Telecom environments rely on SIP/RTP with diverse codecs, while transcription and analytics engines expect WebSocket or REST-based data streams. Converting and synchronizing this data is complex and error-prone.
For real-time use cases, latency adds another layer of difficulty since the voice data must be captured and delivered instantly to maintain accuracy and usability. And for regulated industries like finance or healthcare, compliance is critical. Sensitive voice data must be transmitted securely, without breaching privacy or regulatory standards.
In short: you need a secure, reliable bridge between your telephony systems and transcription engines – one that handles SIP complexity and data delivery so your team can focus on building insights, not managing telecom infrastructure.
It integrates natively with leading transcription platforms supporting partial result stabilization to deliver incremental updates and improve accuracy as conversations progress.
By abstracting SIP, call routing, media handling, and protocol conversion, the AI Connect Service removes the telecom complexity for you. Your developers can easily embed transcription into AI workflows without managing infrastructure – ensuring secure, reliable access to the audio your models need.
Built on a resilient, auto-scaling architecture, it handles large call volumes while maintaining consistent transcription latency and accuracy.
Explore the technical documentation.
A single integration point enables compatibility across a variety of voice platforms, reducing development time.
The AI Connect Service handles SIP protocols, call routing, and media handling, freeing your developers to build transcription models.
Fully compliant with regulations like GDPR and HIPAA, ensuring safe handling of sensitive voice data.
Automatically scales to meet peak traffic demands, with built-in resilience for high availability and consistent performance.
