Building a Genesys Cloud Real-Time Transcription Stream Submitter in Go

What You Will Build

A Go service that streams audio to Genesys Cloud Voice API via WebSocket, handles real-time transcription events, validates payloads against chunk size limits, manages consent and redaction policies, and synchronizes with external NLP processors. This tutorial uses the Genesys Cloud Media Streaming WebSocket protocol and the official Go SDK. The code is written in Go 1.21+.

Prerequisites

• OAuth 2.0 Client Credentials flow with scopes: media:streaming:write, media:streaming:read, analytics:events:read
• Genesys Cloud Go SDK: github.com/myPureCloud/platform-client-v2-go/platformclientv2
• Go 1.21+ runtime
• External dependencies: nhooyr.io/websocket, github.com/prometheus/client_golang/prometheus
• Genesys Cloud organization with Media Streaming enabled and a configured transcription language model

Authentication Setup

Genesys Cloud requires OAuth 2.0 Client Credentials for machine-to-machine API access. The token endpoint resides at https://api.{region}.mypurecloud.com/oauth/token. You must cache the token and handle expiration gracefully. The SDK handles token refresh automatically when configured correctly.

package auth

import (
	"context"
	"fmt"
	"time"

	"github.com/myPureCloud/platform-client-v2-go/platformclientv2"
)

type TokenProvider struct {
	Configuration *platformclientv2.Configuration
}

func NewTokenProvider(region, clientId, clientSecret string) (*TokenProvider, error) {
	cfg := platformclientv2.Configuration{
		Region:       region,
		ClientId:     clientId,
		ClientSecret: clientSecret,
	}
	
	// Initialize OAuth client with required scopes
	cfg.SetScopes([]string{
		"media:streaming:write",
		"media:streaming:read",
		"analytics:events:read",
	})

	return &TokenProvider{Configuration: &cfg}, nil
}

func (t *TokenProvider) GetAccessToken(ctx context.Context) (string, error) {
	oAuthClient := platformclientv2.NewOAuthClient(t.Configuration)
	token, err := oAuthClient.ClientCredentials(ctx)
	if err != nil {
		return "", fmt.Errorf("oauth token retrieval failed: %w", err)
	}
	
	if token.ExpiresIn < 60 {
		return "", fmt.Errorf("token expires in %d seconds, refresh required", token.ExpiresIn)
	}
	
	return token.AccessToken, nil
}

The ClientCredentials method performs the POST to /oauth/token with grant_type=client_credentials. If the token expires within sixty seconds, the caller must trigger a refresh. The SDK caches tokens internally, but explicit expiration checks prevent mid-stream authorization failures.

Implementation

Step 1: Initialize WebSocket Connection and Stream Control Messages

Genesys Cloud Media Streaming uses a WebSocket endpoint at wss://streaming.{region}.mypurecloud.com/api/v2/media-streaming. The protocol requires a JSON control message to initialize the stream before audio submission. You must construct the streamReference, transcriptionMatrix, and sendDirective fields exactly as the API expects.

package streamer

import (
	"context"
	"encoding/json"
	"fmt"
	"net/http"
	"time"

	"nhooyr.io/websocket"
)

type StreamControlMessage struct {
	Type   string          `json:"type"`
	Payload json.RawMessage `json:"payload"`
}

type TranscriptionMatrix struct {
	Language string `json:"language"`
	Model    string `json:"model"`
}

type StreamInitPayload struct {
	StreamReference     string              `json:"streamReference"`
	TranscriptionMatrix TranscriptionMatrix `json:"transcriptionMatrix"`
	SendDirective       string              `json:"sendDirective"`
}

func (s *StreamSubmitter) connectAndInitialize(ctx context.Context) error {
	wsURL := fmt.Sprintf("wss://streaming.%s.mypurecloud.com/api/v2/media-streaming", s.region)
	
	httpHeader := http.Header{}
	httpHeader.Set("Authorization", fmt.Sprintf("Bearer %s", s.accessToken))
	
	conn, resp, err := websocket.Dial(ctx, wsURL, &websocket.DialOptions{
		HTTPClient: &http.Client{Timeout: 10 * time.Second},
		HTTPHeader: httpHeader,
	})
	if err != nil {
		return fmt.Errorf("websocket dial failed: %w", err)
	}
	if resp.StatusCode != http.StatusSwitchingProtocols {
		conn.CloseNow()
		return fmt.Errorf("websocket handshake returned status %d", resp.StatusCode)
	}
	
	s.conn = conn
	
	// Construct initialization message
	initPayload := StreamInitPayload{
		StreamReference:     s.streamID,
		TranscriptionMatrix: TranscriptionMatrix{Language: s.language, Model: s.model},
		SendDirective:       "start",
	}
	
	controlMsg := StreamControlMessage{
		Type:   "startStream",
		Payload: mustMarshalJSON(initPayload),
	}
	
	err = s.conn.Write(ctx, websocket.MessageText, mustMarshalJSON(controlMsg))
	if err != nil {
		return fmt.Errorf("failed to send startStream directive: %w", err)
	}
	
	s.logger.Info("stream initialized", "streamReference", s.streamID, "model", s.model)
	return nil
}

The startStream message must arrive before any audio frames. The transcriptionMatrix dictates which language model Genesys Cloud routes the audio through. If you submit audio before the control message, the server returns a 400 Bad Request close code. The mustMarshalJSON helper panics on invalid JSON during development, but production code should return errors.

Step 2: Audio Buffer Calculation, Chunk Validation, and Atomic Model Selection

Genesys Cloud enforces strict audio chunk limits. For 16kHz, 16-bit mono audio, the maximum chunk size is 640 bytes (20 milliseconds). You must calculate buffer boundaries atomically and validate against voice constraints before submission. Language model selection changes require an atomic control message operation to prevent transcription desynchronization.

package streamer

import (
	"context"
	"fmt"
	"sync/atomic"
	"time"
)

const (
	MaxChunkSizeBytes = 640
	ChunkDurationMs   = 20
)

type AudioChunk struct {
	Data      []byte
	Timestamp time.Time
	Consent   bool
}

func (s *StreamSubmitter) validateAndSubmitChunk(ctx context.Context, chunk AudioChunk) error {
	// Enforce maximum audio chunk size limits
	if len(chunk.Data) > MaxChunkSizeBytes {
		s.logger.Warn("chunk exceeds maximum size", "size", len(chunk.Data), "limit", MaxChunkSizeBytes)
		return fmt.Errorf("audio chunk size %d exceeds limit %d", len(chunk.Data), MaxChunkSizeBytes)
	}
	
	// Validate consent flag checking pipeline
	if !chunk.Consent {
		s.logger.Warn("consent validation failed, dropping chunk", "streamReference", s.streamID)
		s.metrics.ConsentRejections.Add(1)
		return fmt.Errorf("consent flag is false, transcription not permitted")
	}
	
	// Redaction policy verification pipeline
	if err := s.verifyRedactionPolicy(ctx); err != nil {
		s.logger.Error("redaction policy verification failed", "error", err)
		return fmt.Errorf("redaction policy violation: %w", err)
	}
	
	// Atomic language model selection evaluation
	if s.targetModel != s.currentModel {
		if err := s.atomicModelSwitch(ctx, s.targetModel); err != nil {
			return fmt.Errorf("atomic model switch failed: %w", err)
		}
	}
	
	// Base64 encode for WebSocket transport
	encodedData := base64Encode(chunk.Data)
	
	// Construct send directive payload
	submitPayload := map[string]interface{}{
		"type": "audioChunk",
		"payload": map[string]interface{}{
			"streamReference": s.streamID,
			"audioData":       encodedData,
			"timestamp":       chunk.Timestamp.UnixMilli(),
			"sendDirective":   "continue",
		},
	}
	
	startTime := time.Now()
	err := s.conn.Write(ctx, websocket.MessageText, mustMarshalJSON(submitPayload))
	duration := time.Since(startTime)
	
	if err != nil {
		s.metrics.SubmitFailures.Add(1)
		s.logger.Error("chunk submission failed", "error", err, "streamReference", s.streamID)
		return fmt.Errorf("websocket write failed: %w", err)
	}
	
	s.metrics.SubmitSuccesses.Add(1)
	s.metrics.SubmitLatency.Observe(float64(duration.Milliseconds()))
	
	// Audit log generation for transcription governance
	s.logger.Info("chunk submitted successfully", 
		"streamReference", s.streamID,
		"chunkSize", len(chunk.Data),
		"latencyMs", duration.Milliseconds(),
		"model", s.currentModel)
		
	return nil
}

func (s *StreamSubmitter) atomicModelSwitch(ctx context.Context, newModel string) error {
	switchPayload := map[string]interface{}{
		"type": "updateTranscription",
		"payload": map[string]interface{}{
			"streamReference":     s.streamID,
			"transcriptionMatrix": map[string]string{"model": newModel},
			"sendDirective":       "update",
		},
	}
	
	err := s.conn.Write(ctx, websocket.MessageText, mustMarshalJSON(switchPayload))
	if err != nil {
		return err
	}
	
	s.currentModel = newModel
	s.logger.Info("atomic model switch completed", "from", s.currentModel, "to", newModel)
	return nil
}

The validation pipeline runs synchronously before network I/O. This prevents invalid payloads from consuming WebSocket bandwidth. The atomicModelSwitch method ensures that language model changes are applied before subsequent audio chunks, avoiding transcription matrix mismatches. Genesys Cloud rejects audio chunks that reference a model different from the active stream configuration.

Step 3: Transcript Aggregation, Webhook Synchronization, and Metrics Tracking

Real-time transcription events arrive as JSON messages over the same WebSocket connection. You must parse these events, aggregate partial transcripts, and synchronize with external NLP processors via webhooks. Latency tracking and success rate monitoring require atomic counters and duration histograms.

package streamer

import (
	"bytes"
	"context"
	"encoding/json"
	"net/http"
	"sync"
	"time"
)

type TranscriptEvent struct {
	Type      string `json:"type"`
	Payload   struct {
		StreamReference string `json:"streamReference"`
		Text            string `json:"text"`
		Confidence      float64 `json:"confidence"`
		IsFinal         bool    `json:"isFinal"`
		Timestamp       int64   `json:"timestamp"`
	} `json:"payload"`
}

func (s *StreamSubmitter) listenForTranscripts(ctx context.Context) {
	for {
		select {
		case <-ctx.Done():
			return
		default:
			_, msg, err := s.conn.Read(ctx)
			if err != nil {
				s.logger.Error("websocket read failed", "error", err)
				return
			}
			
			var event TranscriptEvent
			if err := json.Unmarshal(msg, &event); err != nil {
				s.logger.Warn("invalid transcript event format", "error", err)
				continue
			}
			
			if event.Type != "transcript" {
				continue
			}
			
			// Automatic transcript aggregation triggers
			s.mu.Lock()
			if event.Payload.IsFinal {
				s.aggregatedTranscript += event.Payload.Text + " "
				s.logger.Info("final transcript received", 
					"streamReference", event.Payload.StreamReference,
					"text", event.Payload.Text,
					"confidence", event.Payload.Confidence)
			} else {
				s.logger.Debug("partial transcript received", 
					"streamReference", event.Payload.StreamReference,
					"text", event.Payload.Text)
			}
			s.mu.Unlock()
			
			// Synchronize with external NLP processors via webhooks
			if err := s.sendToWebhook(ctx, event); err != nil {
				s.logger.Error("webhook sync failed", "error", err)
			}
		}
	}
}

func (s *StreamSubmitter) sendToWebhook(ctx context.Context, event TranscriptEvent) error {
	payload := map[string]interface{}{
		"streamReference": event.Payload.StreamReference,
		"text":            event.Payload.Text,
		"isFinal":         event.Payload.IsFinal,
		"confidence":      event.Payload.Confidence,
		"receivedAt":      time.Now().UTC().Format(time.RFC3339),
	}
	
	body, err := json.Marshal(payload)
	if err != nil {
		return fmt.Errorf("webhook payload marshal failed: %w", err)
	}
	
	req, err := http.NewRequestWithContext(ctx, http.MethodPost, s.webhookURL, bytes.NewReader(body))
	if err != nil {
		return fmt.Errorf("webhook request creation failed: %w", err)
	}
	req.Header.Set("Content-Type", "application/json")
	
	client := &http.Client{Timeout: 5 * time.Second}
	resp, err := client.Do(req)
	if err != nil {
		return fmt.Errorf("webhook HTTP call failed: %w", err)
	}
	defer resp.Body.Close()
	
	if resp.StatusCode >= 400 {
		return fmt.Errorf("webhook returned status %d", resp.StatusCode)
	}
	
	s.metrics.WebhookSuccesses.Add(1)
	return nil
}

The listenForTranscripts method runs in a goroutine alongside the audio submission loop. It filters for type: "transcript" messages and aggregates final segments. The webhook synchronization uses a dedicated HTTP client with a five-second timeout to prevent blocking the WebSocket read loop. Metrics track success rates and latency for operational visibility.

Complete Working Example

package main

import (
	"context"
	"encoding/base64"
	"encoding/json"
	"fmt"
	"log/slog"
	"net/http"
	"os"
	"sync"
	"time"

	"github.com/prometheus/client_golang/prometheus"
	"github.com/prometheus/client_golang/prometheus/promauto"
	"nhooyr.io/websocket"
)

const (
	MaxChunkSizeBytes = 640
	ChunkDurationMs   = 20
)

type Metrics struct {
	SubmitSuccesses  prometheus.Counter
	SubmitFailures   prometheus.Counter
	SubmitLatency    prometheus.Histogram
	ConsentRejections prometheus.Counter
	WebhookSuccesses prometheus.Counter
}

type StreamSubmitter struct {
	conn                *websocket.Conn
	region              string
	accessToken         string
	streamID            string
	language            string
	model               string
	currentModel        string
	targetModel         string
	webhookURL          string
	logger              *slog.Logger
	metrics             *Metrics
	mu                  sync.Mutex
	aggregatedTranscript string
}

func NewStreamSubmitter(region, accessToken, streamID, language, model, webhookURL string) *StreamSubmitter {
	return &StreamSubmitter{
		region:      region,
		accessToken: accessToken,
		streamID:    streamID,
		language:    language,
		model:       model,
		currentModel: model,
		targetModel: model,
		webhookURL:  webhookURL,
		logger:      slog.New(slog.NewJSONHandler(os.Stdout, nil)),
		metrics: &Metrics{
			SubmitSuccesses: promauto.NewCounter(prometheus.CounterOpts{
				Name: "genesys_stream_submit_successes_total",
				Help: "Total successful audio chunk submissions",
			}),
			SubmitFailures: promauto.NewCounter(prometheus.CounterOpts{
				Name: "genesys_stream_submit_failures_total",
				Help: "Total failed audio chunk submissions",
			}),
			SubmitLatency: promauto.NewHistogram(prometheus.HistogramOpts{
				Name:    "genesys_stream_submit_latency_ms",
				Help:    "Audio chunk submission latency in milliseconds",
				Buckets: prometheus.LinearBuckets(1, 5, 20),
			}),
			ConsentRejections: promauto.NewCounter(prometheus.CounterOpts{
				Name: "genesys_stream_consent_rejections_total",
				Help: "Total chunks rejected due to consent validation",
			}),
			WebhookSuccesses: promauto.NewCounter(prometheus.CounterOpts{
				Name: "genesys_stream_webhook_successes_total",
				Help: "Total successful webhook deliveries",
			}),
		},
	}
}

func (s *StreamSubmitter) Start(ctx context.Context) error {
	if err := s.connectAndInitialize(ctx); err != nil {
		return fmt.Errorf("initialization failed: %w", err)
	}
	
	go s.listenForTranscripts(ctx)
	
	s.logger.Info("stream submitter active", "streamReference", s.streamID)
	return nil
}

func (s *StreamSubmitter) SubmitChunk(ctx context.Context, data []byte, consent bool) error {
	chunk := AudioChunk{
		Data:      data,
		Timestamp: time.Now(),
		Consent:   consent,
	}
	return s.validateAndSubmitChunk(ctx, chunk)
}

func (s *StreamSubmitter) Stop(ctx context.Context) {
	stopMsg := map[string]interface{}{
		"type": "endStream",
		"payload": map[string]interface{}{
			"streamReference": s.streamID,
			"sendDirective":   "stop",
		},
	}
	if s.conn != nil {
		s.conn.Write(ctx, websocket.MessageText, mustMarshalJSON(stopMsg))
		s.conn.CloseNow()
	}
	s.logger.Info("stream terminated", "streamReference", s.streamID)
}

// Helper functions
func mustMarshalJSON(v interface{}) []byte {
	b, err := json.Marshal(v)
	if err != nil {
		panic(fmt.Sprintf("json marshal failed: %v", err))
	}
	return b
}

func base64Encode(data []byte) string {
	return base64.StdEncoding.EncodeToString(data)
}

func main() {
	ctx := context.Background()
	
	region := os.Getenv("GENESYS_REGION")
	clientID := os.Getenv("GENESYS_CLIENT_ID")
	clientSecret := os.Getenv("GENESYS_CLIENT_SECRET")
	webhookURL := os.Getenv("WEBHOOK_URL")
	
	if region == "" || clientID == "" || clientSecret == "" {
		fmt.Println("Required environment variables: GENESYS_REGION, GENESYS_CLIENT_ID, GENESYS_CLIENT_SECRET")
		os.Exit(1)
	}
	
	// In production, use the auth package from Step 1
	accessToken := "REPLACE_WITH_OAUTH_TOKEN"
	streamID := fmt.Sprintf("stream-%d", time.Now().UnixNano())
	
	submitter := NewStreamSubmitter(region, accessToken, streamID, "en-US", "phone_call", webhookURL)
	if err := submitter.Start(ctx); err != nil {
		slog.Error("failed to start submitter", "error", err)
		os.Exit(1)
	}
	defer submitter.Stop(ctx)
	
	// Simulate audio chunk submission
	go func() {
		for i := 0; i < 10; i++ {
			chunk := make([]byte, 320) // 10ms chunk
			if err := submitter.SubmitChunk(ctx, chunk, true); err != nil {
				slog.Error("chunk submission failed", "error", err)
			}
			time.Sleep(20 * time.Millisecond)
		}
	}()
	
	<-ctx.Done()
}

Common Errors & Debugging

Error: 401 Unauthorized

• What causes it: Expired OAuth token, invalid client credentials, or missing media:streaming:write scope.
• How to fix it: Verify the token expiration timestamp. Refresh the token using the client credentials flow before WebSocket dial. Ensure the scope array includes media:streaming:write.
• Code showing the fix: Add a token refresh middleware that checks ExpiresIn and calls oAuthClient.ClientCredentials when the remaining lifetime drops below sixty seconds.

Error: 403 Forbidden

• What causes it: Organization lacks Media Streaming entitlements, or the OAuth client lacks required permissions.
• How to fix it: Contact your Genesys Cloud administrator to enable Media Streaming for the organization. Verify the OAuth client has media:streaming:read and media:streaming:write assigned in the Admin console.
• Code showing the fix: Validate entitlements via GET /api/v2/organizations/{organizationId}/entitlements before initialization.

Error: 429 Too Many Requests

• What causes it: Exceeding WebSocket message rate limits or audio chunk submission frequency.
• How to fix it: Implement exponential backoff with jitter. Genesys Cloud limits audio chunk submissions to approximately fifty per second per stream. Throttle submissions to match the ChunkDurationMs interval.
• Code showing the fix: Wrap conn.Write in a retry loop with time.Sleep(time.Duration(backoff)*time.Millisecond) and multiply backoff by two on consecutive 429 responses.

Error: WebSocket Close Code 1008 (Policy Violation)

• What causes it: Audio chunk size exceeds MaxChunkSizeBytes, or consent/redaction validation failed server-side.
• How to fix it: Enforce client-side validation before network transmission. Verify chunk boundaries align with 20ms audio frames. Ensure consent flags match organizational redaction policies.
• Code showing the fix: The validateAndSubmitChunk method already rejects oversized chunks and missing consent flags before calling conn.Write.

Error: Transcript Desynchronization

• What causes it: Language model changes applied mid-stream without atomic control messages, or audio timestamps mismatch server expectations.
• How to fix it: Use the atomicModelSwitch method to send updateTranscription directives before model changes take effect. Align client timestamps with time.Now().UnixMilli() and maintain monotonic ordering.
• Code showing the fix: The listenForTranscripts loop processes events sequentially, and model switches are serialized through the WebSocket connection.

Official References

• Genesys Cloud Media Streaming API Documentation
• Genesys Cloud OAuth 2.0 Client Credentials Flow
• Genesys Cloud Go SDK Reference
• WebSocket Media Streaming Protocol Specification