Fixie AI
AgentFreePlatform for deploying conversational AI agents.
Capabilities10 decomposed
speech-native real-time voice conversation with paralinguistic preservation
Medium confidenceProcesses raw audio input directly through an end-to-end trained speech-native model (ultravox-v0.7) that preserves tone, cadence, pitch, and emotional prosody without intermediate text transcription. Outputs audio responses with integrated text-to-speech, enabling natural conversational flow at sub-second latencies. The model operates on dedicated, purpose-built inference infrastructure managed by Ultravox, not via external LLM API calls.
End-to-end speech-native model trained directly on audio (not transcription-based), preserving paralinguistic signals (tone, cadence, pitch) that are lost in traditional speech-to-text pipelines. Dedicated inference infrastructure with response times faster than GPT-4, Gemini Live, and Claude Sonnet 4.5 per published benchmarks.
Faster and more natural than transcription-based voice AI (e.g., OpenAI Whisper + GPT-4 + TTS) because it eliminates intermediate text conversion and operates on audio natively; more responsive than Gemini Live or Claude Sonnet 4.5 for real-time voice interactions.
managed telephony integration with major carriers
Medium confidenceProvides built-in, pre-configured integrations with 'largest telephony providers' (specific providers not named in documentation) to route inbound and outbound calls directly to Ultravox voice models. Handles SIP, PSTN, and VoIP protocols transparently; developers configure telephony routing via REST API without managing carrier connections or call signaling directly.
Pre-built telephony integrations eliminate the need for custom SIP/PSTN configuration; developers use REST APIs to route calls to voice models without managing carrier connections, call signaling, or infrastructure. Abstracts away telephony complexity entirely.
Simpler than building custom Twilio + LLM integrations because telephony is native to the platform; faster to deploy than self-managed SIP/PSTN solutions because carriers are pre-integrated.
rest api and multi-platform sdk access with real-time streaming
Medium confidenceExposes Ultravox voice models via REST APIs and native SDKs for web, mobile (iOS/Android), and backend platforms. Supports both request-response (single turn) and WebSocket streaming (continuous conversation) patterns. SDKs handle audio encoding/decoding, session management, and error handling transparently; developers interact with simple function calls rather than raw HTTP.
Native SDKs for major platforms (web, iOS, Android, backend) abstract away audio codec handling and WebSocket management; developers use simple function calls instead of raw HTTP. Supports both synchronous request-response and asynchronous streaming patterns.
Easier to integrate than raw REST APIs because SDKs handle audio encoding/decoding and session management; faster to deploy than building custom WebSocket clients for streaming voice.
low-latency inference with real-time response benchmarking
Medium confidenceUltravox v0.7 model runs on dedicated, purpose-built inference infrastructure optimized for sub-second response times. Published benchmarks show response latency faster than GPT-4, Gemini Live, and Claude Sonnet 4.5 on Big Bench Audio tasks (84% pass rate at fastest latency tier). Latency is a first-class optimization metric; specific millisecond latencies not published, but positioning emphasizes speed over accuracy trade-offs.
Dedicated inference infrastructure optimized for latency-first performance; published benchmarks show faster response times than GPT-4, Gemini Live, and Claude Sonnet 4.5. Explicit latency/accuracy trade-off positioning (84% accuracy at fastest speed vs. higher accuracy at slower speeds).
Faster than LLM-based voice pipelines (Whisper + GPT-4 + TTS) because inference is native and not chained; more responsive than Gemini Live or Claude Sonnet 4.5 for real-time voice, per published benchmarks.
tiered concurrency and pricing model with per-minute metering
Medium confidenceUltravox uses a simple per-minute pricing model ($0.05/minute for all usage including TTS) with concurrency limits tied to subscription tier. Free tier: 5 concurrent calls; Pro tier: $100/month (annual) with higher concurrency; Enterprise: custom concurrency and pricing. Metering is transparent and usage-based — no per-call, per-token, or per-interaction surcharges documented.
Simple per-minute pricing ($0.05/min) with no per-token, per-call, or per-interaction surcharges; TTS included in base rate. Concurrency limits tied to subscription tier, enabling free tier experimentation and clear upgrade path to production.
More transparent than LLM-based pricing (e.g., OpenAI's per-token model) because per-minute metering is predictable; simpler than Twilio + LLM combinations that require separate billing for telephony, transcription, and inference.
context and session management across multi-turn conversations
Medium confidenceUltravox maintains conversation context across multiple turns within a session, enabling the model to reference prior messages and maintain coherent dialogue. Implementation details (context window size, session persistence, state management) are not documented. Appears to support continuous conversation without explicit context resets, but no information on how context is managed across calls or sessions.
Speech-native model maintains context across turns without intermediate text representation; context preservation is implicit in the model's audio processing, not a separate retrieval or memory system. Implementation details unknown.
Unknown — insufficient documentation on context management mechanisms to compare vs. alternatives like RAG-based systems or explicit context injection.
audio input/output handling with integrated text-to-speech
Medium confidenceHandles raw audio input (PCM, WAV, or streaming via WebSocket) and generates audio output via integrated text-to-speech (TTS) without requiring external TTS services. Audio encoding/decoding is abstracted by SDKs; developers work with audio streams or files without managing codec details. TTS is included in the per-minute pricing ($0.05/min), not a separate charge.
Integrated TTS bundled into per-minute pricing eliminates need for external TTS services; SDKs abstract audio codec handling, enabling developers to work with audio streams without codec expertise. TTS output is generated from the speech-native model's audio output, not from intermediate text.
Simpler than Twilio + external TTS (e.g., Google Cloud TTS) because TTS is native; more cost-effective than separate TTS services because it's bundled into per-minute pricing.
big bench audio task performance benchmarking
Medium confidenceUltravox v0.7 is benchmarked on Big Bench Audio, a standardized evaluation suite for speech AI models. Published results show 84% pass rate at fastest latency tier, positioning the model's accuracy/latency trade-off vs. competitors (GPT-4, Gemini Live, Claude Sonnet 4.5). Benchmarks are public and reproducible, enabling developers to evaluate suitability before committing.
Published Big Bench Audio benchmarks (84% pass rate) provide transparent, reproducible performance metrics; explicit latency/accuracy trade-off positioning enables developers to make informed model selection decisions.
More transparent than proprietary benchmarks because Big Bench Audio is public and reproducible; enables direct comparison with other voice AI models evaluated on the same suite.
webhook-based call event handling and asynchronous workflow integration
Medium confidenceUltravox exposes call lifecycle events (call initiated, call ended, transcription available, etc.) via webhooks, enabling asynchronous integration with external systems. Developers configure webhook URLs in the API; Ultravox sends HTTP POST requests with call metadata and events. This enables decoupled workflows where voice interactions trigger downstream processes (CRM updates, logging, notifications) without blocking the call.
Webhook-based event system enables decoupled integration with external systems; developers configure webhook URLs and receive call lifecycle events asynchronously without polling or blocking the call. Implementation details (event types, retry logic, payload format) not documented.
More scalable than polling-based integration because events are pushed to external systems; enables real-time downstream workflows without adding latency to voice interactions.
conversation transcript extraction and optional logging
Medium confidenceUltravox can optionally extract and return conversation transcripts (text representation of audio dialogue) via API responses or webhooks. Transcripts are generated from the speech-native model's internal representation (not via separate speech-to-text); transcript availability and format are not fully documented. Transcripts enable logging, compliance, and debugging without requiring separate transcription services.
Transcripts are extracted from the speech-native model's internal representation, not via separate speech-to-text service; this avoids transcription errors and latency from chained services. Transcript generation mechanism and accuracy not documented.
More accurate than separate speech-to-text services (e.g., Whisper) because transcripts come from the model's native audio understanding; no additional latency or cost for transcription.
Capabilities are decomposed by AI analysis. Each maps to specific user intents and improves with match feedback.
Related Artifactssharing capabilities
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Best For
- ✓developers building voice-first applications (customer service, telehealth, accessibility)
- ✓teams deploying conversational AI where naturalness and low latency are critical
- ✓non-technical founders prototyping voice-based MVPs without ML expertise
- ✓customer service teams building IVR replacements or voice support agents
- ✓healthcare providers deploying telehealth voice assistants
- ✓enterprises with existing phone infrastructure wanting to add AI agents
- ✓full-stack developers building voice features into existing web/mobile apps
- ✓mobile-first teams deploying voice AI on iOS or Android
Known Limitations
- ⚠Speech-native architecture means no intermediate text representation — limits debugging and transcript-based logging
- ⚠No documented support for multi-modal input (audio + text + images simultaneously)
- ⚠Concurrency limits vary by tier (free: 5 concurrent calls; pro/enterprise: higher but unspecified)
- ⚠No fine-tuning or custom model training documented — fixed ultravox-v0.7 model only
- ⚠Supported languages not documented; unclear if multilingual or English-only
- ⚠Specific telephony providers not documented — unclear which carriers are supported or if all major US/international carriers are included
Requirements
Input / Output
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Platform for building and deploying conversational AI agents that can integrate with external services, execute multi-step workflows, and maintain context across complex interactions using natural language.
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