***

title: SIP Integration with Talkdesk
subtitle: null
slug: sip-with-talkdesk
description: null
-----------------

<Note>
  SIP trunking is only available on the Synthflow Enterprise Plan.
</Note>

This guide covers connecting Talkdesk to Synthflow via SIP trunk using Talkdesk's External SIP Trunking capability. Talkdesk routes calls to a Synthflow AI agent, which can handle AI conversations and transfer callers back to live Talkdesk agents.

## Prerequisites

| Requirement  | Detail                                                                                        |
| ------------ | --------------------------------------------------------------------------------------------- |
| Talkdesk     | Active account with Admin access; Talkdesk CX Cloud with SIP Trunking enabled                 |
| Synthflow    | Enterprise Plan with a configured AI assistant (v2.0+)                                        |
| Phone Number | At least one DID provisioned in Talkdesk (E.164 format)                                       |
| Network      | Firewall allows SIP (UDP/TCP 32681 or TLS 32682) and RTP media between Talkdesk and Synthflow |

## Talkdesk Configuration

All configuration is done in the **Talkdesk Admin** panel under **Integrations** → **SIP Trunking**.

### Create the SIP Trunk

1. In Talkdesk Admin, navigate to **Integrations** → **SIP Trunking**.
2. Click **Add SIP Trunk**.
3. Configure the trunk:

| Field                | Value                                               |
| -------------------- | --------------------------------------------------- |
| Name                 | `Synthflow AI`                                      |
| SIP Server / Proxy   | `sip.synthflow.ai`                                  |
| Port                 | `32681` (UDP/TCP) or `32682` (TLS)                  |
| Transport            | UDP/TCP or TLS                                      |
| Authentication Type  | IP-based (whitelist Synthflow IPs) or Digest        |
| SIP Username         | From Synthflow — only required if using Digest auth |
| SIP Password         | From Synthflow — only required if using Digest auth |
| Max Concurrent Calls | Per your Synthflow plan / expected call volume      |

4. Under **Allowed IPs**, add Synthflow's SIP signaling IP addresses. Contact Synthflow support for the current list.
5. Click **Save**.

<Note>
  For UDP/TCP signaling use port 32681. For TLS, use port 32682.
</Note>

### Configure Inbound Call Routing

Route inbound calls to Synthflow through Talkdesk's call routing engine.

1. In Talkdesk Admin, go to **Voice** → **Ring Groups** or **IVR** (depending on your setup).
2. Create a new routing action or edit an existing IVR menu.
3. Add an **External Transfer** or **SIP Transfer** step:
   * Set the destination to the DID associated with your Synthflow SIP number.
   * Select the **Synthflow AI** SIP trunk as the outbound route.
4. Set a timeout (recommended: 30 seconds) and configure a fallback to a Talkdesk agent ring group.
5. Save and activate the routing configuration.

<Note>
  For skills-based routing, use Talkdesk's routing rules engine to send specific call types (e.g., by language, department, or IVR selection) to different Synthflow assistant numbers.
</Note>

### Configure Transfer Back to Talkdesk (AI → Live Agent)

1. Note your Talkdesk SIP domain or the DID for the target agent ring group.
2. Configure Synthflow's transfer action to route to the Talkdesk ring group DID via the SIP trunk.
3. Talkdesk will receive the transfer and route to the next available agent in the ring group.

## 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 provisioned in Talkdesk).
4. Fill in the SIP connection details:

| Field                        | Value                                            |
| ---------------------------- | ------------------------------------------------ |
| Termination URI              | Your Talkdesk SBC gateway hostname or IP address |
| SIP Username                 | Your SIP authentication username                 |
| SIP Password                 | Your SIP authentication password                 |
| Outbound Proxy (if required) | Talkdesk SBC proxy address (if required)         |

<Note>
  The Termination URI is your Talkdesk SBC address — this routes Synthflow calls back into Talkdesk for PSTN termination and agent transfers.
</Note>

<Warning>
  Be precise when copying SIP credentials. Incorrect values will cause registration failure.
</Warning>

### 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.

<Note>
  The SIP trunk is created at the moment of attaching the SIP number to the assistant.
</Note>

***

## Call Flow Summary

| Scenario                      | Flow                                                                          |
| ----------------------------- | ----------------------------------------------------------------------------- |
| Inbound to AI                 | Caller → PSTN → Talkdesk → IVR/Ring Group → SIP Transfer → Synthflow AI Agent |
| AI transfers to live agent    | Synthflow AI → SIP transfer → Talkdesk SBC → Ring Group → Live Agent          |
| Outbound from Talkdesk via AI | Talkdesk IVR/campaign → SIP Trunk → Synthflow AI → handles call               |

***

## Verification

| Test                   | What to check                                                                                                                |
| ---------------------- | ---------------------------------------------------------------------------------------------------------------------------- |
| Inbound call           | Call the Talkdesk DID from an external phone. Verify it routes to Synthflow AI. Check two-way audio.                         |
| Transfer to live agent | Trigger escalation during a Synthflow call. Confirm the call lands in the correct Talkdesk ring group with caller ID intact. |
| Outbound from Talkdesk | Route a Talkdesk IVR call to the Synthflow trunk. Verify the AI agent answers.                                               |

***

## Quick Troubleshooting

| Issue                        | Fix                                                                                                                                 |
| ---------------------------- | ----------------------------------------------------------------------------------------------------------------------------------- |
| Registration fails           | Check Termination URI, credentials, and firewall (UDP/TCP 32681 or TLS 32682). Verify Synthflow IPs are in Talkdesk's allowed list. |
| Calls not reaching Synthflow | Confirm IVR/ring group routing action points to the Synthflow DID via the correct SIP trunk.                                        |
| One-way or no audio          | Ensure RTP ports are open bidirectionally. Confirm G.711 codec is negotiated on both sides.                                         |
| Caller ID missing            | Set the From header to the caller's E.164 number. Verify Talkdesk passes ANI through the SIP INVITE.                                |
| Transfer fails               | Verify the Talkdesk ring group DID is correct and the SIP trunk is active. Check Talkdesk transfer logs.                            |

***

## Reference

* Talkdesk SIP Trunking Documentation: docs.talkdesk.com
* Talkdesk Admin Guide: support.talkdesk.com
* Synthflow SIP Overview: docs.synthflow\.ai/about-sip
* Synthflow X-EI SIP Header: docs.synthflow\.ai/x-ei-sip-header