Problem with RTSP stream playback command (Vapix)

[JosephPtn]

Hello! I am developing an application for viewing camera recordings, as a library for playing rtsp streams I use vlc (I am writing an application for Android and IOS). I encountered a problem with playing a recording not from the beginning when it is not yet finished. Cameras ignore the parameter that I pass through VLC — start-time (offset in seconds relative to the beginning of the recording), but if the recording is finished, everything works correctly. Tell me, is this a limitation of the camera? (the inability to play a recording not from the beginning of the recording if it is still in progress)

[github-actions[bot]]

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[vivekatoffice]

Hi @JosephPtn ,
I hope you have checked this discussion 920.

[JosephPtn]

the link is looking at a record that has a stoptime, it is finished. I have no problems with such records, they are played correctly with the offset, the problem is with records that are currently running, they ignore the offset of the start of playback and start playing all over again

[vivekatoffice]

Hi @JosephPtn,
I am working on the web page for the AXIS Q1715 Block Camera, which uses AXIS OS version 12.4.57. It is able to do the offset.

I encountered the same issue you mentioned in VLC with this RTSP URL:

rtsp://root:Axis%40125Apass@10.176.12.49/axis-media/media.amp?recordingid=20250425_093358_D74B_B8A44F279CC0

Have you tried the Recording API's play.cgi with stoptime parameters?

http://http//10.176.12.49/axis-cgi/record/play.cgi?recordingid=20250425_093358_D74B_B8A44F279CC0&starttime=2025-04-25T15%3A20%3A01Z

[vivekatoffice]

Hi @JosephPtn ,
As per my understanding the issue is with the VLC implementation of --start-time as we know VLC uses Live555: LiveMedia from Live555 is a library for streaming data over a RTSP, RTP and SDP connection.

vlc --file-logging --logfile=vlc-log.txt --verbose=3 --start-time=30 --run-time=60 "rtsp://root:pass@10.176.12.49/axis-media/media.amp?recordingid=20250428_053519_2BAF_B8A44F279CC0" --sout="#file{dst=recorded.mp4}" vlc://quit

main warning: INPUT_CONTROL_SET_TIME 30000000 failed or not possible

\vlc\src\input\input.c

You can try building VLC after changing the live555.cpp

I tried the following steps to test if RTSP URL works with Live555:

1. Use openRTSP without ** -s 60**

E:\Vivek\Demo\OpenRTSP\live555-latest\live\testProgs>openRTSP -u root pass -d 30  -F "E:\Vivek\Demo\OpenRTSP\live555-latest\live\testProgs\vivek" "rtsp://10.176.12.49:554/axis-media/media.amp?recordingid=20250428_053519_2BAF_B8A44F279CC0"
E:\Vivek\Demo\OpenRTSP\live555-latest\live\testProgs>"C:\Program Files\VideoLAN\VLC\vlc.exe" vivekvideo-H264-1 --demux=h264

Logs

E:\Vivek\Demo\OpenRTSP\live555-latest\live\testProgs>openRTSP -u root pass -d 30 -F "E:\Vivek\Demo\OpenRTSP\live555-latest\live\testProgs\vivek" "rtsp://10.176.12.49:554/axis-media/media.amp?recordingid=20250428_053519_2BAF_B8A44F279CC0"      
Created new TCP socket 3 for connection
Connecting to 10.176.12.49, port 554 on socket 3...
...remote connection opened
Sending request: OPTIONS rtsp://10.176.12.49:554/axis-media/media.amp?recordingid=20250428_053519_2BAF_B8A44F279CC0 RTSP/1.0
CSeq: 2
User-Agent: openRTSP (LIVE555 Streaming Media v2025.01.17)


Received 201 new bytes of response data.
Received a complete OPTIONS response:
RTSP/1.0 200 OK
CSeq: 2
Public: OPTIONS, DESCRIBE, ANNOUNCE, GET_PARAMETER, PAUSE, PLAY, RECORD, SETUP, SET_PARAMETER, TEARDOWN
Server: GStreamer RTSP server
Date: Mon, 28 Apr 2025 06:23:12 GMT


Sending request: DESCRIBE rtsp://10.176.12.49:554/axis-media/media.amp?recordingid=20250428_053519_2BAF_B8A44F279CC0 RTSP/1.0
CSeq: 3
User-Agent: openRTSP (LIVE555 Streaming Media v2025.01.17)
Accept: application/sdp


Received 228 new bytes of response data.
Received a complete DESCRIBE response:
RTSP/1.0 401 Unauthorized
CSeq: 3
WWW-Authenticate: Digest realm="AXIS_B8A44F279CC0", nonce="0000276eY15758197cfe471d90253baed666813912e5a90", stale=FALSE
Server: GStreamer RTSP server
Date: Mon, 28 Apr 2025 06:23:12 GMT


Resending...
Sending request: DESCRIBE rtsp://10.176.12.49:554/axis-media/media.amp?recordingid=20250428_053519_2BAF_B8A44F279CC0 RTSP/1.0
CSeq: 4
Authorization: Digest username="root", realm="AXIS_B8A44F279CC0", nonce="0000276eY15758197cfe471d90253baed666813912e5a90", uri="rtsp://10.176.12.49:554/axis-media/media.amp?recordingid=20250428_053519_2BAF_B8A44F279CC0", response="a19e0d0f0e9fde07208ceb35ddb6d04f"
User-Agent: openRTSP (LIVE555 Streaming Media v2025.01.17)
Accept: application/sdp


Received 953 new bytes of response data.
Received a complete DESCRIBE response:
RTSP/1.0 200 OK
CSeq: 4
Content-Type: application/sdp
Content-Base: rtsp://10.176.12.49:554/axis-media/record/play.amp/
Server: GStreamer RTSP server
Date: Mon, 28 Apr 2025 06:23:12 GMT
Content-Length: 738

v=0
o=- 14165274191604168190 1 IN IP4 10.176.12.49
s=Session streamed with GStreamer
i=rtsp-server
t=0 0
a=tool:GStreamer
a=type:broadcast
a=range:npt=0-
a=control:rtsp://10.176.12.49:554/axis-media/media.amp?recordingid=20250428_053519_2BAF_B8A44F279CC0
m=video 0 RTP/AVP 96
c=IN IP4 0.0.0.0
a=rtpmap:96 H264/90000
a=framerate:30
a=fmtp:96 packetization-mode=1;sprop-parameter-sets=Z2QAKa0AxSAeAIn5ZuAgIDSgANu6ADN/mAHiRFQ=,aO48sA==;profile-level-id=640029;level-asymmetry-allowed=1
a=ts-refclk:local
a=mediaclk:sender
a=ssrc:3310615151 cname:user4121046466@host-d5ac5a21
a=x-onvif-track:VIDEO000
a=recvonly
a=control:rtsp://10.176.12.49:554/axis-media/media.amp/stream=0?recordingid=20250428_053519_2BAF_B8A44F279CC0

Opened URL "rtsp://10.176.12.49:554/axis-media/media.amp?recordingid=20250428_053519_2BAF_B8A44F279CC0", returning a SDP description:
v=0
o=- 14165274191604168190 1 IN IP4 10.176.12.49
s=Session streamed with GStreamer
i=rtsp-server
t=0 0
a=tool:GStreamer
a=type:broadcast
a=range:npt=0-
a=control:rtsp://10.176.12.49:554/axis-media/media.amp?recordingid=20250428_053519_2BAF_B8A44F279CC0
m=video 0 RTP/AVP 96
c=IN IP4 0.0.0.0
a=rtpmap:96 H264/90000
a=framerate:30
a=fmtp:96 packetization-mode=1;sprop-parameter-sets=Z2QAKa0AxSAeAIn5ZuAgIDSgANu6ADN/mAHiRFQ=,aO48sA==;profile-level-id=640029;level-asymmetry-allowed=1
a=ts-refclk:local
a=mediaclk:sender
a=ssrc:3310615151 cname:user4121046466@host-d5ac5a21
a=x-onvif-track:VIDEO000
a=recvonly
a=control:rtsp://10.176.12.49:554/axis-media/media.amp/stream=0?recordingid=20250428_053519_2BAF_B8A44F279CC0

Created receiver for "video/H264" subsession (client ports 50582-50583)
Sending request: SETUP rtsp://10.176.12.49:554/axis-media/media.amp/stream=0?recordingid=20250428_053519_2BAF_B8A44F279CC0 RTSP/1.0
CSeq: 5
Authorization: Digest username="root", realm="AXIS_B8A44F279CC0", nonce="0000276eY15758197cfe471d90253baed666813912e5a90", uri="rtsp://10.176.12.49:554/axis-media/record/play.amp/", response="cb08cf90672684594e567bc8dbd1ce6c"
User-Agent: openRTSP (LIVE555 Streaming Media v2025.01.17)
Transport: RTP/AVP;unicast;client_port=50582-50583


Received 236 new bytes of response data.
Received a complete SETUP response:
RTSP/1.0 200 OK
CSeq: 5
Transport: RTP/AVP;unicast;client_port=50582-50583;server_port=50000-50001;ssrc=C553FA6F;mode="PLAY"
Server: GStreamer RTSP server
Session: T6c5x6qcldfBk69V;timeout=10
Date: Mon, 28 Apr 2025 06:23:12 GMT


Setup "video/H264" subsession (client ports 50582-50583)
Created output file: "E:\Vivek\Demo\OpenRTSP\live555-latest\live\testProgs\vivekvideo-H264-1"
Sending request: PLAY rtsp://10.176.12.49:554/axis-media/media.amp?recordingid=20250428_053519_2BAF_B8A44F279CC0 RTSP/1.0
CSeq: 6
Authorization: Digest username="root", realm="AXIS_B8A44F279CC0", nonce="0000276eY15758197cfe471d90253baed666813912e5a90", uri="rtsp://10.176.12.49:554/axis-media/record/play.amp/", response="3e44b1f468cc45fab2bc1bac3496251f"
User-Agent: openRTSP (LIVE555 Streaming Media v2025.01.17)
Session: T6c5x6qcldfBk69V
Range: npt=0.000-30.000


Received 293 new bytes of response data.
Received a complete PLAY response:
RTSP/1.0 200 OK
CSeq: 6
RTP-Info: url=rtsp://10.176.12.49:554/axis-media/media.amp/stream=0?recordingid=20250428_053519_2BAF_B8A44F279CC0;seq=30279;rtptime=71983958
Range: npt=0-30
Server: GStreamer RTSP server
Session: T6c5x6qcldfBk69V;timeout=10
Date: Mon, 28 Apr 2025 06:23:12 GMT


Started playing session
Receiving streamed data (for up to 30.000000 seconds)...
MultiFramedRTPSource::doGetNextFrame1(): The total received frame size exceeds the client's buffer size (100000).  23584 bytes of trailing data will be dropped!
FileSink::afterGettingFrame(): The input frame data was too large for our buffer size (100000).  23584 bytes of trailing data was dropped!  Correct this by increasing the "bufferSize" parameter in the "createNew()" call to at least 123584
MultiFramedRTPSource::doGetNextFrame1(): The total received frame size exceeds the client's buffer size (100000).  20829 bytes of trailing data will be dropped!
FileSink::afterGettingFrame(): The input frame data was too large for our buffer size (100000).  20829 bytes of trailing data was dropped!  Correct this by increasing the "bufferSize" parameter in the "createNew()" call to at least 120829
MultiFramedRTPSource::doGetNextFrame1(): The total received frame size exceeds the client's buffer size (100000).  20462 bytes of trailing data will be dropped!
FileSink::afterGettingFrame(): The input frame data was too large for our buffer size (100000).  20462 bytes of trailing data was dropped!  Correct this by increasing the "bufferSize" parameter in the "createNew()" call to at least 120462
MultiFramedRTPSource::doGetNextFrame1(): The total received frame size exceeds the client's buffer size (100000).  21037 bytes of trailing data will be dropped!
FileSink::afterGettingFrame(): The input frame data was too large for our buffer size (100000).  21037 bytes of trailing data was dropped!  Correct this by increasing the "bufferSize" parameter in the "createNew()" call to at least 121037
MultiFramedRTPSource::doGetNextFrame1(): The total received frame size exceeds the client's buffer size (100000).  21197 bytes of trailing data will be dropped!
FileSink::afterGettingFrame(): The input frame data was too large for our buffer size (100000).  21197 bytes of trailing data was dropped!  Correct this by increasing the "bufferSize" parameter in the "createNew()" call to at least 121197
MultiFramedRTPSource::doGetNextFrame1(): The total received frame size exceeds the client's buffer size (100000).  20378 bytes of trailing data will be dropped!
FileSink::afterGettingFrame(): The input frame data was too large for our buffer size (100000).  20378 bytes of trailing data was dropped!  Correct this by increasing the "bufferSize" parameter in the "createNew()" call to at least 120378
Received RTCP "BYE" on "video/H264" subsession (after 30 seconds)
Sending request: TEARDOWN rtsp://10.176.12.49:554/axis-media/media.amp?recordingid=20250428_053519_2BAF_B8A44F279CC0 RTSP/1.0
CSeq: 7
Authorization: Digest username="root", realm="AXIS_B8A44F279CC0", nonce="0000276eY15758197cfe471d90253baed666813912e5a90", uri="rtsp://10.176.12.49:554/axis-media/record/play.amp/", response="a8796d0529a46e3e13f76662b0d75d08"
User-Agent: openRTSP (LIVE555 Streaming Media v2025.01.17)
Session: T6c5x6qcldfBk69V


Received 153 new bytes of response data.
Received a complete TEARDOWN response:
RTSP/1.0 200 OK
CSeq: 7
Server: GStreamer RTSP server
Session: T6c5x6qcldfBk69V;timeout=10
Connection: close
Date: Mon, 28 Apr 2025 06:23:42 GMT



E:\Vivek\Demo\OpenRTSP\live555-latest\live\testProgs>"C:\Program Files\VideoLAN\VLC\vlc.exe" vivekvideo-H264-1 --demux=h264

Output:

2. Use openRTSP with ** -s 60**

E:\Vivek\Demo\OpenRTSP\live555-latest\live\testProgs>openRTSP -u root pass -d 30  -s 60 -F "E:\Vivek\Demo\OpenRTSP\live555-latest\live\testProgs\vivek" "rtsp://10.176.12.49:554/axis-media/media.amp?recordingid=20250428_053519_2BAF_B8A44F279CC0"
E:\Vivek\Demo\OpenRTSP\live555-latest\live\testProgs>"C:\Program Files\VideoLAN\VLC\vlc.exe" vivekvideo-H264-1 --demux=h264

Logs

E:\Vivek\Demo\OpenRTSP\live555-latest\live\testProgs>openRTSP -u root pass -d 30 -s 60 -F "E:\Vivek\Demo\OpenRTSP\live555-latest\live\testProgs\vivek" "rtsp://10.176.12.49:554/axis-media/media.amp?recordingid=20250428_053519_2BAF_B8A44F279CC0"
Created new TCP socket 3 for connection
Connecting to 10.176.12.49, port 554 on socket 3...
...remote connection opened
Sending request: OPTIONS rtsp://10.176.12.49:554/axis-media/media.amp?recordingid=20250428_053519_2BAF_B8A44F279CC0 RTSP/1.0
CSeq: 2
User-Agent: openRTSP (LIVE555 Streaming Media v2025.01.17)


Received 201 new bytes of response data.
Received a complete OPTIONS response:
RTSP/1.0 200 OK
CSeq: 2
Public: OPTIONS, DESCRIBE, ANNOUNCE, GET_PARAMETER, PAUSE, PLAY, RECORD, SETUP, SET_PARAMETER, TEARDOWN
Server: GStreamer RTSP server
Date: Mon, 28 Apr 2025 06:25:23 GMT


Sending request: DESCRIBE rtsp://10.176.12.49:554/axis-media/media.amp?recordingid=20250428_053519_2BAF_B8A44F279CC0 RTSP/1.0
CSeq: 3
User-Agent: openRTSP (LIVE555 Streaming Media v2025.01.17)
Accept: application/sdp


Received 228 new bytes of response data.
Received a complete DESCRIBE response:
RTSP/1.0 401 Unauthorized
CSeq: 3
WWW-Authenticate: Digest realm="AXIS_B8A44F279CC0", nonce="000027f1Y196406a8b21e4906a931de29db442c0366dfa0", stale=FALSE
Server: GStreamer RTSP server
Date: Mon, 28 Apr 2025 06:25:23 GMT


Resending...
Sending request: DESCRIBE rtsp://10.176.12.49:554/axis-media/media.amp?recordingid=20250428_053519_2BAF_B8A44F279CC0 RTSP/1.0
CSeq: 4
Authorization: Digest username="root", realm="AXIS_B8A44F279CC0", nonce="000027f1Y196406a8b21e4906a931de29db442c0366dfa0", uri="rtsp://10.176.12.49:554/axis-media/media.amp?recordingid=20250428_053519_2BAF_B8A44F279CC0", response="0d6eed8bfb814134f741ddc8f08431f1"
User-Agent: openRTSP (LIVE555 Streaming Media v2025.01.17)
Accept: application/sdp


Received 952 new bytes of response data.
Received a complete DESCRIBE response:
RTSP/1.0 200 OK
CSeq: 4
Content-Type: application/sdp
Content-Base: rtsp://10.176.12.49:554/axis-media/record/play.amp/
Server: GStreamer RTSP server
Date: Mon, 28 Apr 2025 06:25:23 GMT
Content-Length: 737

v=0
o=- 4676310354702858958 1 IN IP4 10.176.12.49
s=Session streamed with GStreamer
i=rtsp-server
t=0 0
a=tool:GStreamer
a=type:broadcast
a=range:npt=0-
a=control:rtsp://10.176.12.49:554/axis-media/media.amp?recordingid=20250428_053519_2BAF_B8A44F279CC0
m=video 0 RTP/AVP 96
c=IN IP4 0.0.0.0
a=rtpmap:96 H264/90000
a=framerate:30
a=fmtp:96 packetization-mode=1;sprop-parameter-sets=Z2QAKa0AxSAeAIn5ZuAgIDSgANu6ADN/mAHiRFQ=,aO48sA==;profile-level-id=640029;level-asymmetry-allowed=1
a=ts-refclk:local
a=mediaclk:sender
a=ssrc:3889977587 cname:user3799396400@host-e55a509d
a=x-onvif-track:VIDEO000
a=recvonly
a=control:rtsp://10.176.12.49:554/axis-media/media.amp/stream=0?recordingid=20250428_053519_2BAF_B8A44F279CC0

Opened URL "rtsp://10.176.12.49:554/axis-media/media.amp?recordingid=20250428_053519_2BAF_B8A44F279CC0", returning a SDP description:
v=0
o=- 4676310354702858958 1 IN IP4 10.176.12.49
s=Session streamed with GStreamer
i=rtsp-server
t=0 0
a=tool:GStreamer
a=type:broadcast
a=range:npt=0-
a=control:rtsp://10.176.12.49:554/axis-media/media.amp?recordingid=20250428_053519_2BAF_B8A44F279CC0
m=video 0 RTP/AVP 96
c=IN IP4 0.0.0.0
a=rtpmap:96 H264/90000
a=framerate:30
a=fmtp:96 packetization-mode=1;sprop-parameter-sets=Z2QAKa0AxSAeAIn5ZuAgIDSgANu6ADN/mAHiRFQ=,aO48sA==;profile-level-id=640029;level-asymmetry-allowed=1
a=ts-refclk:local
a=mediaclk:sender
a=ssrc:3889977587 cname:user3799396400@host-e55a509d
a=x-onvif-track:VIDEO000
a=recvonly
a=control:rtsp://10.176.12.49:554/axis-media/media.amp/stream=0?recordingid=20250428_053519_2BAF_B8A44F279CC0

Created receiver for "video/H264" subsession (client ports 61304-61305)
Sending request: SETUP rtsp://10.176.12.49:554/axis-media/media.amp/stream=0?recordingid=20250428_053519_2BAF_B8A44F279CC0 RTSP/1.0
CSeq: 5
Authorization: Digest username="root", realm="AXIS_B8A44F279CC0", nonce="000027f1Y196406a8b21e4906a931de29db442c0366dfa0", uri="rtsp://10.176.12.49:554/axis-media/record/play.amp/", response="e5527b242dd690428f3f53f914a85a2f"
User-Agent: openRTSP (LIVE555 Streaming Media v2025.01.17)
Transport: RTP/AVP;unicast;client_port=61304-61305


Received 236 new bytes of response data.
Received a complete SETUP response:
RTSP/1.0 200 OK
CSeq: 5
Transport: RTP/AVP;unicast;client_port=61304-61305;server_port=50000-50001;ssrc=E7DC58F3;mode="PLAY"
Server: GStreamer RTSP server
Session: Rs6kF2jSY7+nFd7i;timeout=10
Date: Mon, 28 Apr 2025 06:25:23 GMT


Setup "video/H264" subsession (client ports 61304-61305)
Created output file: "E:\Vivek\Demo\OpenRTSP\live555-latest\live\testProgs\vivekvideo-H264-1"
Sending request: PLAY rtsp://10.176.12.49:554/axis-media/media.amp?recordingid=20250428_053519_2BAF_B8A44F279CC0 RTSP/1.0
CSeq: 6
Authorization: Digest username="root", realm="AXIS_B8A44F279CC0", nonce="000027f1Y196406a8b21e4906a931de29db442c0366dfa0", uri="rtsp://10.176.12.49:554/axis-media/record/play.amp/", response="47033d619864aed9e2e73bcaf07d3995"
User-Agent: openRTSP (LIVE555 Streaming Media v2025.01.17)
Session: Rs6kF2jSY7+nFd7i
Range: npt=60.000-90.000


Received 296 new bytes of response data.
Received a complete PLAY response:
RTSP/1.0 200 OK
CSeq: 6
RTP-Info: url=rtsp://10.176.12.49:554/axis-media/media.amp/stream=0?recordingid=20250428_053519_2BAF_B8A44F279CC0;seq=7067;rtptime=934229973
Range: npt=58.4-90
Server: GStreamer RTSP server
Session: Rs6kF2jSY7+nFd7i;timeout=10
Date: Mon, 28 Apr 2025 06:25:23 GMT


Started playing session
Receiving streamed data (for up to 31.600000 seconds)...
MultiFramedRTPSource::doGetNextFrame1(): The total received frame size exceeds the client's buffer size (100000).  21100 bytes of trailing data will be dropped!
FileSink::afterGettingFrame(): The input frame data was too large for our buffer size (100000).  21100 bytes of trailing data was dropped!  Correct this by increasing the "bufferSize" parameter in the "createNew()" call to at least 121100
MultiFramedRTPSource::doGetNextFrame1(): The total received frame size exceeds the client's buffer size (100000).  21086 bytes of trailing data will be dropped!
FileSink::afterGettingFrame(): The input frame data was too large for our buffer size (100000).  21086 bytes of trailing data was dropped!  Correct this by increasing the "bufferSize" parameter in the "createNew()" call to at least 121086
MultiFramedRTPSource::doGetNextFrame1(): The total received frame size exceeds the client's buffer size (100000).  21181 bytes of trailing data will be dropped!
FileSink::afterGettingFrame(): The input frame data was too large for our buffer size (100000).  21181 bytes of trailing data was dropped!  Correct this by increasing the "bufferSize" parameter in the "createNew()" call to at least 121181
MultiFramedRTPSource::doGetNextFrame1(): The total received frame size exceeds the client's buffer size (100000).  21384 bytes of trailing data will be dropped!
FileSink::afterGettingFrame(): The input frame data was too large for our buffer size (100000).  21384 bytes of trailing data was dropped!  Correct this by increasing the "bufferSize" parameter in the "createNew()" call to at least 121384
Created new TCP socket 3 for connection
Connecting to 10.176.12.49, port 554 on socket 3...
...remote connection opened
Sending request: TEARDOWN rtsp://10.176.12.49:554/axis-media/media.amp?recordingid=20250428_053519_2BAF_B8A44F279CC0 RTSP/1.0
CSeq: 7
Authorization: Digest username="root", realm="AXIS_B8A44F279CC0", nonce="000027f1Y196406a8b21e4906a931de29db442c0366dfa0", uri="rtsp://10.176.12.49:554/axis-media/record/play.amp/", response="62aeb49a2051775714d409b17bda77d4"
User-Agent: openRTSP (LIVE555 Streaming Media v2025.01.17)
Session: Rs6kF2jSY7+nFd7i


Received 111 new bytes of response data.
Received a complete TEARDOWN response:
RTSP/1.0 454 Session Not Found
CSeq: 7
Server: GStreamer RTSP server
Date: Mon, 28 Apr 2025 06:25:54 GMT



E:\Vivek\Demo\OpenRTSP\live555-latest\live\testProgs>"C:\Program Files\VideoLAN\VLC\vlc.exe" vivekvideo-H264-1 --demux=h264

Output:

[JosephPtn]

"You can try building VLC after changing the"

Thank you for your answer, as far as I understand correctly I need to make changes to the code, from your photo I could not understand what exactly needs to be corrected

[vivekatoffice]

Hi @JosephPtn ,
I understand that is not so easy to modify the Live555(openRTSP) part in VLC code. 😞

[janssen70]

Can you add an example Url and which Axis OS version you are testing against?

[JosephPtn]

Example: rtsp://172.20.0.4/axis-media/media.amp?recordingid=20250425_144858_BF11_B8A44F20C59D

if the recording is completed, then specifying the vlc parameter ("--start-time=60) the recording is performed with an offset of the specified time in the parameter, if the recording is still in progress, the camera ignores this parameter and plays the recording from the beginning. Tried on several versions of Axis OS: 6, 9, 10, 11

[rocky-ji]

I have similar findings as your test in VLC, my OS is 12.4.59.

While when testing with ffplay, it can seek even the recoding ID is still recording. URL as following:
ffplay -ss 100 rtsp://root:pass@192.168.1.104/axis-media/media.amp?recordingid=20250427_061142_20B6_B8A44F284CF7

Not sure whether it is limitation in VLC or camera side.

[JosephPtn]

Thanks for the answer, I use vlc as a library connected to the project in android studio. It would be interesting to hear a comment from the Axis development team, and what exactly is this limitation.

[Cacsjep]

just for reference with this script, you can export all recordinds, just if any body needs it. I had made it for a customer, may it helps others to. =)

import argparse
import os
import sys
import requests
from requests.auth import HTTPDigestAuth
from urllib.parse import urljoin
from datetime import datetime
import xml.etree.ElementTree as ET
import time

def parse_arguments():
    parser = argparse.ArgumentParser(description='Export Axis camera recordings from SD card.')
    parser.add_argument('--ip', required=True, help='Camera IP address')
    parser.add_argument('--username', required=True, help='Camera username')
    parser.add_argument('--password', required=True, help='Camera password')
    parser.add_argument('--start', required=True, help='Start datetime in ISO 8601 format (e.g., 2025-05-01T00:00:00Z)')
    parser.add_argument('--end', required=True, help='End datetime in ISO 8601 format (e.g., 2025-05-21T23:59:59Z)')
    parser.add_argument('--output', default='recordings', help='Output directory for recordings')
    return parser.parse_args()

def list_recordings(base_url, auth, start_time, end_time):
    list_url = urljoin(base_url, '/axis-cgi/record/list.cgi')
    params = {
        'starttime': start_time,
        'endtime': end_time
    }
    response = requests.get(list_url, auth=auth, params=params)
    if response.status_code != 200:
        print(f"Failed to list recordings: {response.status_code}")
        sys.exit(1)
    return response.text

def parse_recordings(xml_data):
    recordings = []
    try:
        root = ET.fromstring(xml_data)
        recordings_node = root.find('recordings')
        if recordings_node is None:
            print("No <recordings> node found.")
            return recordings

        for recording in recordings_node.findall('recording'):
            recording_id = recording.attrib['recordingid']
            start_time = recording.attrib['starttime']
            end_time = recording.attrib['stoptime']
            recordings.append({
                'recording_id': recording_id,
                'start_time': start_time,
                'end_time': end_time
            })
    except ET.ParseError as e:
        print(f"Error parsing XML: {e}")
        sys.exit(1)
    return recordings

def export_recording(base_url, auth, recording, output_dir):
    export_url = urljoin(base_url, '/axis-cgi/record/export/exportrecording.cgi')

    params = {
        'schemaversion': '1',
        'recordingid': recording['recording_id'],
        'diskid': 'SD_DISK',
        'exportformat': 'matroska'
    }

    filename = f"{recording['recording_id']}.mkv"
    filepath = os.path.join(output_dir, filename)

    for attempt in range(1, 4):  # Try up to 3 times
        try:
            response = requests.get(export_url, auth=auth, params=params, stream=True, timeout=60)
            if response.status_code == 200:
                with open(filepath, 'wb') as f:
                    for chunk in response.iter_content(chunk_size=8192):
                        f.write(chunk)
                print(f"[SUCCESS] Exported recording to {filepath}")
                return True
            else:
                print(f"[Attempt {attempt}] Failed to export {recording['recording_id']}: {response.status_code}")
        except Exception as e:
            print(f"[Attempt {attempt}] Error exporting {recording['recording_id']}: {e}")
        time.sleep(2)  # wait before retry
    print(f"[FAILED] Recording {recording['recording_id']} failed after 3 attempts.")
    return False

def main():
    args = parse_arguments()
    base_url = f"http://{args.ip}"
    auth = HTTPDigestAuth(args.username, args.password)

    if not os.path.exists(args.output):
        os.makedirs(args.output)

    print("Listing recordings...")
    xml_data = list_recordings(base_url, auth, args.start, args.end)
    recordings = parse_recordings(xml_data)

    if not recordings:
        print("No recordings found in the specified date range.")
        return

    print(f"Found {len(recordings)} recordings. Starting export...")

    success_list = []
    failed_list = []

    for recording in recordings:
        success = export_recording(base_url, auth, recording, args.output)
        if success:
            success_list.append(recording['recording_id'])
        else:
            failed_list.append(recording['recording_id'])



    if success_list:
        print("\nSuccessful Exports:")
        for r in success_list:
            print(f"  - {r}")

    if failed_list:
        print("\nFailed Exports:")
        for r in failed_list:
            print(f"  - {r}")

    print("\n==== EXPORT SUMMARY ====")
    print(f"Total recordings found: {len(recordings)}")
    print(f"Succeeded: {len(success_list)}")
    print(f"Failed: {len(failed_list)}")

if __name__ == '__main__':
    main()

[vivekatoffice]

Feel free to re-open it OR share the solution/alternative/comment if already resolved👁️‍🗨️