
Contact center QA teams evaluate 1 to 5% of calls manually. QEval eliminates that bottleneck by applying AI speech analytics and automated scoring to 100% of interactions across voice, chat, and email, using a classification engine trained on 138M+ real conversations.
Capabilities span quality monitoring, compliance detection for PCI, HIPAA, and GDPR at 98% accuracy, sentiment analysis, keyword identification, agent coaching workflows, performance gamification, and predictive analytics across 110+ configurable dashboards. Quality scoring runs at 94% accuracy with zero manual intervention.
Deployment takes 30 days. Industry standard is 90 to 120. No disruption to live operations. Etech Global Services built QEval from two decades of running Fortune 500 contact centers in healthcare, telecom, retail, banking, and BPO. ISO 27001, SOC 2, PCI-DSS certified. Built for QA leaders and operations teams scaling coverage without adding headcount.
QEval also provides call recording management, screen capture, custom evaluation forms, calibration tools for QA consistency, root cause analysis, trend identification, and automated alert systems for compliance breaches. The voice of customer module tracks customer sentiment across touchpoints to identify service gaps and training opportunities. Real-time monitoring lets supervisors intervene during live interactions. Role-based access controls, audit trails, and data encryption ensure enterprise-grade security. QEval supports multi-site and multilingual contact center environments with centralized reporting across locations.
API integrations connect QEval with existing CRM, telephony, and workforce management systems. Automated report scheduling delivers insights to stakeholders without manual effort.
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An API powered by Google's AI technology allows you to accurately convert speech into text. You can accurately caption your content, provide a better user experience with products using voice commands, and gain insight from customer interactions to improve your service. Google's deep learning neural network algorithms are the most advanced in automatic speech recognition (ASR). Speech-to-Text allows for experimentation, creation, management, and customization of custom resources. You can deploy speech recognition wherever you need it, whether it's in the cloud using the API or on-premises using Speech-to-Text O-Prem. You can customize speech recognition to translate domain-specific terms or rare words. Automated conversion of spoken numbers into addresses, years and currencies. Our user interface makes it easy to experiment with your speech audio.
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Fish Audio
Fish Audio delivers cutting-edge AI-driven technologies for text-to-speech (TTS), voice replication, and speech recognition (STT). This platform caters to businesses and developers aiming to incorporate lifelike voice generation into their software applications. With its advanced voice cloning capabilities, users can easily mimic specific voices, while the generative AI can generate expressive and natural speech across various languages. Moreover, Fish Audio features an API that facilitates seamless integration, along with enhanced functionalities like voice activity detection. This versatility makes Fish Audio an invaluable resource for diverse sectors, including content production, virtual assistant development, and customer service enhancements, ensuring that users can engage their audiences effectively. It stands out as a comprehensive solution for anyone seeking to elevate their audio-related projects with sophisticated technology.
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Qwen3-TTS
Qwen3-TTS represents an innovative collection of advanced text-to-speech models created by the Qwen team at Alibaba Cloud, released under the Apache-2.0 license, which delivers stable, expressive, and real-time speech output with functionalities like voice cloning, voice design, and precise control over prosody and acoustic features. This suite supports ten prominent languages—Chinese, English, Japanese, Korean, German, French, Russian, Portuguese, Spanish, and Italian—along with various dialect-specific voice profiles, enabling adaptive management of tone, speech rate, and emotional delivery tailored to text semantics and user instructions. The architecture of Qwen3-TTS incorporates efficient tokenization and a dual-track design, facilitating ultra-low-latency streaming synthesis, with the first audio packet generated in approximately 97 milliseconds, making it ideal for interactive and real-time applications. Additionally, the range of models available offers diverse capabilities, such as rapid three-second voice cloning, customization of voice timbres, and voice design based on given instructions, ensuring versatility for users in many different scenarios. This flexibility in design and performance highlights the model's potential for a wide array of applications in both commercial and personal contexts.
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