Bridging Genesys Cloud Voice API SIP Trunks to a Custom PBX via a Java WebSocket-to-SIP Proxy

What You Will Build

• A Java middleware service that accepts WebSocket signaling from a custom PBX and translates it to SIP/SDP traffic toward a Genesys Cloud SIP trunk.
• The proxy uses the Genesys Cloud Java SDK to validate trunk configurations and enforces strict codec negotiation and SDP re-INVITE handling during active media sessions.
• The implementation covers OAuth2 client credentials authentication, SIP stack initialization, WebSocket routing, and full SDP offer/answer exchange logic.

Prerequisites

• Genesys Cloud OAuth 2.0 Client Credentials grant with voice:trunks:read, voice:trunks:write, and analytics:conversations:read scopes
• Genesys Cloud Java SDK PureCloudPlatformClientV2 version 140.0.0 or higher
• Java 17 LTS runtime with module path configured for javax.sip
• External dependencies: org.java-websocket:Java-WebSocket:1.5.4, com.google.code.gson:gson:2.10.1, javax.sip:jain-sip-ri:1.3.0-91

Authentication Setup

Genesys Cloud Voice API endpoints require a bearer token obtained via the OAuth 2.0 client credentials flow. The Java SDK handles token caching and automatic refresh, but you must initialize the client with explicit scope validation to prevent 403 Forbidden responses during trunk configuration reads.

import com.mypurecloud.platform.client.ApiClient;
import com.mypurecloud.platform.client.auth.OAuth;
import com.mypurecloud.platform.client.auth.OAuthFlow;
import com.mypurecloud.platform.client.auth.Scopes;
import com.mypurecloud.platform.client.PureCloudPlatformClientV2;
import java.util.Set;

public class GenesysAuth {
    public static PureCloudPlatformClientV2 initializeClient(String clientId, String clientSecret) throws Exception {
        ApiClient apiClient = new ApiClient();
        apiClient.setBasePath("https://api.mypurecloud.com");
        
        OAuth oAuth = apiClient.getOAuth();
        oAuth.setClientId(clientId);
        oAuth.setClientSecret(clientSecret);
        oAuth.setGrantType(OAuthFlow.CLIENT_CREDENTIALS);
        
        // Explicitly declare required scopes to fail fast on misconfigured credentials
        Set<String> requiredScopes = Set.of(
            Scopes.VOICE_TRUNKS_READ,
            Scopes.VOICE_TRUNKS_WRITE,
            Scopes.ANALYTICS_CONVERSATIONS_READ
        );
        oAuth.setScopes(requiredScopes);
        
        // SDK caches tokens and handles refresh automatically
        PureCloudPlatformClientV2 client = new PureCloudPlatformClientV2(apiClient);
        return client;
    }
}

The SDK intercepts HTTP 401 responses and triggers a silent token refresh before retrying the original request. You do not need to implement manual refresh logic for standard REST calls.

Implementation

Step 1: Fetch Trunk Configuration and Validate Codec Preferences

Before establishing any SIP signaling, the proxy must retrieve the target Genesys Cloud trunk configuration to determine supported codecs and encryption requirements. The endpoint /api/v2/voice/trunks/{trunkId} returns the trunk object containing media settings.

import com.mypurecloud.platform.client.PureCloudPlatformClientV2;
import com.mypurecloud.platform.client.api.VoiceApi;
import com.mypurecloud.platform.client.model.TrunkConfiguration;
import com.mypurecloud.platform.client.model.TrunkMediaConfiguration;
import java.util.List;
import java.util.Map;
import java.util.stream.Collectors;

public class TrunkConfigLoader {
    private final PureCloudPlatformClientV2 client;
    private final String trunkId;

    public TrunkConfigLoader(PureCloudPlatformClientV2 client, String trunkId) {
        this.client = client;
        this.trunkId = trunkId;
    }

    public Map<String, Boolean> loadAllowedCodecs() throws Exception {
        VoiceApi voiceApi = new VoiceApi(client);
        TrunkConfiguration trunk = voiceApi.getVoiceTrunk(trunkId);
        TrunkMediaConfiguration media = trunk.getMedia();
        
        if (media == null || media.getCodecPreferences() == null) {
            throw new IllegalStateException("Trunk " + trunkId + " has no media configuration defined.");
        }

        // Extract codec names and mark them as allowed for negotiation
        return media.getCodecPreferences().stream()
            .collect(Collectors.toMap(
                codec -> codec.getName().toUpperCase(),
                codec -> true
            ));
    }
}

Raw HTTP equivalent for debugging:

GET /api/v2/voice/trunks/{trunkId} HTTP/1.1
Host: api.mypurecloud.com
Authorization: Bearer <access_token>
Accept: application/json

HTTP/1.1 200 OK
Content-Type: application/json
{
  "id": "a1b2c3d4-e5f6-7890-abcd-ef1234567890",
  "name": "Primary SIP Trunk",
  "media": {
    "codecPreferences": [
      { "name": "G711U", "priority": 1 },
      { "name": "G711A", "priority": 2 },
      { "name": "G729", "priority": 3 }
    ]
  }
}

Step 2: Initialize SIP Stack and WebSocket Server

The proxy requires a SIP User Agent (UA) stack to communicate with Genesys Cloud and a WebSocket server to accept commands from your custom PBX. Jain-SIP provides the standard Java SIP API. You must configure the SIP stack with the correct transport (TCP/TLS), local address, and message handler.

import javax.sip.*;
import javax.sip.address.*;
import javax.sip.header.*;
import javax.sip.message.*;
import org.java_websocket.server.WebSocketServer;
import org.java_websocket.handshake.ClientHandshake;
import java.net.InetSocketAddress;
import java.util.Map;
import java.util.concurrent.ConcurrentHashMap;

public class SipWebSocketProxy extends WebSocketServer {
    private final SipStack sipStack;
    private final SipProvider sipProvider;
    private final Map<String, Boolean> allowedCodecs;
    private final ConcurrentHashMap<String, String> callMap; // WS session ID -> Call-ID

    public SipWebSocketProxy(int wsPort, SipStack sipStack, SipProvider sipProvider, Map<String, Boolean> allowedCodecs) {
        super(new InetSocketAddress(wsPort));
        this.sipStack = sipStack;
        this.sipProvider = sipProvider;
        this.allowedCodecs = allowedCodecs;
        this.callMap = new ConcurrentHashMap<>();
    }

    @Override
    public void onOpen(ClientHandshake handshake) {
        System.out.println("PBX WebSocket connected.");
    }

    @Override
    public void onMessage(WebSocket conn, String message) {
        try {
            handlePbxCommand(conn, message);
        } catch (Exception e) {
            conn.send("{\"error\": \"" + e.getMessage() + "\"}");
        }
    }

    @Override
    public void onClose(WebSocket conn, int code, String reason, boolean remote) {
        System.out.println("PBX WebSocket disconnected: " + reason);
    }

    @Override
    public void onError(WebSocket conn, Exception ex) {
        ex.printStackTrace();
    }

    private void handlePbxCommand(WebSocket conn, String jsonPayload) throws Exception {
        // Parse incoming PBX JSON command
        com.google.gson.JsonObject payload = com.google.gson.JsonParser.parseString(jsonPayload).getAsJsonObject();
        String action = payload.get("action").getAsString();
        
        if ("INVITE".equals(action)) {
            String targetUri = payload.get("target").getAsString();
            String sdpOffer = payload.get("sdp").getAsString();
            createSipInvite(conn, targetUri, sdpOffer);
        }
    }

    private void createSipInvite(WebSocket conn, String targetUri, String sdpOffer) throws Exception {
        Request invite = sipProvider.createRequest(targetUri);
        invite.setRequestLine(new RequestLine("INVITE", targetUri));
        
        // Attach SDP body
        invite.setContent(new SipContent(new byte[0], sdpOffer, "application/sdp"));
        
        // Set standard headers
        invite.setHeader(new CallIdHeader(sipStack.getNewCallId()));
        invite.setHeader(new CSeqHeader(1, "INVITE"));
        invite.setHeader(new FromHeader(new Address(new NameAddress("proxy@local")), null, null));
        invite.setHeader(new ToHeader(new Address(new NameAddress(targetUri))));
        invite.setHeader(new MaxForwardsHeader(70));
        invite.setHeader(new ContactHeader(new Address(new NameAddress("proxy@local"))));
        
        // Send and store mapping
        String callId = ((CallIdHeader) invite.getHeader(CallIdHeader.NAME)).getCallId();
        callMap.put(conn.getRemoteSocketAddress().toString(), callId);
        sipProvider.sendRequest(invite);
    }
}

Step 3: Handle SDP Negotiation and Re-INVITE Signaling

Genesys Cloud sends a 183 Session Progress or 200 OK containing an SDP answer. Your proxy must validate the answer against the allowed codecs, then forward the selected codec to the PBX via WebSocket. Mid-call codec changes or media renegotiations arrive as SIP re-INVITEs. You must parse the SDP, verify codec compatibility, and relay the updated media parameters.

import javax.sip.message.Response;
import javax.sip.header.ContactHeader;
import java.util.regex.Matcher;
import java.util.regex.Pattern;

public class SdpNegotiator implements javax.sip.message.MessageListener {
    private final SipWebSocketProxy proxy;
    private final Map<String, Boolean> allowedCodecs;

    public SdpNegotiator(SipWebSocketProxy proxy, Map<String, Boolean> allowedCodecs) {
        this.proxy = proxy;
        this.allowedCodecs = allowedCodecs;
    }

    @Override
    public void processRequest(RequestEvent requestEvent) {
        Request request = requestEvent.getRequest();
        if ("INVITE".equals(request.getMethod()) && requestEvent.isReInvite()) {
            handleReInvite(request);
        }
    }

    @Override
    public void processResponse(ResponseEvent responseEvent) {
        Response response = responseEvent.getResponse();
        if (response.getStatusCode() == 183 || response.getStatusCode() == 200) {
            String sdpAnswer = extractSdp(response);
            String negotiatedCodec = validateSdpCodecs(sdpAnswer);
            
            if (negotiatedCodec != null) {
                // Forward negotiated result to PBX via WebSocket
                String wsMessage = String.format(
                    "{\"action\": \"SDP_ANSWER\", \"callId\": \"%s\", \"codec\": \"%s\", \"sdp\": \"%s\"}",
                    getCallId(response), negotiatedCodec, escapeJson(sdpAnswer)
                );
                proxy.getWebSocketSessions().forEach(ws -> ws.send(wsMessage));
            } else {
                // Send SIP 488 Not Acceptable Here if codec mismatch
                try {
                    javax.sip.message.Response reject = responseEvent.getClientTransaction()
                        .createResponse(488, "Not Acceptable Here");
                    responseEvent.getClientTransaction().sendResponse(reject);
                } catch (Exception e) {
                    e.printStackTrace();
                }
            }
        }
    }

    private String validateSdpCodecs(String sdp) {
        Pattern payloadPattern = Pattern.compile("a=rtpmap:(\\d+) (\\w+)/\\d+");
        Matcher matcher = payloadPattern.matcher(sdp);
        while (matcher.find()) {
            String codecName = matcher.group(2).toUpperCase();
            if (allowedCodecs.containsKey(codecName)) {
                return codecName;
            }
        }
        return null;
    }

    private void handleReInvite(Request request) {
        String sdpOffer = extractSdp(request);
        String selectedCodec = validateSdpCodecs(sdpOffer);
        
        if (selectedCodec == null) {
            try {
                Response reject = request.createResponse(488, "Codec mismatch on re-INVITE");
                ((ServerTransaction) requestEvent.getServerTransaction()).sendResponse(reject);
            } catch (Exception ignored) {}
            return;
        }

        // Construct SDP answer with negotiated codec and send back
        String sdpAnswer = generateSdpAnswer(selectedCodec);
        try {
            Response ok = request.createResponse(200, "OK");
            ok.setContent(new javax.sip.message.SipContent(new byte[0], sdpAnswer, "application/sdp"));
            ((ServerTransaction) requestEvent.getServerTransaction()).sendResponse(ok);
        } catch (Exception e) {
            e.printStackTrace();
        }
    }

    private String extractSdp(javax.sip.message.Message msg) {
        try {
            return new String(msg.getContent().getData());
        } catch (Exception e) {
            return "";
        }
    }

    private String getCallId(Response response) {
        return ((javax.sip.header.CallIdHeader) response.getHeader("Call-ID")).getCallId();
    }

    private String escapeJson(String input) {
        return input.replace("\\", "\\\\").replace("\"", "\\\"").replace("\n", "\\n");
    }

    private String generateSdpAnswer(String codec) {
        // Simplified SDP answer generation for demonstration
        return "v=0\r\no=- 0 0 IN IP4 0.0.0.0\r\ns=Media Session\r\nc=IN IP4 0.0.0.0\r\nt=0 0\r\nm=audio 9 RTP/AVP " + getPayloadType(codec) + "\r\na=rtpmap:" + getPayloadType(codec) + " " + codec + "/8000\r\n";
    }

    private int getPayloadType(String codec) {
        return switch (codec) {
            case "G711U" -> 0;
            case "G711A" -> 8;
            case "G729" -> 18;
            default -> 96;
        };
    }
}

Complete Working Example

The following module combines authentication, configuration loading, SIP stack initialization, and the WebSocket server into a single executable class. Replace the placeholder credentials and trunk ID before running.

import com.mypurecloud.platform.client.PureCloudPlatformClientV2;
import org.java_websocket.WebSocket;
import javax.sip.*;
import javax.sip.address.SipURI;
import javax.sip.header.CallIdHeader;
import javax.sip.header.FromHeader;
import javax.sip.header.ToHeader;
import javax.sip.header.MaxForwardsHeader;
import javax.sip.header.CSeqHeader;
import javax.sip.header.ContactHeader;
import javax.sip.header.Address;
import javax.sip.header.NameAddress;
import javax.sip.header.RequestLine;
import javax.sip.message.Request;
import javax.sip.message.Response;
import javax.sip.message.SipContent;
import java.net.InetAddress;
import java.util.Map;
import java.util.Properties;

public class WebSocketToSipProxy {
    public static void main(String[] args) throws Exception {
        // 1. Authentication
        PureCloudPlatformClientV2 client = GenesysAuth.initializeClient("YOUR_CLIENT_ID", "YOUR_CLIENT_SECRET");
        
        // 2. Load Trunk Configuration
        String trunkId = "YOUR_TRUNK_ID";
        TrunkConfigLoader configLoader = new TrunkConfigLoader(client, trunkId);
        Map<String, Boolean> allowedCodecs = configLoader.loadAllowedCodecs();
        
        // 3. Initialize SIP Stack
        Properties sipProps = new Properties();
        sipProps.setProperty("gov.nist.javax.sip.TRACE_LEVEL", "DEBUG");
        sipProps.setProperty("gov.nist.javax.sip.SERVER_MODE", "true");
        sipProps.setProperty("javax.sip.STACK_NAME", "GenesysBridge");
        sipProps.setProperty("javax.sip.IP_ADDRESS", InetAddress.getLocalHost().getHostAddress());
        
        SipFactory sipFactory = SipFactory.getInstance();
        sipFactory.initialize(sipProps);
        SipStack sipStack = sipFactory.createSipStack(sipProps);
        
        ListeningPoint lp = sipStack.createListeningPoint(
            InetAddress.getLocalHost().getHostAddress(),
            5060,
            "tcp"
        );
        
        SipProvider sipProvider = sipStack.createSipProvider(lp);
        
        // 4. Initialize WebSocket Server and Proxy
        SipWebSocketProxy proxy = new SipWebSocketProxy(8080, sipStack, sipProvider, allowedCodecs);
        SdpNegotiator negotiator = new SdpNegotiator(proxy, allowedCodecs);
        sipProvider.addMessageListener(negotiator, new RequestLine("INVITE", null));
        
        proxy.start();
        System.out.println("Proxy running. WS listening on 8080, SIP listening on 5060/TCP");
    }
}

Common Errors & Debugging

Error: HTTP 401 Unauthorized or 403 Forbidden on SDK Calls

• Cause: The OAuth client credentials lack the required voice:trunks:read scope, or the token expired without SDK refresh.
• Fix: Verify the OAuth client in the Genesys Cloud Admin console has the exact scopes listed in Prerequisites. Ensure PureCloudPlatformClientV2 is instantiated with OAuthFlow.CLIENT_CREDENTIALS. The SDK automatically retries 401s with a fresh token. If the issue persists, clear the local token cache by restarting the JVM.
• Code showing the fix:

// Force token refresh if SDK cache is stale
client.getApiClient().getOAuth().clearAccessToken();
client.getApiClient().getOAuth().setScopes(Set.of(Scopes.VOICE_TRUNKS_READ));

Error: SIP 488 Not Acceptable Here

• Cause: The custom PBX offered a codec that Genesys Cloud does not support, or the SDP payload type mapping is incorrect.
• Fix: Cross-reference the a=rtpmap lines in the PBX SDP offer against the codecPreferences array returned by /api/v2/voice/trunks/{trunkId}. Update the validateSdpCodecs method to reject unsupported codecs before sending the INVITE.
• Code showing the fix:

if (!allowedCodecs.containsKey(codecName)) {
    System.out.println("Rejecting unsupported codec: " + codecName);
    continue;
}

Error: HTTP 429 Too Many Requests

• Cause: Rapid trunk configuration polling or concurrent SDK calls exceeded Genesys Cloud rate limits.
• Fix: Implement exponential backoff for SDK calls that return 429. The Genesys Cloud Java SDK does not include automatic 429 retry, so you must wrap API calls in a retry loop.
• Code showing the fix:

public <T> T fetchWithRetry(java.util.function.Supplier<T> apiCall) throws Exception {
    int retries = 0;
    while (retries < 3) {
        try {
            return apiCall.get();
        } catch (com.mypurecloud.platform.client.ApiException e) {
            if (e.getCode() == 429 && retries < 2) {
                long delay = (long) Math.pow(2, retries) * 1000;
                Thread.sleep(delay);
                retries++;
            } else {
                throw e;
            }
        }
    }
    throw new Exception("Max retries exceeded for 429 response");
}

Error: WebSocket Close Code 1006 (Abnormal Closure)

• Cause: The custom PBX dropped the connection during SDP negotiation, or the SIP stack failed to route the response back to the correct WebSocket session.
• Fix: Maintain a strict session-to-call mapping using ConcurrentHashMap. Implement WebSocket ping/pong keep-alives to detect stale connections before sending SDP answers.
• Code showing the fix:

proxy.setConnectionLostTimeout(30); // Force cleanup after 30s inactivity

Official References

• Genesys Cloud Voice Trunks API Documentation
• Genesys Cloud OAuth 2.0 Client Credentials Flow
• Jain-SIP Java API Specification
• RFC 3261: SIP Session Initiation Protocol
• RFC 4566: SDP Session Description Protocol