Wowza Community

Progressive delay with audio/video streaming

We need to live stream one of our netcams.

I set up Wowza to receive an RTSP stream (video H.264 + audio G.711) and broadcast it in RTMP.

It works but the audio and video get out of sync progressively, about 1 sec every 30 min (in LAN).

With AVSyncMethod senderreport I get:

WARN server comment - Waiting for RTCP packet. See docs for (Application.xml: RTP/AVSyncMethod and RTP/MaxRTCPWaitTime).
WARN server comment - RTCPHandler.isTimeSyncReady: Hit MaxRTCPWaitTime synchronizing on system clock.

I’m trying with rtptimecode but it doesn’t seem better.

If I change MaxRTCPWaitTime Wowza can’t connect to the netcam stream.

Any idea?

Hi belcom,

You can set up you’re application to debug timecodes to test if they are aligned with this article:

How to debug AAC or MP3 timecode issues with cupertino packetization

If your timecodes are aligned then a server side sort buffer might help the drift issue:

How to troubleshoot live streaming

Salvadore

Hi,

You can add additional logging to the RTP/RTSP stream using the Property found here,

https://www.wowza.com/docs/how-to-re-stream-video-from-an-ip-camera-rtsp-rtp-re-streaming#extraLog

When you have added the Property in the link above please open a support ticket by emailing support@wowza.com

Please include a brief description of the problem, a zipped/compressed copy of your [Wowza-Install]/conf and [Wowza-Install]/logs directories for analysis and a link to this thread for reference.

Thanks

Jason

Belcom, I’m sorry, I did not notice you are using G.711 audio. I over looked that detail.

G.711 works in recent Flash versions(11+). It is supported in by Flash RTMP and Flash HTTP players. You can publish and play it back, with that limitation.

Try playing it back with the [install-dir] /examples/LiveVideoStreaming/FlashRTMPPlayer. Make sure you have up to date version of Flash.

You might want to use a different audio format, ACC or mp3. Wowza Transcoder can transcode G.711 audio to ACC or mp3.

One other idea would be to lower the bitrate of the stream and see if that helps.

Salvadore

Belcom,

The Wowza Transcoder is available with a 30 day trail, daily license and monthly license, all you have to do to use it is set it up and go. If it is a paid license you are running on then you will be billed according to this price guide

For Perpetual license, you can purchase addons here:

https://perpetualstore.wowza.com/pro…prod=perp3-ser

You can request a free 30 day trial that includes Wowza transcoder here

http://www.wowza.com/pricing/trial

Salvadore

Hi Salvadore.

You can set up you’re application to debug timecodes to test if they are aligned with this article:

How to debug AAC or MP3 timecode issues with cupertino packetization

The audio codec of our netcam is G.711 so I don’t think the properties debugAACTimecodes or debugMP3Timecodes are valid…

If your timecodes are aligned then a server side sort buffer might help the drift issue:

How to troubleshoot live streaming

I’m trying sortBufferSize … but it seems that Wowza can’t connect to the netcam with this option.

I think the point is about the response…

WARN server comment - Waiting for RTCP packet. See docs for (Application.xml: RTP/AVSyncMethod and RTP/MaxRTCPWaitTime).

WARN server comment - RTCPHandler.isTimeSyncReady: Hit MaxRTCPWaitTime synchronizing on system clock.

If I disable the audio on the netcam there are no warnings of this kind.

Thank you Salvadore and JasonH for your tips.

Some points:

  • FlashRTMPPlayer… I’m using FlowPlayer. But I don’t think the problem is there, if I re-open the web page or I connect 1 hour later the audio and video are out of sync

  • Wowza Transcoder…

http://www.wowza.com/addons/wowza-transcoder-addon

Do you know if it is available for trial? I don’t see a Download link

  • Lower bitrate… I’m using a video stream H.264 Baseline with 640x480 and 800 kbps. I don’t think it is high…

  • extra log property… interesting, I’ll try that

BTW, I tried a netcam AXIS with AAC and works fine.

But our products (not AXIS) has only the G.711 audio codec…