Streaming Genesys Cloud Real-time Media API Audio Frames with Python

What You Will Build

• This tutorial builds a production-grade Python WebSocket client that streams PCMU audio frames to a Genesys Cloud Real-time Media stream.
• The implementation uses the Genesys Cloud Real-time Media API WebSocket endpoint with explicit RTP sequencing, jitter buffer validation, and congestion control.
• The code is written in Python 3.9+ and includes async WebSocket management, external webhook synchronization, latency tracking, and structured audit logging.

Prerequisites

• OAuth 2.0 Client Credentials grant with scopes: realtime:stream:send realtime:media:send
• Genesys Cloud Real-time Media API v2
• Python 3.9 or higher
• External dependencies: pip install websockets aiohttp requests pydantic
• A valid Genesys Cloud streamId from an active Real-time Media session
• Network access to api.mypurecloud.com on port 443

Authentication Setup

The Real-time Media API requires an OAuth 2.0 access token attached to the WebSocket handshake. The following code fetches the token, caches it, and implements a simple refresh trigger when expiration approaches.

import requests
import time
import os
from typing import Optional

GENESYS_DOMAIN = os.getenv("GENESYS_DOMAIN", "api.mypurecloud.com")
OAUTH_URL = f"https://{GENESYS_DOMAIN}/oauth/token"

class OAuthTokenManager:
    def __init__(self, client_id: str, client_secret: str):
        self.client_id = client_id
        self.client_secret = client_secret
        self.token: Optional[str] = None
        self.expires_at: float = 0.0
        self.scopes = "realtime:stream:send realtime:media:send"

    def get_token(self) -> str:
        if self.token and time.time() < self.expires_at - 60:
            return self.token
        
        payload = {
            "grant_type": "client_credentials",
            "client_id": self.client_id,
            "client_secret": self.client_secret,
            "scope": self.scopes
        }
        
        response = requests.post(OAUTH_URL, data=payload, timeout=10)
        response.raise_for_status()
        
        data = response.json()
        self.token = data["access_token"]
        self.expires_at = time.time() + data["expires_in"]
        return self.token

The token manager checks expiration before each connection attempt. The - 60 buffer prevents handshake failures during token rollover. The requests library handles HTTP 401/403 responses via raise_for_status(), which you must catch in the calling layer.

Implementation

Step 1: WebSocket Connection and Stream Initialization

The Real-time Media API uses a persistent WebSocket connection. You must establish the connection, send a control message to initialize the audio track, and verify the codec matrix before streaming frames.

import asyncio
import websockets
import json
import logging
from pydantic import BaseModel, ValidationError

logging.basicConfig(level=logging.INFO, format="%(asctime)s | %(levelname)s | %(message)s")
logger = logging.getLogger("genesys_media_streamer")

class StreamControlMessage(BaseModel):
    type: str
    action: str
    codec: str
    sample_rate: int

class AudioFramePayload(BaseModel):
    type: str
    frameId: str
    timestamp: int
    payload: str
    codec: str
    sequence: int

class GenesysAudioStreamer:
    def __init__(self, stream_id: str, token_manager: OAuthTokenManager, webhook_url: str):
        self.stream_id = stream_id
        self.token_manager = token_manager
        self.webhook_url = webhook_url
        self.ws_url = f"wss://{GENESYS_DOMAIN}/api/v2/realtime/media/streams/{stream_id}/audio"
        self.ws: Optional[websockets.WebSocketClientProtocol] = None
        self.sequence_number = 1
        self.rtp_timestamp = 0
        self.sample_rate = 8000
        self.frame_duration_ms = 20
        self.max_jitter_buffer_ms = 150
        self.congestion_active = False
        self.success_count = 0
        self.failure_count = 0
        self.last_frame_time = time.time()

    async def connect_and_init(self) -> None:
        token = self.token_manager.get_token()
        ws_url = f"{self.ws_url}?access_token={token}"
        
        try:
            self.ws = await websockets.connect(ws_url, ping_interval=20, ping_timeout=10)
            logger.info("WebSocket connected successfully")
            
            init_msg = StreamControlMessage(
                type="control",
                action="start",
                codec="PCMU",
                sample_rate=self.sample_rate
            ).model_dump()
            
            await self.ws.send(json.dumps(init_msg))
            response = await asyncio.wait_for(self.ws.recv(), timeout=5.0)
            logger.info("Stream initialization response: %s", response)
        except websockets.exceptions.InvalidStatusCode as e:
            logger.error("WebSocket handshake failed: %s", e)
            raise
        except asyncio.TimeoutError:
            logger.error("Stream initialization timed out")
            raise

The StreamControlMessage uses Pydantic to enforce schema validation before transmission. The action: "start" directive tells Genesys Cloud to allocate the audio track. The response typically contains a confirmation object with the assigned track ID. If the server returns a 403 or 401, the WebSocket handshake fails with InvalidStatusCode.

Step 2: RTP Packet Construction and Jitter Buffer Validation

Real-time audio requires strict timestamp synchronization and sequence tracking. The following method constructs frames, validates them against network constraints, and enforces jitter buffer limits.

import base64
import uuid

    def construct_audio_frame(self, raw_audio_bytes: bytes) -> dict:
        frame_id = str(uuid.uuid4())
        self.rtp_timestamp += (self.sample_rate * self.frame_duration_ms) // 1000
        
        payload_b64 = base64.b64encode(raw_audio_bytes).decode("utf-8")
        
        frame = AudioFramePayload(
            type="audio",
            frameId=frame_id,
            timestamp=self.rtp_timestamp,
            payload=payload_b64,
            codec="PCMU",
            sequence=self.sequence_number
        ).model_dump()
        
        self.sequence_number = (self.sequence_number + 1) % 65536
        return frame

    def validate_frame_latency(self, current_time: float) -> bool:
        elapsed_ms = (current_time - self.last_frame_time) * 1000
        if elapsed_ms > self.max_jitter_buffer_ms:
            logger.warning("Jitter buffer limit exceeded: %.2fms", elapsed_ms)
            return False
        return True

The RTP timestamp increments by sample_rate * frame_duration / 1000. For 8000Hz PCMU with 20ms frames, the increment is 160 samples. The sequence number wraps at 65536 to match standard RTP behavior. The validate_frame_latency method checks the time delta between consecutive frames. If the delta exceeds max_jitter_buffer_ms, the frame is rejected to prevent buffer bloat and downstream decoding artifacts.

Step 3: Atomic Frame Transmission and Congestion Control

WebSocket sends must be atomic to avoid fragmentation. The following method handles format verification, congestion control triggers, and success/failure tracking.

    async def send_frame_atomic(self, frame_data: dict) -> bool:
        if self.congestion_active:
            logger.info("Congestion control active. Dropping frame.")
            self.failure_count += 1
            return False
            
        try:
            payload_str = json.dumps(frame_data)
            # Format verification: ensure payload is valid JSON before send
            json.loads(payload_str)
            
            await self.ws.send(payload_str)
            self.success_count += 1
            return True
        except websockets.exceptions.ConnectionClosed as e:
            logger.error("WebSocket closed during send: code=%s, reason=%s", e.code, e.reason)
            self.failure_count += 1
            return False
        except json.JSONDecodeError:
            logger.error("Payload format verification failed")
            self.failure_count += 1
            return False

The send_frame_atomic method serializes the frame, verifies the JSON structure, and performs a single await self.ws.send() call. This guarantees atomic delivery at the transport layer. If the connection drops, the method catches ConnectionClosed and increments the failure counter. Congestion control is triggered when the success rate drops below a threshold or when jitter validation fails repeatedly.

Step 4: External Webhook Synchronization and Audit Logging

Stream events must synchronize with external media servers. The following method POSTs frame metadata to a webhook endpoint, tracks latency, and writes structured audit logs.

import aiohttp
from datetime import datetime, timezone

    async def sync_webhook_and_log(self, frame_id: str, success: bool, latency_ms: float) -> None:
        audit_entry = {
            "timestamp": datetime.now(timezone.utc).isoformat(),
            "stream_id": self.stream_id,
            "frame_id": frame_id,
            "success": success,
            "latency_ms": round(latency_ms, 2),
            "success_rate": self._calculate_success_rate(),
            "congestion_active": self.congestion_active,
            "sequence": self.sequence_number - 1
        }
        
        logger.info("AUDIT | %s", json.dumps(audit_entry))
        
        webhook_payload = {
            "event": "frame_streamed",
            "stream_id": self.stream_id,
            "frame_id": frame_id,
            "status": "delivered" if success else "dropped",
            "latency_ms": latency_ms
        }
        
        try:
            async with aiohttp.ClientSession() as session:
                async with session.post(
                    self.webhook_url,
                    json=webhook_payload,
                    timeout=aiohttp.ClientTimeout(total=3)
                ) as resp:
                    if resp.status in (200, 202):
                        logger.debug("Webhook synced successfully for frame %s", frame_id)
                    else:
                        logger.warning("Webhook sync failed with status %s", resp.status)
        except Exception as e:
            logger.error("Webhook sync exception: %s", str(e))

    def _calculate_success_rate(self) -> float:
        total = self.success_count + self.failure_count
        if total == 0:
            return 0.0
        return (self.success_count / total) * 100.0

The audit log records every frame attempt with UTC timestamps, latency, and success rates. The webhook POST uses aiohttp for non-blocking delivery. The method catches network timeouts and logs them without halting the primary stream loop. This separation ensures media transport continues even if the external sync endpoint experiences latency.

Complete Working Example

The following script combines all components into a runnable streamer. It simulates PCMU audio generation, runs the validation pipeline, and manages the connection lifecycle.

import asyncio
import os
import random

async def run_streamer():
    client_id = os.getenv("GENESYS_CLIENT_ID")
    client_secret = os.getenv("GENESYS_CLIENT_SECRET")
    stream_id = os.getenv("GENESYS_STREAM_ID")
    webhook_url = os.getenv("EXTERNAL_WEBHOOK_URL", "https://example.com/webhooks/media")
    
    if not all([client_id, client_secret, stream_id]):
        raise ValueError("Missing required environment variables")
        
    token_mgr = OAuthTokenManager(client_id, client_secret)
    streamer = GenesysAudioStreamer(stream_id, token_mgr, webhook_url)
    
    await streamer.connect_and_init()
    
    try:
        while True:
            # Simulate 20ms PCMU frame (160 bytes)
            raw_audio = bytes([random.randint(0, 255) for _ in range(160)])
            
            frame = streamer.construct_audio_frame(raw_audio)
            current_time = time.time()
            
            # Latency and jitter validation
            if not streamer.validate_frame_latency(current_time):
                streamer.congestion_active = True
                await asyncio.sleep(0.05)  # Backpressure
                continue
                
            send_start = time.time()
            success = await streamer.send_frame_atomic(frame)
            latency = (time.time() - send_start) * 1000
            
            await streamer.sync_webhook_and_log(frame["frameId"], success, latency)
            streamer.last_frame_time = current_time
            
            # Congestion control trigger based on success rate
            if streamer._calculate_success_rate() < 85.0:
                streamer.congestion_active = True
                logger.warning("Success rate dropped below 85%%. Activating congestion control.")
                await asyncio.sleep(0.1)
            else:
                streamer.congestion_active = False
                
            # Maintain 50 FPS (20ms per frame)
            await asyncio.sleep(0.02)
            
    except KeyboardInterrupt:
        logger.info("Streamer stopped by user")
    finally:
        if streamer.ws:
            await streamer.ws.close()
            logger.info("WebSocket closed")

if __name__ == "__main__":
    asyncio.run(run_streamer())

The loop generates 160-byte PCMU frames, validates jitter, sends atomically, logs metrics, and applies backpressure when success rates fall below 85 percent. The asyncio.sleep(0.02) call maintains the 50 frames per second cadence required for 20ms audio blocks.

Common Errors & Debugging

Error: WebSocket 401 Unauthorized

• Cause: OAuth token expired, missing realtime:stream:send scope, or incorrect client credentials.
• Fix: Verify the token manager refreshes tokens before handshake. Check the scope parameter in the OAuth payload. Ensure the client ID has Real-time Media permissions in the Genesys Cloud admin console.
• Code Adjustment: Add explicit scope validation in OAuthTokenManager and log the exact error response body.

Error: WebSocket 403 Forbidden

• Cause: The streamId does not belong to the authenticated client, or the stream state is not active.
• Fix: Confirm the stream was created via the Real-time Media API with the same OAuth client. Verify the stream status is active before connecting.
• Code Adjustment: Poll /api/v2/realtime/media/streams/{streamId} before WebSocket connection to verify state.

Error: ConnectionClosed (code=1006)

• Cause: Network instability, NAT timeout, or Genesys Cloud scaling event dropping idle connections.
• Fix: Implement exponential backoff reconnection logic. Keep the connection alive with periodic ping frames.
• Code Adjustment: Wrap connect_and_init in a retry loop with asyncio.sleep(2 ** attempt) and reset RTP state on reconnection.

Error: Jitter Buffer Exceeded / High Latency

• Cause: Frame generation slower than 50 FPS, GC pauses, or network congestion.
• Fix: Reduce frame size, increase thread priority, or trigger congestion control earlier.
• Code Adjustment: Lower max_jitter_buffer_ms to 100ms and increase backpressure sleep duration when triggered.

Error: JSONDecodeError on Payload

• Cause: Base64 encoding corruption or invalid characters in frameId.
• Fix: Use base64.urlsafe_b64encode if URL transmission is required. Validate uuid4() output format.
• Code Adjustment: Add frame_data["payload"] = frame_data["payload"].replace("\n", "") before serialization.

Official References

• Genesys Cloud Real-time Media API Documentation
• Real-time Media WebSocket Specification
• OAuth 2.0 Client Credentials Flow
• Python websockets Library Reference
• Pydantic V2 Validation Guide