00.056363954 113743      0x1d50400 LOG                  rtmpamf amf.c:960:gst_amf_parse_command: Parsing command '_result', transid 1
0:00:00.056377670 113743      0x1d50400 LOG                  rtmpamf amf.c:609:dump_argument: Argument #0: { "fmsVer": "LNX 9,0,124,2", "capabilities": 31 }
0:00:00.056386657 113743      0x1d50400 LOG                  rtmpamf amf.c:609:dump_argument: Argument #1: { "level": "status", "code": "NetConnection.Connect.Success", "description": "Connection succeeded.", "objectEncoding": 0 }
0:00:00.056393400 113743      0x1d50400 DEBUG         rtmpconnection rtmpconnection.c:1052:gst_rtmp_connection_handle_cm:<GstRtmpConnection@0x1d5e5a0> got control message "_result" transaction 1 size 190
0:00:00.056400694 113743      0x1d50400 LOG           rtmpconnection rtmpconnection.c:1068:gst_rtmp_connection_handle_cm:<GstRtmpConnection@0x1d5e5a0> calling transaction callback send_connect_done
0:00:00.056406385 113743      0x1d50400 INFO              rtmpclient rtmpclient.c:734:send_connect_done: connect result: NetConnection.Connect.Success
0:00:00.056420181 113743      0x1d50400 INFO              rtmpclient rtmpclient.c:1178:send_create_stream: Creating stream 'myStream'
0:00:00.056425260 113743      0x1d50400 DEBUG         rtmpconnection rtmpconnection.c:1155:gst_rtmp_connection_send_command:<GstRtmpConnection@0x1d5e5a0> Sending command 'createStream' on stream id 0
0:00:00.056430019 113743      0x1d50400 LOG           rtmpconnection rtmpconnection.c:1164:gst_rtmp_connection_send_command:<GstRtmpConnection@0x1d5e5a0> Registering create_stream_done for transid 2
0:00:00.056435069 113743      0x1d50400 LOG                  rtmpamf amf.c:1188:gst_amf_serialize_command_valist: Serializing command 'createStream', transid 2
0:00:00.056440419 113743      0x1d50400 LOG                  rtmpamf amf.c:609:dump_argument: Argument #0: null
0:00:00.056460838 113743      0x1d50400 LOG           rtmpconnection rtmpconnection.c:624:gst_rtmp_connection_write_buffer_done:<GstRtmpConnection@0x1d5e5a0> write completed; wrote 16 bytes
0:00:00.056465597 113743      0x1d50400 INFO          rtmpconnection rtmpconnection.c:1366:gst_rtmp_connection_apply_protocol_control:<GstRtmpConnection@0x1d5e5a0> applied window ack size 2500000
0:00:00.056469805 113743      0x1d50400 INFO          rtmpconnection rtmpconnection.c:1322:gst_rtmp_connection_prepare_protocol_control:<GstRtmpConnection@0x1d5e5a0> pending window ack size: 2500000
0:00:00.056473762 113743      0x1d50400 LOG          rtmpchunkstream rtmpchunkstream.c:150:dts_diff_to_delta_ts: Timestamps not valid; using delta TS 0
0:00:00.056479222 113743      0x1d50400 LOG              rtmpmessage rtmpmessage.c:269:gst_rtmp_buffer_dump: >>> message buffer: 0x7fc71402c920, pts 99:99:99.999999999, dts 99:99:99.999999999, dur 99:99:99.999999999, size 4, offset none, offset_end none, flags 0x4000
0:00:00.056484913 113743      0x1d50400 LOG              rtmpmessage rtmpmessage.c:273:gst_rtmp_buffer_dump: >>> message cstream:2    mstream:0    ts:0        len:4      type:window-ack-size
0:00:00.056491045 113743      0x1d50400 LOG              rtmpmessage rtmpmessage.c:269:gst_rtmp_buffer_dump: >>> chunk buffer: 0x7fc71400fe50, pts 99:99:99.999999999, dts 99:99:99.999999999, dur 99:99:99.999999999, size 5, offset 4, offset_end 8, flags 0x4000
0:00:00.056506785 113743      0x1d50400 LOG           rtmpconnection rtmpconnection.c:624:gst_rtmp_connection_write_buffer_done:<GstRtmpConnection@0x1d5e5a0> write completed; wrote 5 bytes
0:00:00.056510782 113743      0x1d50400 INFO          rtmpconnection rtmpconnection.c:1366:gst_rtmp_connection_apply_protocol_control:<GstRtmpConnection@0x1d5e5a0> applied window ack size 2500000
0:00:00.056514189 113743      0x1d50400 LOG          rtmpchunkstream rtmpchunkstream.c:150:dts_diff_to_delta_ts: Timestamps not valid; using delta TS 0
0:00:00.056519338 113743      0x1d50400 LOG              rtmpmessage rtmpmessage.c:269:gst_rtmp_buffer_dump: >>> message buffer: 0x7fc714034cb0, pts 99:99:99.999999999, dts 99:99:99.999999999, dur 99:99:99.999999999, size 10, offset none, offset_end none, flags 0x4000
0:00:00.056524729 113743      0x1d50400 LOG              rtmpmessage rtmpmessage.c:273:gst_rtmp_buffer_dump: >>> message cstream:2    mstream:0    ts:0        len:10     type:user-control
0:00:00.056530570 113743      0x1d50400 LOG              rtmpmessage rtmpmessage.c:269:gst_rtmp_buffer_dump: >>> chunk buffer: 0x7fc71402c920, pts 99:99:99.999999999, dts 99:99:99.999999999, dur 99:99:99.999999999, size 18, offset 8, offset_end 18, flags 0x4000
0:00:00.056541931 113743      0x1d50400 LOG           rtmpconnection rtmpconnection.c:624:gst_rtmp_connection_write_buffer_done:<GstRtmpConnection@0x1d5e5a0> write completed; wrote 18 bytes
0:00:00.056546239 113743      0x1d50400 LOG          rtmpchunkstream rtmpchunkstream.c:150:dts_diff_to_delta_ts: Timestamps not valid; using delta TS 0
0:00:00.056550968 113743      0x1d50400 LOG              rtmpmessage rtmpmessage.c:269:gst_rtmp_buffer_dump: >>> message buffer: 0x7fc71402b640, pts 99:99:99.999999999, dts 99:99:99.999999999, dur 99:99:99.999999999, size 25, offset none, offset_end none, flags 0x4000
0:00:00.056555817 113743      0x1d50400 LOG              rtmpmessage rtmpmessage.c:273:gst_rtmp_buffer_dump: >>> message cstream:3    mstream:0    ts:0        len:25     type:command-amf0
0:00:00.056562991 113743      0x1d50400 LOG              rtmpmessage rtmpmessage.c:269:gst_rtmp_buffer_dump: >>> chunk buffer: 0x1d590b0, pts 99:99:99.999999999, dts 99:99:99.999999999, dur 99:99:99.999999999, size 33, offset 192, offset_end 217, flags 0x4000
0:00:00.056573531 113743      0x1d50400 LOG           rtmpconnection rtmpconnection.c:624:gst_rtmp_connection_write_buffer_done:<GstRtmpConnection@0x1d5e5a0> write completed; wrote 33 bytes
0:00:00.097069064 113743      0x1d50400 ERROR         rtmpconnection rtmpconnection.c:496:gst_rtmp_connection_input_ready:<GstRtmpConnection@0x1d5e5a0> read error: g-io-error-quark 44 connection closed remotely
0:00:00.097080756 113743      0x1d50400 LOG           rtmpconnection rtmpconnection.c:382:cancel_all_commands:<GstRtmpConnection@0x1d5e5a0> calling transaction callback create_stream_done
0:00:00.097092167 113743      0x1d50400 ERROR               rtmp2src gstrtmp2src.c:963:send_connect_error:<rtmp2src0> Failed to connect: g-io-error-quark 0 'createStream' cmd failed: connection closed remotely
0:00:00.097098890 113743      0x1d50400 WARN                rtmp2src gstrtmp2src.c:973:send_connect_error:<rtmp2src0> error: Failed to connect: 'createStream' cmd failed: connection closed remotely
0:00:00.097102296 113743      0x1d50400 WARN                rtmp2src gstrtmp2src.c:973:send_connect_error:<rtmp2src0> error: domain g-io-error-quark, code 0
ERROR: from element /GstPipeline:pipeline0/GstRtmp2Src:rtmp2src0: Failed to connect: 'createStream' cmd failed: connection closed remotely
Additional debug info:
../subprojects/gst-plugins-bad/gst/rtmp2/gstrtmp2src.c(973): send_connect_error (): /GstPipeline:pipeline0/GstRtmp2Src:rtmp2src0:
domain g-io-error-quark, code 0
ERROR: pipeline doesn't want to preroll.

[gdesmott]

Here are the server logs when rtmp2src tries to read a stream:

2023/07/28 10:51:15 INF [RTMP] [conn [::1]:57132] opened
2023/07/28 10:51:15 INF [RTMP] [conn [::1]:57132] closed (received type 3 chunk without previous chunk)

[usama-liaqat]

@aler9 rtmp2src is still not working in new versions. as it is working well with nginx rtmp module. as of my knowledge nginx rtmp is not rtmps and still working with gstreamer.

[m1k1o]

The sequence of messages received from rtmp2src is following:

readMessage, typ: 0
readMessage, typ: 3
readMessage, typ: 0
readMessage, typ: 3

When commenting out these lines:

mediamtx/internal/protocols/rtmp/rawmessage/reader.go

Lines 210 to 212 in 308b7a9

	if !rc.curTimestampDeltaAvailable {
	return nil, fmt.Errorf("received type 3 chunk without previous chunk")
	}

Everything seems to be working. Not sure what other implications could there be.

[ErikBahena]

@m1k1o That did it for me. Thank you