CALL QUALITY TEST COMPLETE

VOIP CALL QUALITY TEST

Jitter, packet-delay variation & MOS over WebRTC

OFFLINE
--:--:--
01

CONNECTION

YOUR PUBLIC IP 216.73.217.103
Test server (echo)— connect to see
Your VoIP traffic leaves from—
Media path to the server—

These are the real IPs the call media uses (from the WebRTC candidate pair). Behind a VPN or NAT this can differ from your public IP above. Some browsers hide the on-device address as a .local name for privacy.

Signallingidle
ICE—
Connection—
Candidate—
Base RTT—
02

TEST

TEST DURATION
PACKET RATE (ptime)
PAYLOAD SIZE
CONCURRENT LOAD
Runs a bulk transfer alongside the probe — spikes usually appear only when the line is busy.
The test connects automatically and disconnects when it finishes. You can run 2 tests per day; you'll get a quick “I'm human” check the first time.
03

LIVE MONITOR

CURRENT RTT
--ms
BASE RTT
--ms
JITTER (RFC 3550)
--ms
MAX EXCURSION
--ms
p99 EXCURSION
--ms
LATE @60ms
--%
EFF LOSS @60ms
--%
LIVE MOS · G.711
--
Round-trip PDV — excursion above best-case (last 60s)
03b

SESSION

PACKETS
0
sent / echoed
LOSS
0.00
%
SPIKES >60ms
0
excursions
WORST SPIKE
0
ms
ELAPSED0s / 0s
08

TEST LOG · THIS CONNECTION

TimeKindLoadPathBase RTTJitterp99 excMax excLossSpikesBufferMOSGrade
No tests logged yet from this connection.
Every completed run from your IP (216.73.217.103), newest first. Kept for reference; the figures are round-trip.