RTSP stream failed

[revanreddy]

Hi,

RTSP Streaming is failed using flashphoner Embed Player and Player example on demo site

Error as shown in the attachment

 

[Max]

Good day.
First, to play via unsecure connection, you must disable WebRTC switch, click Test button and then play the stream

Also, seems like your stream can not be connected

 

[Max]

We checked your report. Seems like the RTSP source does not support H264 codec with 90000 Hz sample rate:

We see the following message in server log

10:22:15,704 ERROR         MediaHandler - pool-50-thread-1 No supported codecs found!

Here is the example of SDP form RTSP source supporting the codec needed:

 

[revanreddy]

We checked your report. Seems like the RTSP source does not support H264 codec with 90000 Hz sample rate:

That means, is our rtsp stream giving more resolution than what Flashphoner supports ?

Please let us know the stream range supported by Flashphoner.

Thanks

 

[Max]

That means, is our rtsp stream giving more resolution than what Flashphoner supports ?

No.

Your RTSP server says:

a=rtpmap:98 H264/1000

WCS supports

a=rtpmap:98 H264/90000

90000 means clock rate. 90k samples per second.
90k samples will be sent per second. It is measure of packetization.

Default clock rate is 90000 and this clock rate is supported in the most of applications.

1000 clock rate is not common. And this clock rate is not supported in the WCS server.

So it is not related to video resolution. It is clock rate and it should be 90000. Try to configure your RTSP source to this clock rate.

 

[revanreddy]

Thanks Max,

We have fixed the clock rate issue at rtsp source and we are able to get the stream now we are experiencing the delay about 3 mins in stream

We are providing the report here

Thanks

 

[Max]

Did you see the same delay using VLC player?
If VLC works without delay, please capture pcap dump using Wireshark and VLC player.

1. Wireshark start capture.
2. Start VLC playback.

Need to capture all RTSP traffic.

 

[Max]

We checked the report. There seems to be two issues:
1. You're playing the stream via MSE. This can add up to 3 seconds delay because TCP (websocket) is used as media packets transport.
2. According to client logs, the RTSP stream has timing problem. The client log contains a lot of the following messages:

17:48:10,321 WARN   BitstreamNormalizer - pool-284-thread-1 Timing problem: Last received frame time 2411741900; syncTime - 1655156890311.
Current frame time - 2411741933; syncTime - 1655156890311; frame type - videoFrameKey
Increasing on 90 samples

Seems like every frame timestamp is very different from syncronization time value in the same frame. Please check the RTSP source settings to fix this.
Also, please check how VLC plays the same RTSP stream. If it plays correctly, collect traffic dump using Wireshark on the same PC where VLC is running:
1. Start Wireshark
2. Start traffic collection in Wireshark
3. Start VLC
4. Open RTSP link in VLC, make sure it is playing correctly
5. One minute later stop VLC
6. Stop traffic collection in Wireshark
7. Save the traffic dump collected as tcpdump pcap.
Then send the traffic dump using this form. If the traffic dump file size is more than 30 M, place it to a cloud drive (GoogleDrive, OneDrive etc) and send the link using Comment field of this form.

 

[revanreddy]

Max,

We have submitted the Wireshark dump using form.

we are seeing that java is using 100+% of CPU and stays at this high level even when no streaming is occurring

please find the attachment

 

[Max]

Unfortunately, you've probably cut the beginning of translation from the dump (RTSP handshake), so we can't play this dump for test. Please repeat the dump collection.
Also we recommend to check RTSP camera settings. Seems like you've fixed clock rate to 90000 (as required) in SDP only, but camera still generates frames with clock rate 1000. This can be a reason for delay. Please correct the clock setup or use another camera.

we are seeing that java is using 100+% of CPU and stays at this high level even when no streaming is occurring

The issue was fixed in latest builds, please update to 5.2.1252 or later.

 

[revanreddy]

Max,

As we mentioned earlier we are facing the 3 mins delay in streaming in our application and demo app as well. even after 3 mins delay streaming is not smooth.

our observations are:

1. Flashphoner(WCS) using TCP for transport as default, we suspect that causing the initial delay to pick up the stream.

2. When we tried the RTSP stream from the same source in VLC it streams smoothly without any delay and it uses UDP for transport

Now

Do we have any way(i.e configuration change) to use UDP instead of TCP to get stream without any delay?

Please give us your inputs to overcome this issue.

Thanks

 

[Max]

1. Flashphoner(WCS) using TCP for transport as default, we suspect that causing the initial delay to pick up the stream.
2. When we tried the RTSP stream from the same source in VLC it streams smoothly without any delay and it uses UDP for transport

Try the following parameter

rtsp_interleaved_mode=false

In this case, WCS tries to capture RTSP stream as non-interleaved via UDP if camera supports it.
Anyway, TCP gives a maximum delay up to 3 seconds, not 3 minutes. So we still suppose a problem with stream itself. Please provide a public RTSP link to test using this form.
If this is not possible, collect the traffic dump at VLC side as described in this post in details. The dump must include RTSP handshake (DESCRIBE->OPTIONS->PLAY) otherwise we cannot play in on test stand. Send the traffic dump using this form. If the traffic dump file size is more than 30 M, place it to a cloud drive (GoogleDrive, OneDrive etc) and send the link using Comment field of this form.