Wireshark · Wireshark-users: [Wireshark-users] Debugging intermittent audio dropout with RTP over OpenVPN

[Andrew Martin <amartin@xxxxxxxxxxx>]

Hello,

I am using an Asterisk server and Yealink SIP phones (e.g SIP-T32G) both on a
LAN as well as a few phones connecting over the Internet via an OpenVPN
connection (so that NAT is not required). The phones on the LAN work well,
however calls involving the phones connecting over the Internet intermittently
experience audio quality problems (e.g. calls will drop audio for 15 seconds or
audio will be very garbled when someone talks). This is very intermittent;
sometimes a call is completely fine. I have performed some pcap captures of
calls that are having problems, with the resulting RTP statistics:

Call 1:
Max delta: 20.77ms
Max jitter: 0.24ms
Mean jitter: 0.14ms
Max skew: 1.37ms
Total RTP Packets: 3030 (expected 3030)
Lost RTP Packets: 95 (3.14%)
Duration: 99.21s (-3792ms clock drift)

Call 2:
Max delta: 119.14ms
Max jitter: 23.56ms
Mean jitter: 4.24ms
Max skew: -228.31ms
Total RTP Packets: 7668 (expected 7668)
Lost RTP packets: 37 (0.48%)
Duration: 261.06s (-243ms clock drift)

Call 3:
Max delta: 112.05ms
Max jitter: 7.73ms
Mean jitter: 3.40ms
Max skew: -46.48ms
Total RTP Packets: 4055 (expected 4055)
Lost RTP packets: 33 (0.81%)
Duration: 162.59s (-25460ms clock drift)

Looking at statistics from the router (where the OpenVPN server is also running)
I cannot see any packet loss or other problems. What else can I try to narrow 
down the source of these VoIP quality problems? Is there other information I can
provide or that I should look for to help identify the problem?

Thanks,

Andrew Martin