Legacy PBX ran on dedicated circuits. SIP trunking and cloud voice run on your data network. That network was never designed for real-time voice. MCS tells you whether it's ready, and exactly where it isn't.
When voice ran on dedicated PRI or T1 circuits, quality was guaranteed by the carrier. There was no contention, no jitter, no packet loss. The circuit was yours.
SIP trunking and cloud PBX push voice onto a shared data network. That network carries everything else too: file transfers, backups, video streams, cloud sync. Voice now competes for bandwidth, and voice always loses when the network is congested.
The worst time to discover this is after cutover, when the PRI lines are gone and there's no fallback.
A successful SIP migration depends on more than bandwidth. MCS tests the specific conditions that determine whether voice will work on your data network.
Simulate full SIP call setup, session, and teardown. Measure response times for each transaction and detect failures before they affect production calls.
Automatically identify SIP Application Layer Gateway interference, the single most common and hardest-to-diagnose cause of one-way audio and registration failures.
Upstream and downstream, with distribution analysis. Know whether loss is random or burst-based. Burst loss destroys voice quality far more than spread loss.
Industry-standard Mean Opinion Score calculated from real network conditions. A MOS below 3.5 means users will notice. Below 3.0, they'll complain.
Test whether SIP signaling ports (TCP/UDP 5060) and RTP media ports are open and reachable through every firewall, NAT device, and SBC in the path.
Test with G.711, G.729, and configurable packet rates to match your actual trunk provider's codec requirements. Different codecs have different network demands.
Measure sustained throughput, not burst speed. Know whether your connection can maintain voice quality when the network is under normal business-hour load.
Bidirectional traceroute reveals the path between your site and the SIP trunk provider. Identify routing inefficiencies, asymmetric paths, and high-latency hops.
Confirm that QoS markings are being applied and honored across the network path. Unmarked or stripped DSCP bits mean voice gets no priority.
A voice migration has distinct phases. Network quality should be validated at each one, not just once before cutover.
Before committing to a migration timeline, test every site that will move to SIP. Measure jitter, loss, MOS, SIP responsiveness, and port availability. Identify sites that need network remediation before they can support voice.
After network changes are made (QoS configured, ports opened, SIP ALG disabled), run the tests again. Confirm that remediation worked. Run automated tests for a baseline period to catch intermittent issues.
After cutover, schedule continuous automated tests from every migrated site. Establish quality baselines and get alerted to degradation before users start calling the help desk.
Migrating from on-premise PBX to cloud voice or SIP trunking across multiple sites. Validate every location before and after cutover.
Qualify customer sites before provisioning trunks. Prove the network can support your service, or prove it can't, before you get blamed for quality issues.
Managing PBX-to-cloud migrations for clients. Include site qualification as a standard part of every project to eliminate post-migration support escalations.
Validate media quality across SIP interconnects, peering points, and customer handoffs. Prove SLA compliance with hard data.
Every failed voice migration has the same root cause: someone assumed the network would handle it. MCS replaces that assumption with data.
