Enterprise SIP Trunking
Scalable Voice Channels for Cloud PBX and Unified Communications
Elastic SIP trunking designed to connect on-premises and cloud-hosted PBX systems—such as Asterisk, FreeSWITCH, 3CX, Cisco CUCM, and Microsoft Teams—to the global PSTN. Delivers immediate channel elasticity, DID routing, and verified Caller ID presentation.
Architectural Capabilities & Features
PBX Interoperability
Certified interoperability across standard RFC 3261 IP-PBX systems, enterprise session border controllers, and contact center dialers.
Dynamic Concurrency
Instantaneous channel elasticity accommodating abrupt inbound and outbound traffic peaks without manual administrative provisioning.
Trunk Partitioning
Logically segment SIP credentials, concurrent channel limits, and rate thresholds by department, branch, or tenant.
Optional Media Encryption
Apply TLS signaling encryption and SRTP media payload encryption to secure corporate communications traversing public networks.
Technical Specifications
| Signaling Standards | SIP v2.0 (RFC 3261), WebRTC WSS |
| Supported Platforms | FreeSWITCH, Asterisk, 3CX, Cisco CUCM, Avaya, Kamailio |
| Channel Density | Scalable based on account allocation limits |
| Regulatory Notice | Wholesale voice transit service; does not supply residential E911 routing without explicit contractual configuration |