SIP Trunking
SIP Trunking That Doesn't Flinch Under Pressure
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SIP Trunking — a Voice over IP (VoIP) technology that uses the Session Initiation Protocol (SIP) to deliver telephone calls and unified communications over an internet connection, replacing traditional analog phone lines or ISDN PRI circuits. SIP trunks connect an on-premise or cloud-based PBX to the public switched telephone network (PSTN), enabling businesses to retain existing phone hardware while reducing line costs and gaining scalable, on-demand call capacity.
<50ms
Latency
Automatic
Failover
G.711/G.729
Codec Support
US + Intl
Coverage
🛡️ AureVox SIP trunks are compatible with Cisco, Avaya, 3CX, FreePBX, and Asterisk. All deployments include 24/7 NOC monitoring by AureVox SIP engineers.
Last updated: Q1 2026
Specifications
Engineered to Enterprise Spec
Concurrent Channels
Scalable on demand — add capacity in minutes, not weeks.
Codec Support
G.711 (uLaw/aLaw), G.729, G.722 HD Voice
Failover
Active/passive redundancy. Automatic failover — zero manual intervention.
Latency
Sub-50ms typical. Engineered for real-time voice, not afterthoughts.
Compatibility
Cisco, Avaya, 3CX, FreePBX, Asterisk, and most SIP-standard PBX systems.
Transport
SIP over UDP, TCP, and TLS. SRTP media encryption available.
Geographic Coverage
US, Canada, and international DID origination and termination.
Support
24/7 US-based NOC with dedicated SIP engineering team.
Use Cases
Built for the Environments That Can't Afford Failure
Legacy PBX Modernization
Keep your existing on-premise PBX investment but move to modern SIP trunking. Eliminate analog lines and ISDN PRI without replacing hardware.
Contact Center Connectivity
High-volume inbound and outbound trunking for call centers. Burst capacity, queue management, and sub-second failover.
Disaster Recovery Voice Routing
Automatic failover to backup routing when primary infrastructure is unreachable. Keep your numbers live even when your data center isn't.
Compatible with
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Ready to modernize your phone lines?
Our engineers will assess your current setup and design the right solution for your call volume — no jargon, no guesswork.
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