AI Speech Processing With Open-Source ASR for Voice Networks

We help service providers design and deploy production-grade AI speech processing pipelines – built on open-source automatic speech recognition (ASR) models and telephony infrastructure.

Built for Carrier-Grade Voice Infrastructure

The AI Automatic Speech Recognition Partner That Understands Telecom

We understand both ends of adding AI speech processing to voice networks, so you get a partner who knows how to make the carrier infrastructure and the ASR system work together.

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ASR Pipeline Design

We evaluate open-weight ASR models against your actual audio and workload needs, then we build the architecture to take it into production.
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SIP/SIPREC Integration

Most AI vendors don't extract audio from live telephone networks. We implement SIPREC-based media forking without hurting call quality.

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Transcription Architecture

Some applications need low latency, others need accuracy. We build both – streaming architectures for real-time transcription and two-pass POST-call pipelines.

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Speaker Diarization

We integrate diarization models that account for failure modes like overlapping speech and produce speaker-labeled, timestamped output.
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VAD & Audio Pre-Processing

We configure voice activity detection to prevent hallucinated transcriptions from silence and background noise, and convert the codec into PCM audio format.

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Privacy-First Deployment

For HIPAA, GDPR, or data residency requirements, we deploy the full stack inside your data center to ensure no audio leaves your infrastructure.
AI ASR That Works on Real Telephony Audio

Your AI Speech Processing Problems, Solved

We bring AI automatic speech recognition into production environments that cloud-only AI vendors simply cannot reach.

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ASR Inaccuracies

An ASR system benchmarked on clean audio won't be accurate for real-life telephony audio. We evaluate against your call data to choose the right fit.

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Unused Transcription Output

Capturing audio doesn't mean producing value. We build the LLM layer that generates summaries, extracts action items, and pushes data into your CRM.
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Contact Center Intelligence

Adding sentiment analysis requires ASR, diarization, speaker labeling, and LLM inference. We design these pipelines with the telephony integration layer.

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Compliance Barriers

HIPAA, GDPR, and attorney-client privilege require on-premises deployment for audio privacy. We design for data sovereignty from the start.

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Integration Complexity

SIPREC implementations vary across vendors and complicate AI ASR integration. We know where those issues are and how to fix them.

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Changing Landscape

The ASR AI landscape is moving fast. You need the latest models and vCon compatibility.

OUR CLIENTS

Trusted by Industry Leaders

Join other organizations that enjoy expert engineering support with ECG.

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WHY CHOOSE ECG

Your AI Automatic Speech Recognition Experts

We've spent years inside carrier-grade VoIP platforms, so we understand the telecom layer that most AI vendors skip.

ECG has been deep inside carrier-grade voice infrastructure for decades. We built the first hosted PBX platform for Coca-Cola in collaboration with BellSouth, integrated with Verizon and AT&T networks, and trained engineers across the service provider industry. We bring that background when we build AI automatic speech recognition into voice networks, resulting in systems that actually work in production – not just in demos.

Success Stories From Our Clients

ECG is definitely the right team for our network!

Nicole Rodriguez

AVP Switching and Wireless Data Engineering | AT&T Mobility

ECG's broad scope of clients means they know what's happening before we do. We stay competitive with ECG as our guide.

Mark Hayes

VP of Voice Engineering | Momentum Telecom

ECG has really cool technology!

Jeff Pulver

Voice over IP Pioneer

ECG delivers exceptional quality and service via their software products and consulting services. Speaking as someone with direct large scale enterprise delivery with their team, my personal experience has been universally positive.

Joe Pfiefer

Assistant Director | U.S. Department of Justice

I'm happy to say I've partnered with ECG at a number of service providers. You guys have been an outstanding engineering and operations partner for my teams.

Tom Faherty

VP | Databank

ECG is a reliable partner.

Edwin Martirosyan

COO | BluIP

GET STARTED WITH ECG TODAY

Book Your 30-Minute Connect Call

Get in touch with ECG for products and services that support your crucial voice infrastructure needs. 

Experience the ECG Advantage

Whether you’re a service provider, enterprise, or government agency, your voice infrastructure is in good hands with ECG.

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Proven Expertise

Our team has decades of proven experience building and supporting voice networks.

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Powerful Partnerships

Our strategic alliances are designed to help deliver customer-centric, total solutions to our clients.

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Elevated Network Design

We draw from experience with dozens of service providers to create straightforward, manageable designs.

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Comprehensive Support

Our team will assist in your technical projects, support your goals, automate processes, and train your team.

How We Help

Expert Support for Automatic Speech Recognition AI

From design to troubleshooting to optimization, we help service providers deploy automatic speech recognition that works on real telephone networks.

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Deploying AI Automatic Speech Recognition

Standing up a production AI speech processing pipeline on top of real telephone infrastructure requires making architectural decisions early. At ECG, our engineers help you design the full stack from day one.

  • We map your existing environment to the right audio capture method, configure SIPREC, and validate media forking without affecting call quality.
  • We help you select the right open-weights model based on your audio characteristics, language requirements, latency budget, and compute constraints.
  • We design the LLM extraction layer, structured output format, and integrations that push intelligence to the systems your team already uses.
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ASR System Troubleshooting and Support

When an ASR system produces bad transcripts, missing speakers, hallucinated text, or drops audio, the root cause is rarely where it first appears. We dig into the full stack to identify and fix the real issue. Trust us to:

  • Capture and analyze RTP streams to identify problems that degrade ASR input before the model ever sees it.
  • Analyze SIPREC interoperability problems and normalize handling so the recording server processes calls correctly.
  • Improve ASR accuracy by checking transcripts against ground truth audio, identifying gaps, and implementing corrections.
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ASR Voice and Speech Processing Optimization

A working transcription pipeline is a foundation, not a finished product. Real value comes from integrations and intelligence layers built on top. We help you expand from basic transcription to a system that drives business outcomes. We will:

  • Push structured call intelligence into CRM platforms like Salesforce, Zendesk, and HubSpot with LLM prompting.
  • Implement the vCon Virtual Conversation standard as your internal data structure and prepare for interoperability.
  • Add streaming on top of POST-call baseline: live closed captioning, real-time keyword detection, and consent detection for recording.
Frequently Asked Questions

Common AI Speech Processing Questions, Answered

Get quick answers to questions about automatic speech recognition, ASR artificial intelligence, and voice processing.

ASR is the process of converting spoken audio into text using AI models. In VoIP telephony, ASR sits downstream of audio capture – typically SIPREC-based media forking from a PBX or SBC – and converts RTP audio streams into transcripts.

Telephony ASR is harder than transcribing a podcast: the audio is often narrowband at 8kHz, subject to packet loss, may carry multiple speakers, and includes background noise and codec artifacts. ECG accounts for these obstacles and builds AI ASR systems that actually work in real-world telephony environments.

ASR and AI automated speech recognition are essentially the same thing. The term AI ASR or AI automated speech recognition emphasizes that the system uses artificial intelligence and deep learning models rather than older rule-based or template-matching approaches.

We use the Hugging Face Open ASR Leaderboard to identify the leading open source options. As of 2026, we use NVIDIA Parakeet for English telephony efficiency, Whisper Large V3 Turbo for multilingual support across 100 languages, IBM Granite Speech 3.3 for high-accuracy regulated industries, and Moonshine for low-latency streaming.

Choosing the right ASR model will depend on your workload. We assess your needs and evaluate against your actual audio before committing to an architecture. 



vCon (Virtual Conversation) is an emerging IETF Internet standard for encapsulating conversation artifacts, including audio recordings, transcripts, metadata, and analysis, in a structured, interoperable format.

vCon enables interoperability between different call recording and AI processing platforms. We implement vCon as the internal data structure so your platform can ingest recordings from third-party compliance recording systems without custom integration for each vendor.

Accuracy depends heavily on audio quality. Noisy contact center audio, strong accents, or industry-specific terminology can drop accuracy, which is why it's important to evaluate ASR models against your actual call data before deployment.

Yes. The primary method for accomplishing this is SIPREC (SIP Recording), which is an IETF standard protocol that forks a copy of RTP audio from a call to a recording server.

ECG's platform acts as the SIPREC endpoint. It accepts the SIPREC INVITE from the SBC, negotiates media, and receives a one-way audio stream. The media fork happens at the SBC before reaching any other system, so call quality for the actual participants is unaffected regardless of what happens in the transcription pipeline.

Vision and speech processing in artificial intelligence refers to multimodal AI systems that combine computer vision (understanding images and video) with speech processing (understanding audio).

Together, vision and speech processing enable applications like video conferencing with automatic captions, or analyzing recorded videos with both visual and audio understanding.

Both image and speech processing rely on deep learning models trained on large datasets. For speech, the models learn to recognize patterns in audio waveforms and convert them to text. For images, models learn to recognize visual patterns and extract meaning. What enables them is access to training data, computing power for training and inference, and open source frameworks like PyTorch and TensorFlow.

In voice applications, speech processing is the focus. Multimodal systems that combine vision and speech exist, but telephony solutions typically only require speech understanding.

There are four big ones:

  1. Benchmark accuracy doesn't match real-world performance. Models tested on clean audio perform worse on telephony's narrowband codec and background noise.
  2. Production deployment takes about 3x the effort of initial prototyping once you add streaming, VAD, diarization, and system integrations.
  3. Most SaaS transcription routes audio to third-party clouds, which breaks HIPAA, GDPR, and data governance requirements. You need an on-premises deployment for regulated data.
  4. SIPREC vendor implementations vary from the standard, so you have to normalize XML handling across different SBCs.
At ECG, we design for all of these from day one.

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Get in touch for products that support your crucial voice infrastructure needs.