SIP & WebRTC Engine
Flexible call connectivity with carrier-agnostic SIP trunking and a browser-based WebRTC softphone. No PBX hardware required.
The Problem
Telephony infrastructure is the unglamorous foundation of every contact center, and for most enterprises it is also the most rigid and costly component. Legacy PBX hardware requires significant capital expenditure, demands specialised maintenance, and imposes hard limits on concurrent call capacity. SIP trunk relationships are often locked to a single carrier, leaving operations teams with no leverage during outages or price negotiations. When agents need to work remotely, organisations face the choice of deploying VPN-connected IP phones, purchasing and supporting hardware softphone licenses, or compromising call quality with consumer-grade solutions. Call quality monitoring is typically limited to post-call MOS score reporting, by which time the poor experience has already been delivered. Organisations running contact centers across multiple regions deal with the additional complexity of managing local SIP connections, number porting, regulatory compliance for telecommunications, and latency optimisation, all while keeping costs under control and maintaining the redundancy required for enterprise SLAs.
How Genesis AI solves it
The Genesis AI SIP and WebRTC Engine decouples your contact center from hardware dependency and carrier lock-in. On the SIP side, the platform supports multi-trunk configurations with carrier failover: you can connect multiple SIP trunk providers simultaneously, with automatic traffic routing to the lowest-cost or highest-quality path for each call. Direct inward dialling, toll-free number management, and PSTN connectivity are managed through a centralised trunk administration panel. The engine supports standard SIP signalling and is compatible with all major telecommunications carriers globally, with specific optimisations and pre-validated configurations for carriers serving the MENA region. On the WebRTC side, agents access a fully featured softphone directly in the browser with no plugins or installed clients. The WebRTC engine handles echo cancellation, noise suppression, jitter buffer management, and automatic bitrate adaptation to network conditions. Call quality is monitored continuously throughout every call, not just at the end, with real-time MOS score calculation, packet loss measurement, jitter tracking, and latency alerts. Quality events below threshold trigger automatic alerts to supervisors and are flagged in the post-call analytics record. Network diagnostics tools help agents and IT teams identify and resolve quality issues before they impact customers.
Key Benefits
Carrier-Agnostic Flexibility
Connect multiple SIP trunk providers simultaneously with automatic failover, eliminating single-carrier dependency and enabling competitive cost management.
No Hardware PBX Required
The cloud-native SIP engine replaces on-premises PBX infrastructure, eliminating capital expenditure and ongoing hardware maintenance costs.
Browser-Native Agent Softphone
WebRTC delivers enterprise-quality voice directly in the browser with adaptive echo cancellation and noise suppression. No client installation required.
Real-Time Call Quality Monitoring
Continuous in-call MOS scoring and packet loss tracking detect quality degradation before the call ends, enabling proactive intervention.
Global MENA Coverage
Pre-validated configurations for major carriers in Jordan, Saudi Arabia, UAE, Egypt, and across the MENA region simplify regional deployments.
Elastic Capacity
Scale concurrent call capacity up or down in minutes through trunk configuration changes, with no hardware provisioning or lead time.
What's included
- SIP trunk management with multi-carrier failover and direct PSTN connectivity
- Browser-based WebRTC softphone with adaptive audio quality
- Real-time call quality monitoring with in-call MOS scoring and alerts
Frequently Asked Questions
Related Modules
AI Voice Agents
AI handles 100% of inbound calls with native support for 20+ Arabic dialects and intelligent human handoff when it matters.
Agent & Supervisor Portal
One portal for everything: softphone, call controls, live transcripts, AI summaries, and real-time supervisor dashboards, all in the browser.
Compliance & Security
Enterprise-grade security with HIPAA, GDPR, and PCI-DSS compliance built in from the ground up, not bolted on as an afterthought.
Ready to see SIP & WebRTC Engine in action?
Book a personalised demo and see exactly how this module fits into your contact center operation.