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title: SIP Integration with Genesys Cloud
subtitle: null
slug: sip-with-genesys-cloud
description: null
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SIP trunking is only available on the Synthflow Enterprise Plan.
This guide covers connecting Genesys Cloud CX to Synthflow via a BYOC Cloud SIP trunk for both inbound and outbound calling. Genesys will route calls to a Synthflow AI agent, which can also transfer calls back to live agents or initiate outbound calls through Genesys.
Genesys Cloud has two integration paths with different levels of access. The standard BYOC trunk approach documented here provides basic SIP connectivity but may have limited throughput. For full integration (higher concurrency, priority routing), Synthflow must be listed in the Genesys AppFoundry — contact your Genesys account team and Synthflow sales to discuss availability.
## Prerequisites
| Requirement | Detail |
| ------------- | -------------------------------------------------------------------------------------- |
| Genesys Cloud | Active CX org with Admin access and BYOC Cloud enabled |
| Synthflow | Enterprise Plan with a configured AI assistant (v2.0+) |
| Phone Number | At least one DID provisioned (E.164 format) |
| Network | Firewall allows SIP (UDP/TCP 32681 or TLS 32682) and RTP between Genesys and Synthflow |
## Genesys Cloud Configuration
All configuration is done in the Genesys Cloud Admin console under **Admin** → **Telephony** → **Trunks** → **External Trunks**.
### Create the External Trunk
1. Navigate to **External Trunks** and click **Create New**.
2. Enter an External Trunk Name (e.g. `Synthflow AI`).
3. Under **Trunk Type**, select **BYOC Carrier**.
4. Under **Subtype**, select **Generic BYOC Carrier**.
5. Set **Trunk State** to **In-Service**.
6. Select the transport **Protocol** (UDP or TLS).
### Configure Inbound (Calls from Synthflow → Genesys)
Inbound SIP routing lets Genesys identify which organization incoming calls belong to.
1. Under **Inbound**, enter a unique **Inbound SIP Termination Identifier** (e.g. `synthflow`).
2. This generates your org-specific FQDN (e.g. `synthflow.byoc.usw2.pure.cloud`). Copy this — you will need it for Synthflow configuration.
3. Select the **Number Plan Site** that contains the number plans for your DIDs.
The generated FQDN is the address Synthflow will use as the Termination URI to send calls into Genesys. The region portion (usw2, use1, euw1, etc.) depends on your Genesys Cloud deployment region.
### Configure Outbound (Calls from Genesys → Synthflow)
Outbound settings tell Genesys where to send calls destined for Synthflow.
1. Under **Outbound**, enter `sip.synthflow.ai` in the **Outbound SIP Termination FQDN** field.
2. Under **SIP Servers or Proxies**, enter `sip.synthflow.ai` as the hostname and `32681` as the port. Click **Add**.
3. Enable **Digest Authentication**.
4. Enter the **Realm**: `sip.synthflow.ai`
5. Enter the **User Name** and **Password** (these are your Synthflow SIP credentials — see Synthflow Configuration below).
### Configure Caller ID
1. Under **Caller ID**, enter the Caller Address (your DID in E.164 format) and Caller Name.
2. Set the **Prioritized Caller Selection** order as needed for your use case.
### Configure SIP Access Control
1. Under **SIP Access Control**, add the Synthflow SIP signaling IP addresses to the Allow list. Contact Synthflow support for the current IP list.
2. Click **Save External Trunk**.
Do not enable "Allow All" in SIP Access Control. Always whitelist specific IPs to prevent unauthorized access.
### Assign Trunk to a Site
1. Navigate to **Telephony** → **Sites** and select your site.
2. Under **Outbound Routes**, add the **Synthflow AI** trunk to the appropriate route.
3. Ensure the site has the correct **Number Plans** assigned for the DIDs you want to route.
For inbound call routing, use Genesys Architect to build a call flow that directs incoming calls to a queue, IVR, or transfer destination. You can route specific DIDs or IVR branches to the Synthflow trunk for AI handling.
## Synthflow Configuration
### Add the SIP Number
1. Go to the Synthflow **Phone Numbers** page.
2. Click **Add Number** → **Custom / SIP**.
3. Enter the phone number in E.164 format (same DID configured in Genesys).
4. Fill in the SIP connection details:
| Field | Value |
| ---------------------------- | ---------------------------------------------------------------------- |
| Termination URI | Your Genesys BYOC inbound FQDN (e.g. `synthflow.byoc.usw2.pure.cloud`) |
| SIP Username | Your SIP authentication username |
| SIP Password | Your SIP authentication password |
| Outbound Proxy (if required) | Leave blank unless Genesys requires a specific proxy |
The Termination URI is the Genesys FQDN generated from the Inbound SIP Termination Identifier. This is what Synthflow uses to route outbound calls back into Genesys.
Be precise when copying SIP credentials. Incorrect values will cause registration failure.
### Attach to the Assistant
1. Go to **Assistants** and create or select an assistant (v2.0+).
2. In the assistant settings, attach the SIP number you just created.
3. Wait 1–2 minutes for SIP registration to complete.
The SIP trunk is created at the moment of attaching the SIP number to the assistant.
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## Call Flow Summary
| Scenario | Flow |
| ---------------------------- | -------------------------------------------------------------------------------- |
| Inbound to AI | Caller → PSTN → Genesys Cloud → Architect flow → BYOC trunk → Synthflow AI Agent |
| AI transfers to live agent | Synthflow AI → SIP transfer → Genesys BYOC trunk → Queue → Live Agent |
| Outbound from AI | Synthflow AI → SIP INVITE → Genesys BYOC trunk → PSTN → Callee |
| Outbound from Genesys via AI | Genesys campaign/flow → BYOC trunk → Synthflow AI → handles call |
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## Verification
| Test | What to check |
| ---------------------- | ------------------------------------------------------------------------------------------------------------------------------- |
| Inbound call | Call the DID from an external phone. Verify it routes through Genesys Architect to the Synthflow AI agent. Check two-way audio. |
| Transfer to live agent | Trigger a transfer during a Synthflow call. Confirm the call lands in the correct Genesys queue with caller ID intact. |
| Outbound from Genesys | Route a Genesys call flow to the Synthflow trunk. Verify the AI agent answers. |
***
## Quick Troubleshooting
| Issue | Fix |
| ---------------------------- | ------------------------------------------------------------------------------------------------------------------------ |
| Registration fails | Check Termination URI matches the Genesys BYOC FQDN. Verify credentials and firewall rules (UDP/TCP 32681 or TLS 32682). |
| "No route found" on outbound | Ensure the Synthflow trunk is assigned to the correct Site and the Site has matching Number Plans. |
| Calls rejected by Genesys | Verify Digest Authentication credentials match between Genesys and Synthflow. Check SIP Access Control whitelist. |
| One-way or no audio | Ensure RTP ports are open bidirectionally. Confirm G.711 codec on both sides. |
| Caller ID missing | Set Caller Address in Genesys trunk config. Verify E.164 format in SIP From header. |
| Wrong org / calls go nowhere | The Inbound SIP Termination Identifier must be unique per org and match the FQDN used in Synthflow's Termination URI. |
Genesys Cloud offers a Call Simulator tool (under Telephony) to test dial plans and routing without impacting production. Use it to validate your setup before live testing.
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## Reference
* Genesys BYOC Cloud Trunk Setup: help.genesys.cloud/articles/create-a-byoc-cloud-trunk/
* Genesys SIP Routing Configuration: help.mypurecloud.com/articles/configure-sip-routing-for-a-byoccloud-trunk/
* Genesys External Trunk Settings: help.genesys.cloud/articles/external-trunk-settings/
* Synthflow SIP Overview: docs.synthflow\.ai/about-sip
* Synthflow X-EI SIP Header: docs.synthflow\.ai/x-ei-sip-header