Voice Agent Troubleshooting: Complete Diagnostic Checklist

Sumanyu Sharma
Sumanyu Sharma
Founder & CEO
, Voice AI QA Pioneer

Has stress-tested 4M+ voice agent calls to find where they break.

January 26, 202613 min read
Voice Agent Troubleshooting: Complete Diagnostic Checklist

TL;DR: Troubleshoot voice agent failures using this symptom-to-diagnosis approach:

SymptomLikely LayerFirst CheckProduction Threshold
Calls not connectingTelephonySIP registration, networkICE state: "connected"
No sound or garbled audioAudio/ASRCodec, WebRTC, VADPacket loss <1%, jitter <20ms
Wrong responses or timeoutsIntelligence/LLMLLM endpoint, promptsResponse <1s, no 429 errors
No agent speechOutput/TTSTTS service, audio encodingTTFB <200ms
Agent cuts off usersTurn DetectionVAD threshold, endpointingSilence threshold 400-600ms
High latency (>2s)Multiple layersComponent-level tracesP95 end-to-end <5s

Start at the infrastructure layer. Move up only when that layer is verified working. Most issues (50%+) are in telephony or audio—don't jump to LLM debugging first.

Methodology Note: Diagnostic frameworks and thresholds in this guide are derived from Hamming's analysis of 4M+ production voice agent calls and incident response patterns across 10K+ voice agents (2025-2026).

Related Guides:


If You're Debugging an AI Voice Agent...

AI voice agents add complexity layers on top of traditional VoIP infrastructure. Before debugging ASR, LLM, or TTS issues, verify your network and telephony stack is healthy. Most AI voice agent problems trace back to underlying VoIP issues.

VoIP SymptomAI Agent ImpactWhat Breaks
High jitter (>30ms)ASR receives corrupted audio framesTranscription errors, wrong words, gibberish
Packet loss (>1%)Audio gaps confuse speech recognitionMissed utterances, incomplete sentences
Poor MOS (<3.5)Degraded audio quality throughout pipelineASR confidence drops, user frustration
NAT/firewall issuesWebRTC ICE failures, one-way audioAgent can't hear user or vice versa
SIP registration failuresCalls don't connectComplete call failure before agent loads
Codec mismatchAudio format incompatibilityGarbled audio, no audio, echo

Bottom line: If your VoIP layer has problems, your AI pipeline will magnify them. A 2% packet loss that's "acceptable" for human calls causes 10-15% ASR word error rate increases for voice agents.

→ Skip to: VoIP Call Quality Checklist if you suspect network issues


Why Do Voice Agents Fail?

Voice agents fail across multiple interdependent layers: telephony, ASR, LLM orchestration, tool execution, and TTS. Single component failures cascade through subsequent decisions, making root cause diagnosis difficult. Systematic troubleshooting requires isolating whether issues stem from audio quality, semantic understanding, model failures, API integrations, or synthesis latency.

What you'll learn:

  • How to identify which component (ASR, LLM, tool execution, TTS) causes specific failure patterns
  • Diagnostic techniques using logs, traces, and component-level testing to isolate root causes
  • Production monitoring strategies to catch issues before they impact users

Quick filter: If you're restarting services before understanding which layer failed, you're wasting time.

What Are the Common Voice Agent Failure Categories?

Voice agents combine STT (speech-to-text), NLU (natural language understanding), decision logic, response generation, and TTS. Each layer depends on previous outputs: ASR errors corrupt LLM inputs, causing downstream tool execution failures.

Failure Category Reference Table

CategoryLayerSymptomsRoot CausesDiagnostic Priority
Retrieval failuresIntelligenceIrrelevant responses, wrong factsRAG returning wrong contextMedium
Instruction adherenceIntelligenceIgnoring guidelines, scope creepPrompt drift, temperature too highHigh
Reasoning failuresIntelligenceLogical errors, contradictionsContext overflow, model limitationsMedium
Tool integrationIntelligenceAPI errors, timeouts, wrong callsAuth failures, parameter issuesHigh
ASR failuresAudioEmpty transcripts, wrong wordsAccents, noise, phonetic ambiguityHigh
Latency bottlenecksMultipleAwkward pauses, interruptionsSlow APIs, model inference, synthesisHigh
Context lossIntelligenceForgetting earlier detailsToken limits, state managementMedium
Turn-taking errorsAudioCutting off users, not respondingVAD misconfiguration, endpointingHigh

How Do Failures Cascade Across Layers?

Single root-cause ASR errors propagate: incorrect transcription leads to misclassified intent, which triggers wrong tool selection. External service failures cascade when slow CRM responses delay agent replies beyond user tolerance (1-2 seconds).

Initial FailureCascade EffectUser Experience
Network latency (Telephony)ASR timeouts → LLM timeoutsCall drops, no response
ASR returning garbage (Audio)LLM hallucinating (Intelligence)Wrong actions, frustration
LLM slow (Intelligence)Turn-taking brokenUsers talk over agent
TTS slow (Output)User thinks agent diedPremature hangup

How Do You Troubleshoot ASR (Speech Recognition) Failures?

ASR Error Types and Patterns

Error TypeExampleRoot CauseDiagnostic Check
Accent variation"async" → "ask key"Regional pronunciationTest with accent datasets
Background noiseRandom word insertionsPoor microphone, artifactsCheck audio quality scores
Code-mixed speechMixed language confusionMultiple languagesEnable multilingual ASR
Low confidenceNames, numbers wrongCritical utterance issuesLog confidence scores
TruncationSentences cut offAggressive endpointingCheck silence threshold

ASR Diagnostic Checklist

  • Audio reaching server? Check for audio frames in logs, verify WebRTC connection
  • Codec negotiated correctly? Expected: Opus (WebRTC) or PCMU/PCMA (SIP)
  • ASR returning transcripts? Empty transcripts = no audio or VAD issue
  • Confidence scores acceptable? Target >0.85, investigate <0.7
  • WER within threshold? Target <5% clean audio, <10% with noise
  • Provider status? Check Deepgram, AssemblyAI, Google STT status pages

ASR-Specific Fixes

  • Incorporate diverse training data: accented audio, noisy environments, varied speech patterns from real production calls
  • Implement noise-canceling technologies: beamforming microphones, suppression algorithms, acoustic models trained on real-world audio
  • Apply LLM-guided refinement to ASR output: use language models to correct transcription errors using conversational context
  • Deploy hardware-accelerated VAD (voice activity detection) to filter background noise before ASR processing

For detailed ASR failure patterns, see Seven Voice Agent ASR Failure Modes in Production.

How Do You Debug LLM and Intent Recognition Failures?

LLM Failure Mode Reference

Failure ModeSymptomsRoot CauseFix
HallucinationsMade-up facts, wrong policiesNo grounding in verified dataAdd RAG validation, lower temperature
Misclassified intentWrong action triggeredAmbiguous user input, poor NLUImprove prompt, add disambiguation
Context overflowForgets earlier detailsToken limit exceededImplement summarization, truncation
Cascading errorsMultiple wrong decisionsSingle root mistake propagatesAdd validation checkpoints
Rate limitingSlow/no responses429 errors from providerImplement backoff, upgrade tier
Prompt driftInconsistent behaviorRecent prompt changesVersion control prompts, A/B test

LLM Diagnostic Checklist

  • LLM endpoint responding? Direct API test, check provider status
  • Rate limiting? Look for 429 errors, check tokens per minute
  • Prompt changes? Review recent deployments, check for injection
  • Context window? Calculate tokens per conversation, approaching limit?
  • Tool calls working? Check function call logs, tool timeout errors
  • Response quality? Compare to baseline, check for hallucinations

Mitigation Strategies

  • Ground with verified data: integrate agents with reliable, up-to-date databases (CRM, knowledge bases, APIs)
  • Implement prompt engineering: design prompts that constrain model outputs to factual, verified responses
  • Set appropriate model configurations: lower temperature (0.3-0.5) for factual tasks, restrict token generation length
  • Add validation checkpoints: verify critical information before executing irreversible actions

How Do You Fix Tool Execution and API Integration Failures?

Tool Call Failure Patterns

Failure TypeSymptomInvestigation StepsFix
Tool not recognizedAgent continues instead of actionCheck intent classification, tool definitionsImprove tool descriptions
Wrong tool selectionEmail API called instead of SMSReview tool descriptions, disambiguationAdd explicit tool routing
Parameter formattingTool rejects requestValidate data types, ranges, fieldsAdd parameter validation
Response misinterpretationIncorrect follow-up actionsCheck response parsing, schema validationFix response handling
TimeoutNo response from toolCheck API latency, timeout settingsIncrease timeout, add caching

Tool Integration Diagnostic Steps

  • Navigate to API Logs to monitor all requests/responses, check authentication errors, verify request payload structure
  • Check webhook logs to verify deliveries, server response codes, timing, monitor event delivery failures
  • Track tool execution results and errors through trace views showing input parameters and returned data
  • Test tool integrations independently before end-to-end testing: verify API calls work outside agent context
  • Measure API response latency to identify slow external services creating conversation pauses

Tool Execution Fixes

  • Implement fallback logic for when external services fail or respond slowly: retry with exponential backoff
  • Cache frequently used data to avoid unnecessary database lookups mid-conversation
  • Set timeout thresholds for external API calls (500-1000ms) to prevent indefinite waiting
  • Build circuit breakers to prevent small failures from cascading into system-wide problems

How Do You Optimize TTS Latency and Quality?

TTS Performance Benchmarks

MetricExcellentGoodAcceptablePoor
TTS TTFB<100ms<200ms<400ms>400ms
Full synthesis<150ms<300ms<500ms>500ms
Audio quality (MOS)>4.3>4.0>3.5<3.5

TTS Diagnostic Methods

  • Measure total latency including time-to-first-byte (TTFB) and complete audio synthesis duration
  • Track component-level breakdowns to isolate delays between STT, LLM inference, and TTS generation
  • Monitor tail latencies (p99) as users remember worst experiences, not average performance
  • Log synthesis quality metrics: audio artifacts, volume consistency, unnatural pauses in generated speech

Optimizing TTS Performance

  • Use dual streaming TTS: accepts text incrementally (token by token), begins speaking while LLM generates remaining response
  • Pre-connect and reuse SpeechSynthesizer to avoid connection latency on each request
  • Implement text streaming via websocket v2 endpoints for real-time synthesis as text arrives
  • Chunk long outputs at punctuation marks, stream incrementally to accelerate multi-sentence replies

For detailed latency optimization, see Voice AI Latency: What's Fast, What's Slow, and How to Fix It.

Why Are Customers Hanging Up on My Voice Bot?

Audio Quality Degradation Patterns

SymptomLikely CauseDiagnosticFix
Choppy audioPacket loss >5%Check webrtc-internals statsImprove network, enable FEC
Echo/feedbackAEC failureTest different device/browserEnable echo cancellation
One-way audioAsymmetric NAT/firewallCheck inbound/outbound packetsOpen UDP ports, use TURN
Robotic voiceHigh jitterCheck jitter buffer statsIncrease buffer, improve network
Audio cuts outNetwork instabilityMonitor packet loss patternsUse wired connection

Call Drop Root Causes

  • Insufficient internet speed for VoIP bandwidth requirements (minimum 100 kbps per call)
  • Network overload when multiple applications compete for bandwidth during voice calls
  • Weak or disrupted Wi-Fi signals cause packet loss, forcing call termination
  • Application conflicts when other apps request microphone access, breaking audio connection

Audio Quality Fixes

  • Upgrade internet connection to meet VoIP requirements: minimum 100 kbps upload/download per concurrent call
  • Use wired Ethernet connections for critical calls instead of Wi-Fi to reduce packet loss
  • Optimize Quality of Service (QoS) settings to prioritize voice traffic over other network activity
  • Implement jitter buffers to smooth packet arrival timing and reduce audio stuttering

VoIP Call Quality (Jitter/Packet Loss/MOS) Checklist

This section covers traditional VoIP diagnostics that directly impact AI voice agent performance. Fix these first before debugging ASR/LLM/TTS.

Network Quality Metrics Reference

MetricMeasurementGoodAcceptablePoorAI Agent Impact
Packet Loss% of lost RTP packets<0.5%<1%>2%ASR misses words, sentences cut off
JitterVariance in packet arrival (ms)<15ms<30ms>50msAudio distortion, robotic voice
Latency (RTT)Round-trip time (ms)<100ms<150ms>200msConversation delays, overlapping speech
MOS ScoreMean Opinion Score (1-5)>4.0>3.5<3.0User/agent audio quality degrades

VoIP Diagnostic Checklist

Network & Bandwidth:

  • Sufficient bandwidth? Minimum 100 kbps per concurrent call (G.711), 30 kbps (Opus)
  • QoS configured? Voice traffic prioritized (DSCP 46/EF marking)
  • Packet loss under threshold? Use ping -c 100 or VoIP quality tools
  • Jitter acceptable? Check with iperf3 or RTP stream analysis
  • No bandwidth contention? Other applications competing during calls

NAT & Firewall:

  • SIP ALG disabled? Router SIP ALG causes registration failures, one-way audio
  • UDP ports open? SIP: 5060/5061, RTP: 10000-20000 (varies by provider)
  • STUN/TURN configured? Required for WebRTC NAT traversal
  • Symmetric NAT handled? May require TURN relay server
  • Firewall allowing RTP? Stateful inspection may block return packets

SIP & Signaling:

  • SIP registration successful? Check for 401/403/408 errors
  • Correct SIP trunk credentials? Authentication failures = no calls
  • DNS SRV records resolving? SIP often uses SRV lookups
  • TLS/SRTP configured? Encryption may be required by provider
  • SIP timers appropriate? Session timers, registration refresh

Codec & Audio:

  • Codec negotiated correctly? Check SDP in SIP INVITE/200 OK
  • Codec priority set? Opus > G.722 > G.711 (for quality)
  • Sample rate matched? Mismatch causes audio distortion
  • Echo cancellation enabled? AEC required for full-duplex
  • Comfort noise configured? Prevents "dead air" during silence

Common VoIP Issues and Fixes

IssueSymptomsDiagnostic CommandFix
SIP ALG interferenceOne-way audio, registration dropsDisable in router settingsTurn off SIP ALG on all routers/firewalls
NAT traversal failureICE connection timeout, no audioCheck webrtc-internals ICE candidatesConfigure STUN/TURN, open UDP ports
Codec mismatchGarbled audio, no audioInspect SDP in SIP tracesForce compatible codec on both ends
RTP packet lossChoppy audio, words missingtcpdump -i eth0 udp port 10000-20000Enable FEC, increase jitter buffer
DNS resolutionIntermittent call failuresdig SRV _sip._udp.provider.comUse IP directly or fix DNS
TLS handshake failureSecure calls not connectingopenssl s_client -connect sip.provider.com:5061Update certificates, check TLS version

WebRTC-Specific Diagnostics

For browser-based voice agents using WebRTC:

chrome://webrtc-internals (Chrome)
about:webrtc (Firefox)

Key metrics to check:

  • ICE connection state: Should be "connected" or "completed"
  • DTLS state: Should be "connected"
  • Packets lost: Incoming/outgoing RTP packet loss
  • Jitter buffer: Current delay and target delay
  • Audio level: Verify audio is flowing (not 0)

RTP Stream Analysis

For deep packet inspection when standard tools don't reveal issues:

Capture RTP traffic:

tcpdump -i any -w voip_capture.pcap udp portrange 10000-20000

Analyze in Wireshark:

  1. Navigate to: Telephony → RTP → RTP Streams
  2. Check for packet loss percentage, jitter, delta (inter-packet timing)
  3. Look for sequence number gaps indicating lost packets

Key RTP metrics:

MetricWhere to FindHealthy Value
Lost packetsRTP stream analysis<0.5%
Max jitterRTP stream analysis<30ms
Mean jitterRTP stream analysis<15ms
Sequence errorsRTP stream analysis0

MOS Score Interpretation

Mean Opinion Score (MOS) predicts perceived call quality:

MOS ScoreQualityUser ExperienceTypical Cause
4.3-5.0ExcellentToll quality, no perceptible issuesGood network, proper codec
4.0-4.3GoodMinor impairments, still clearSlight jitter, minimal loss
3.5-4.0FairNoticeable issues, still usableModerate packet loss
3.0-3.5PoorAnnoying, hard to understandHigh jitter, significant loss
<3.0BadUnusable, call should be terminatedSevere network issues

For AI voice agents: Target MOS >4.0. Below 3.5, ASR accuracy drops significantly.

What Logging and Tracing Do You Need for Voice Agent Debugging?

Essential Logging Schema

Turn-level data (per exchange):

{
  "call_id": "call_abc123",
  "turn_index": 3,
  "timestamp": "2026-01-26",
  "user_transcript": "I need to reschedule my appointment",
  "asr_confidence": 0.94,
  "intent": {"name": "reschedule_appointment", "confidence": 0.91},
  "latency_ms": {"stt": 180, "llm": 420, "tts": 150, "total": 750},
  "tool_calls": [{"name": "get_appointments", "success": true, "latency_ms": 85}],
  "agent_response": "I can help you reschedule..."
}

Production Monitoring Essentials

MetricWhat It MeasuresAlert Threshold
Call success rateCalls completing without errorsAlert if <95%
P95 end-to-end latencyWorst-case response timeAlert if >5s
ASR confidenceTranscription qualityAlert if avg <0.8
Task completionGoal achievement rateAlert if <85%
Error rateFailed calls/total callsAlert if >0.2%

Tracing Voice Agent Workflows

Tracing captures every call step: audio input, ASR output, semantic interpretation, internal prompts, model generations, tool calls, TTS output. Use OpenTelemetry for metrics, logs, traces to keep data portable across observability tools.

For detailed observability implementation, see Voice Agent Observability: The Missing Discipline.

How Do You Fix Conversation Flow and Turn-Taking Issues?

Context Loss and Memory Issues

Agents hit context window limits (4k-32k tokens), causing "forgetting" of important earlier conversation details. As conversations grow, critical information gets pushed out, leading to contradictions or lost problem tracking.

IssueSymptomFix
Token overflowForgets early detailsImplement conversation summarization
State lossAsks same question twicePersist state externally
Context driftContradicts earlier statementsAdd context anchoring prompts

How Do You Prevent Agents from Interrupting Users?

  • Implement dynamic silence thresholds: 300ms for quick exchanges, 800ms for slower speakers
  • Use hardware-accelerated Voice Activity Detection (VAD) that handles interruptions gracefully
  • Move beyond Voice Activity Detection to consider semantics, context, tone, conversational cues
  • Tune VAD sensitivity based on use case: customer service needs longer thresholds than quick commands

Fixing Conversation Flow Issues

  • Use hybrid context management: full server-side history for high-stakes sessions, lightweight vector summaries for general chat
  • Implement explicit context anchoring: have users restate critical constraints every 3-4 turns
  • Test conversation state management: verify handling of interruptions, corrections, topic changes
  • Implement conversation summarization at regular intervals to maintain context within token limits

How Do You Build Error Handling and Recovery Patterns?

Resilience Design Patterns

PatternImplementationWhen to Use
Circuit breakerStop calling failed serviceExternal API failures
Exponential backoffRetry with increasing delaysTransient network issues
Graceful degradationFall back to simpler responsesKnowledge retrieval failures
Timeout limitsMax 500-1000ms for tool callsSlow external services
Retry limitsMax 3-5 attemptsBefore escalating to human

User-Facing Error Recovery

  • Provide clear, actionable feedback: "I'm having trouble accessing our product database. Let me try a different approach" instead of "Error 500"
  • Build fallback logic into customer journeys: "Press 0 to speak to a live representative" when agent reaches capability limits
  • Acknowledge errors transparently: "I missed that, could you repeat?" rather than guessing at misheard inputs

Continuous Improvement from Failures

  • Feed production failures back into offline evaluation datasets to create continuous improvement loops
  • Convert any live conversation into replayable test case with caller audio, ASR text, expected intent
  • When production call fails, convert to regression test with one click, preserving original audio and timing
  • Track failure resolution rates: measure time from issue identification to deployed fix

What Testing and Evaluation Strategies Work for Voice Agents?

Automated Testing Approaches

  • Auto-generate test cases from agent prompts and documentation to ensure coverage
  • Run 1000+ concurrent calls with real-world conditions: accents, background noise, interruptions, edge cases
  • Test agents in multiple languages, simulate global accents and real-world noise environments
  • Implement synthetic user simulation: generate varied conversation paths to stress-test agent logic

Evaluation Metrics That Matter

Metric CategoryKey MetricsTarget Threshold
ConversationalLatency, interruptions, turn-takingP95 <800ms response time
OutcomesTask completion, escalation rate>85% completion
QualityWER, intent accuracy, entity extraction<5% error rate
CompliancePII handling, script adherence100% compliance

CI/CD Integration for Voice Agents

  • Integrate testing into GitHub Actions, Jenkins, or CI/CD pipeline to trigger tests and block bad prompts automatically
  • After each build, send predefined prompts to agent; if more than 5% responses differ from baseline, deployment halts
  • Version control agent configurations (prompts, tools, models) alongside code for reproducible deployments

For comprehensive testing methodology, see How to Evaluate and Test Voice Agents.


Summary and Next Steps

Systematic troubleshooting requires component-level isolation: test ASR, LLM, tool execution, TTS independently before end-to-end diagnosis. Production monitoring with comprehensive logging, tracing, and observability catches issues before they impact users.

Next steps:

  • Implement structured logging capturing every component's inputs, outputs, latency, confidence scores
  • Set up production monitoring with alerts for latency spikes, error rate increases, quality degradation
  • Build automated testing pipelines that run diverse scenarios before deployment to catch failures early

How Hamming Helps with Voice Agent Troubleshooting

Hamming provides the observability and testing layer that makes troubleshooting faster:

  • 4-Layer Visibility: Unified dashboards showing health across Telephony, Audio, Intelligence, and Output
  • Instant Root Cause: One-click from alert to transcript, audio, and model logs
  • Session Replay: Full audio playback with transcripts and component traces
  • Regression Detection: Automated alerts when metrics deviate from baseline
  • Scenario Generation: Auto-generate test cases from prompts, execute in <10 minutes

Instead of manually debugging across multiple dashboards, get automated visibility into every layer of your voice agent stack.

Debug your voice agents with Hamming →

Related Guides:

Frequently Asked Questions

Start by measuring network quality metrics: packet loss should be under 1%, jitter under 30ms, and MOS score above 3.5. Check that SIP ALG is disabled on your router (a common cause of one-way audio and registration failures). Verify NAT traversal is working by configuring STUN/TURN servers for WebRTC. Open required UDP ports for SIP (5060/5061) and RTP (10000-20000). Use tcpdump or Wireshark to analyze RTP streams for packet loss and sequence errors.

Jitter (variance in packet arrival timing) and packet loss are typically caused by network congestion, insufficient bandwidth, Wi-Fi interference, or misconfigured QoS settings. For voice agents, target jitter under 30ms and packet loss under 1%. Enable QoS with DSCP 46/EF marking to prioritize voice traffic. Use wired Ethernet instead of Wi-Fi for critical deployments. Implement jitter buffers to smooth packet arrival, and enable Forward Error Correction (FEC) on your codec to recover from minor packet loss.

SIP registration failures manifest as calls not connecting at all. Check for 401 (Unauthorized), 403 (Forbidden), or 408 (Request Timeout) errors in your SIP logs. Verify your SIP trunk credentials are correct. Ensure DNS SRV records resolve properly using 'dig SRV _sip._udp.provider.com'. Disable SIP ALG on your router which often corrupts SIP messages. For TLS connections, verify certificates are valid and TLS versions match. Check that SIP timers and registration refresh intervals are configured appropriately.

One-way audio (you can hear the user but they can't hear the agent, or vice versa) is almost always a NAT/firewall issue. The most common cause is SIP ALG being enabled on routers—disable it immediately. Check that UDP ports for RTP (typically 10000-20000) are open inbound and outbound. For WebRTC, verify STUN/TURN servers are configured and ICE candidates are being exchanged. Symmetric NAT requires a TURN relay server. Use chrome://webrtc-internals to verify ICE connection state reaches 'connected'.

For AI voice agents, target a MOS (Mean Opinion Score) of 4.0 or higher. MOS above 4.3 is excellent toll-quality audio. Between 3.5-4.0 is acceptable but users will notice minor impairments. Below 3.5, ASR accuracy degrades significantly because speech recognition models struggle with distorted audio. A MOS below 3.0 means calls are essentially unusable. Monitor MOS continuously in production and alert when it drops below 3.8 to catch issues before they impact ASR performance.

Use component-level tracing to capture every step and drill into individual spans to isolate STT, intent classification, or response generation issues. Get component-level breakdowns showing latency for STT, LLM inference, and TTS synthesis to pinpoint delays. Test each component in isolation: evaluate STT accuracy on recorded audio, test intent classification on transcribed text, and verify response logic with mocked inputs.

Context window limits cause agents to forget earlier conversation details as exchanges grow longer. Tool integration failures where external APIs respond slowly or fail completely can block agent progress. Network or hardware issues, applications requesting microphone access, and unhandled exceptions in agent code can also halt execution without proper error recovery.

Train ASR systems using diverse datasets including a wide range of accents, dialects, and speech patterns from your target user base. Incorporate accent detection mechanisms to identify and adjust recognition models for different accents. Allow users to specify their accent/dialect during onboarding, and apply LLM-guided refinement to ASR output using conversational context to correct phonetic errors.

Human-normal response time falls between 300 milliseconds and 1,200 milliseconds. Users expect responses within 1-2 seconds; longer delays feel broken and destroy engagement. Pauses longer than 800 milliseconds start feeling unnatural, and anything over 1.5 seconds breaks conversational flow. Target p99 latency under 2 seconds since users remember worst experiences, not average performance.

Implement dynamic silence thresholds: 300ms for quick exchanges, 800ms for users who speak more slowly. Use hardware-accelerated Voice Activity Detection (VAD) that handles interruptions gracefully. Measure false-positive interruptions closely as being cut off drives user frustration. Move beyond Voice Activity Detection to consider semantics, context, tone, and conversational cues.

Monitor error rates (should be within 0.2%), overall success rates, latency percentiles, and containment rates through dashboards. Track token usage spikes indicating infinite loops in agentic reasoning, sentiment drift, and feedback scores. Monitor turn-level latency at every exchange, interruption count, and talk-to-listen ratio for conversation balance. Set up alerts for latency spikes over 500ms, tone anomalies, and quality drops below acceptable limits.

Navigate to API Logs to monitor all requests and responses, check for authentication errors, and verify payload structure. Check webhook logs to verify deliveries to your server, response codes, and timing. Use CLI to forward webhooks to local development server for real-time debugging. Capture every tool call with complete input/output, token usage, cost, and timing information. Test tool integrations independently before end-to-end testing to isolate API issues from agent logic.

Convert any live conversation into a replayable test case with caller audio, ASR text, and expected intent in one click. When a production call fails, convert it to a regression test preserving original audio, timing, and caller behavior. Capture full traces including audio attachments alongside transcriptions and responses. Implement versioning for agent configurations so replays use the exact same prompts, models, and tools as the original call.

Sumanyu Sharma

Sumanyu Sharma

Founder & CEO

Previously Head of Data at Citizen, where he helped quadruple the user base. As Senior Staff Data Scientist at Tesla, grew AI-powered sales program to 100s of millions in revenue per year.

Researched AI-powered medical image search at the University of Waterloo, where he graduated with Engineering honors on dean's list.

“At Hamming, we're taking all of our learnings from Tesla and Citizen to build the future of trustworthy, safe and reliable voice AI agents.”