Scalable Voice API for worldwide calling. Build fast, reliable, and secure voice capabilities with our powerful REST APIs.
Voice challenges that hurt your business.
Poor call quality
Complex infrastructure
Limited global reach
The unified voice solution you need.
Enterprise-grade quality
Complete Voice suite
Global coverage
Crystal-clear connections worldwide.
High-quality calling
Smart Routing Engine
Use direct carrier connectivity to ensure high-quality calls with optimal routing.
Global infrastructure
Access reliable voice services across 140+ countries with carrier-grade network quality.
Performance monitoring
Track call duration, volume, and quality metrics to optimize voice operations continuously.
Build sophisticated voice experiences.
Advanced features
Optimize with comprehensive insights.
Advanced analytics
Power critical communications across your business.
Use cases
Connect customers instantly
Make and receive high-quality calls with enterprise-grade reliability and features.
Automate support
Deploy IVR systems to route calls efficiently and provide self-service options.
Protect privacy
Use masked calling to secure customer and employee information during transactions.
Verify identity
Implement voice-based authentication and two-factor verification systems.
Drive sales calls
Connect prospects with sales teams through reliable, professional calling solutions.
Scale call operations
Handle high call volumes with automated routing and intelligent distribution.
Protect your communications and your customers.
Compliance
Trusted by companies that demand results.
Frequently asked questions
The Voice API supports G.711 (PCMU/PCMA), G.729, and Opus codecs. WebRTC connections use Opus by default for optimal quality at variable bandwidth. SIP trunking supports G.711 and G.729 with automatic codec negotiation.
Transcription accuracy varies by language and audio quality — English transcription achieves 90-95% word accuracy on clean audio with speaker diarization. The system supports 30+ languages with automatic language detection and returns interim results with less than 500ms latency.
Yes. Elastic SIP trunking connects to any standards-compliant PBX, SBC, or contact center platform. Configuration requires your PBX IP address and a Bird SIP endpoint — no hardware changes needed. Trunks scale automatically and support TLS and SRTP encryption.
Conference calls support up to 250 concurrent participants with moderator controls (mute, hold, remove). Each participant can be recorded on a separate audio track. Real-time events notify your application when participants join, leave, or are muted.
Voice API pricing is per-minute based on the call destination, with separate rates for inbound and outbound legs. Recording storage, transcription, and phone numbers are billed separately. There are no monthly minimums — you pay only for what you use.