Voice & Unified Communications
Reliable SIP trunking for scalable voice communication and modern unified communication environments.
Overview
Our SIP trunking services provide flexible and scalable voice connectivity built for modern communication systems. We support migration from legacy circuits, deliver high-quality call routing, and ensure seamless integration with existing platforms and unified communication environments.
With reliable capacity management, secure call handling, and multi-region reach, we help organizations improve communication performance while reducing operational cost and removing the limitations of traditional telephony.
Our SIP trunks work with standard SIP protocols including PJSIP, chan_sip, and other implementations. Connect Asterisk, FreePBX, Cisco, Avaya, Mitel, 3CX, and other PBX platforms without proprietary requirements. If your system speaks SIP, it works with our trunks.
For developers and organizations with programmatic requirements, REST APIs provide access to voice capabilities. Automate number provisioning, manage call routing rules, configure trunks, and integrate telephony into your applications and business workflows.
Features
Wideband codecs (G.722, Opus) deliver crystal clear audio quality for professional communications.
Automatic recording with configurable formats and storage options for compliance and training.
Route calls by quality, lowest cost, or specialized traffic types with flexible rule configuration.
Filter unwanted calls and protect against fraud with inbound screening and spam detection.
Detailed call analytics and troubleshooting tools through the management portal.
Emergency calling compliant with Kari's Law and RAY BAUM's Act with location registration.
Capabilities
Transition from PRI, T1, and analog lines with number porting and minimal disruption.
Microsoft Teams, Zoom, and Genesys direct routing to connect platforms to PSTN.
Local and toll-free numbers across US, Canada, and international markets.
Time-based routing rules with automatic failover to backup destinations.
TLS/SRTP encryption and fraud detection for secure call handling.
Add or remove trunks in seconds without circuit orders or contracts.
Related Infrastructure
Integrate wireless with dedicated fiber and global network infrastructure for complete connectivity.
Explore Global Network →24/7 SOC/NOC monitoring for your entire wireless and network infrastructure.
Explore Monitoring →Fast and flexible wireless connectivity for your modern digital enviroments.
Explore Wireless Solutions →FAQ
We support IP-based authentication (IP ACL), digest authentication with username/password credentials, or both combined. For IP auth, whitelist your PBX public IP. For credential auth, configure the provided username and password in your trunk settings. TLS with certificate validation is available for environments requiring encrypted signaling.
G.711 (PCMU/PCMA), G.722, G.729, and Opus. For HD voice, configure G.722 or Opus as the preferred codec in your PBX. We recommend G.711 for compatibility and G.722 for quality when bandwidth permits. Codec negotiation follows standard SDP offer/answer.
If your PBX is behind NAT, enable STUN or configure a static public IP mapping. Set your external media IP in the PBX SIP settings. We support rport for NAT detection and can work with most NAT configurations. For persistent NAT issues, consider placing your PBX in a DMZ or using a session border controller.
SIP signaling uses UDP/TCP 5060 or TLS 5061. RTP media uses UDP ports 10000-20000 (configurable). Ensure your firewall allows outbound to our SIP endpoints and permits return traffic. For TLS, use port 5061 and verify certificate chain in your PBX configuration.
We provide multiple SIP proxy endpoints for DNS SRV-based failover. Configure SRV lookup in your PBX to automatically route to available endpoints. For active-passive setups, configure primary and secondary trunks with priority weighting. We also support outbound failover rules at the platform level if your PBX doesn't handle SRV.
Yes. Custom headers can be passed for caller ID manipulation, call tagging, or integration with downstream systems. Configure X-headers in your PBX dial plan. Our API also allows setting headers programmatically for advanced routing or analytics use cases.
Contact us to discuss your current telephony environment and how SIP trunking can reduce costs and support your unified communications strategy.