SIP door station setup

Debug mode on NovoSIP web interface

In normal operation, our NovoSIP doorphone ensures quality communication without delay between pressing the call button with great image quality provided by the camera of the Raspberry Pi.

Computer networks can be complex and may induce undesirable behaviors such as important reaction times, video or audio stream of poor quality etc …
For this reason we integrated an option box to generate a log file containing the network behavior data.

For those interested, following a discussion between a client and our support for the analysis and resolution of unusual behavior:

Elements Here are some statistics on flows received by the PC:


Max delta = 2039.48 ms packet at no. 4726
Max Jitter = 1.21 ms. Mean = 0.54 ms jitter.
Max skew = -10.39 ms.
Total RTP packets = 162 (expected 162) Lost RTP packets = 15 (9.26%) Sequence errors = 13
Duration 32.64 s (-22 323 ms clock drift, Corresponding to 28450 Hz (-68.39%)

10% loss on the video stream: this means in reality of very large losses (30%, 50% view, images can be reconstructed by the SIP application).


Max delta = 79.87 ms packet at no. 4330
Max Jitter = 3.02 ms. Mean = 0.33 ms jitter.
Max skew = -20.26 ms.
Total RTP packets = 1770 (expected in 1770) Lost RTP packets = 128 (7.23%) Sequence errors = 120
Duration 35.38 s (-6754 ms clock drift, Corresponding to 6473 Hz (-19.09%)

We see also more than 7% of packets lost. what is more alarming is the “drift”: time elapsed 20% slower than expected …

My first impression is that the network is saturated.

With “Voip by Antisip PC” so you have to configure the video card to use “static picture” to minimize bandwidth video to avoid overloading the network with a non-required flow.

If such a gap exists is perhaps the CPU was running full speed. If there are too many losses networks, it is possible that the system tries to compensate with the PLC. The PLC being “overused”, the doorman did not have time and capture is no longer continuous (more real-time) this can lead to this situation or system only produces 80% of its time. Added to this, the loss of 10%, and the end is missing 30% of the audio …


1 / I think we should look by logging onto the doorman if that is the case

 -> Check CPU usage 100%? during calls.

2 / Use only “static picture” for Voip By Antisip PC. (Minimize the network impact and the impact on the CPU board.) This should already improve.

 -> Check CPU usage 100%? during calls.

3 / Thirdly, try to solve the problem of network loss.

 -> Check CPU usage 100%? during calls.

Leave a Reply