Teams, Zoom, Webex, and every other real-time media platform have one thing in common: they need a network that can deliver packets consistently, in order, and on time. Not just fast. Consistent.
Most organizations roll out video conferencing across dozens or hundreds of sites and assume the network will handle it. When it doesn't, they spend months chasing complaints site by site. The network should have been validated first.
These are the moments when video conferencing quality is most likely to break. MCS lets you catch issues at each one, before users do.
Deploying Teams, Zoom, or Webex across the organization? Validate every site before go-live so day one works, not day thirty after escalations.
Switching ISPs, adding MPLS, or migrating to SD-WAN? Prove the new path supports real-time media before you cut over production traffic.
Opening a new office or migrating an existing one? Test the local network, the WAN path, and the last mile before staff arrive and start complaining.
Home networks are unpredictable. Test each remote worker's connection to confirm it can sustain video calls, screen sharing, and real-time collaboration.
Networks degrade over time. Schedule automated tests from every site to catch intermittent jitter, congestion, and routing changes before they impact meetings.
"My Zoom keeps freezing." Instead of guessing, run a test from that user's location and get hard data on jitter, loss, and route quality in minutes.
Video conferencing platforms don't just need bandwidth. They need consistent, low-latency packet delivery. MCS measures the specific metrics that determine whether real-time media will work.
Variation in packet arrival time. Even small amounts cause audio distortion, video stuttering, and lip-sync issues in conference calls.
Missing packets mean missing audio frames and corrupted video. MCS measures upstream and downstream loss separately to pinpoint the direction.
Mean Opinion Score rates call quality on a 1–5 scale. MCS calculates MOS from real network conditions so you know exactly how calls will sound.
High or erratic latency creates talk-over and awkward pauses. MCS measures min, average, max, and consistency across the full path.
Out-of-order packets force the receiver to reorder or discard, adding delay and degrading audio/video playback quality.
Is packet loss spread evenly or concentrated in bursts? Burst loss is far more destructive to media quality. MCS shows the pattern.
Teams, Zoom, and Webex require specific UDP port ranges. MCS tests whether those ports are open and reachable through every firewall in the path.
Bidirectional traceroute reveals the exact network path and identifies where latency, loss, or routing inefficiencies occur.
Video conferencing at scale requires sustained throughput, not just peak speed. MCS measures real-world TCP equilibrium throughput and UDP capacity.
Every major video conferencing platform publishes network requirements. MCS tests your actual network against those thresholds at every site, so you know where you stand before you deploy.
Requires <30ms jitter, <1% packet loss, and <50ms one-way latency for optimal audio. Video needs 1.5–4 Mbps sustained. UDP ports 3478–3481 and 49152–65535 must be open.
Recommends <40ms jitter, <1% packet loss, and latency under 150ms. HD video requires 3.0 Mbps up/down. UDP port 8801–8810 range is preferred.
Similar thresholds across all real-time platforms. MCS gives you the raw data to validate against any vendor's requirements, current or future.
MCS doesn't wait for trouble tickets. Deploy test points at every office, data center, and remote location, then measure real media quality from each one. On-demand or automated, 24/7.