Voice & Telephony Services

Call quality is maintained through multiple measures: our voice network uses dedicated low-latency paths between European points of presence and global carriers; we support TLS for signaling encryption and SRTP for media encryption; our infrastructure is fully redundant with automatic failover; and we continuously monitor call quality metrics (MOS scores, jitter, packet loss) with real-time alerts.

Yes. We support number porting (transferring your existing numbers from another provider to our platform) in most European countries and many international markets. The porting process typically takes 5-15 business days depending on the country and the releasing carrier. During the transition, there is no service interruption — your numbers continue to work throughout the porting process.

Our infrastructure supports thousands of concurrent calls per customer with no hard upper limit. SIP trunk capacity is flexible — you can scale from a few simultaneous calls to hundreds or thousands without hardware changes. Simply adjust your trunk configuration in the dashboard. Enterprise customers with very high concurrency requirements should contact our team for dedicated capacity planning.

Our platform works with any standards-compliant SIP device or software. This includes hardware IP phones (Yealink, Polycom, Grandstream, Cisco, Snom), desktop softphones (Zoiper, MicroSIP, Linphone, Bria), mobile apps (Onesip, Groundwire), and PBX systems (Asterisk, FreePBX, 3CX, and any SIP-compatible platform). If your device supports SIP, it works with Voiping.

Emergency calling availability depends on your number type and location. Geographic DID numbers in supported countries can be configured for emergency services. We recommend discussing your emergency calling requirements with our sales team during setup to ensure proper configuration and compliance with local regulations.