[Preview] Talkdesk SIP Interconnect – Knowledge Base

Important Note: This feature is available in Preview for select customers.

Overview

Talkdesk SIP Interconnect enables Admins to connect Talkdesk to third-party telephony platforms over Session Initiation Protocol (SIP) trunking, so contact center and back-office teams on separate systems can communicate as a single, unified environment. This guide explains the feature and takes Admins through the full configuration, from trunk creation to verification.

Understanding Talkdesk SIP Interconnect
Key Features
Use Cases
Prerequisites
Creating a SIP Trunk
Configuring the SIP Trunk
Configuring Call Routing
Setting Up a Microsoft Teams SIP Trunk
Verifying the Setup
Peering Requirements and FAQs
List of Supported Certificate Authorities
Resources

Understanding Talkdesk SIP Interconnect

Organizations rarely run on a single communications platform. Contact center agents and back-office teams often sit on separate systems, which creates silos, interoperability problems, and inefficient call routing. Replacing an existing back-office telephony investment is rarely practical, so most organizations prefer to integrate their existing platform with Talkdesk rather than replace it.

A basic call handoff between two platforms is not enough on its own. Admins need an integration that is easy to set up, customizable, secure by default, and able to carry context between environments. Talkdesk SIP Interconnect is built for this purpose.

Admins configure SIP trunks between Talkdesk and a third-party telephony environment, such as a Session Border Controller (SBC), a PBX, or Microsoft Teams. Once a trunk is established, calls flow between the two environments according to dial plans and call patterns configured on the Talkdesk Voice Platform, so every call crossing the boundary is routed and restricted according to policies Admins have already defined.

Key Features

UI-Driven Setup: Admins configure SIP trunks entirely through a dedicated Workspace application, from specifying the remote network endpoint to setting security profiles and codecs.
Flexible Dialing: Calls between Talkdesk and connected platforms are not restricted to E.164 formatting. Extensions, variable-length dial strings, or any combination can be used, so existing dialing habits carry over without disruption.
Dial Plan Integration: Talkdesk SIP Interconnect integrates directly with dial plans configured on the Talkdesk Voice Platform, so every call crossing the boundary between platforms is routed, restricted, and handled according to existing policies.
Security by Default: Every SIP trunk is encrypted automatically. Signaling is secured with Transport Layer Security (TLS) 1.2 or 1.3, and media is protected using Secure Real-Time Transport Protocol (SRTP).
Trunk Health Monitoring: Admins enable automatic health checks and email alerts, so they know immediately if a trunk stops responding.

Use Cases

Interconnecting with an existing PBX or SBC: Organizations that have invested in on-premises or cloud telephony infrastructure can connect that environment to Talkdesk without a "rip and replace" migration.
Microsoft Teams interoperability: Agents can reach back-office subject matter experts (SMEs) in Microsoft Teams directly from the Conversations app, consult with them, and transfer calls with context, without incurring Public Switched Telephone Network (PSTN) charges.
UCaaS/CCaaS coexistence: Businesses that have standardized on a Unified Communications as a Service (UCaaS) platform can layer Talkdesk's contact center capabilities on top without disrupting their existing communications investment.

Prerequisites

Permissions: Admin.
Before starting, confirm access to the third-party telephony environment (for example, an SBC or PBX) that the trunk will connect to, since some steps require entering values on that system as well.

Creating a SIP Trunk

Navigate to Admin [1] > SIP Trunk [2].
Click Create Trunk [3].
In the modal, select the SIP trunk type:

Generic SIP Trunk: Used for interconnecting Talkdesk with third-party systems other than Microsoft Teams.
Microsoft Teams: Used specifically for interconnecting Talkdesk with Microsoft Teams.

Note: Selecting the wrong trunk type can lead to unintended call routing outcomes.

Configuring the SIP Trunk

A Generic SIP Trunk has four configuration sections: Basic Information, Network Endpoints, Security and Authentication, and Advanced Settings.

Basic Information

Enter a friendly name for the SIP trunk [1].
Select the region physically closest to the telephony deployment: US, Europe, Asia Pacific, Canada, or South America [2].
Copy the Fully Qualified Domain Name (FQDN) generated for the selected region [3]. Enter this FQDN on the third-party system (for example, the SBC) to peer with Talkdesk.
Optionally, enter a description for the trunk [4].
Select the integration type: Inbound Only, Outbound Only, or Bidirectional. Bidirectional is the default [5].

Network Endpoints

Select the endpoint identifier type for the device in the third-party environment that the trunk connects to: FQDN (default), IPv4 address, or DNS SRV record [1].

Note: Selecting the correct region in terms of geographical proximity to the physical deployment is important, as it directly affects latency and call quality

Note: Talkdesk uses this value to establish a TLS connection to the peer and as the host portion for SIP requests generated from Talkdesk. Admins can optionally override the target host used for signaling.

Configure an Access Control List (ACL) [2] to restrict the source IP address or network range from which SIP traffic will be admitted.

Optionally, enable SIP OPTIONS monitoring to continuously verify trunk health and endpoint availability.

SIP OPTIONS Monitoring

This feature uses the SIP OPTIONS method to send periodic keepalive probes to the peer device and confirm the trunk is reachable. Ensure SIP OPTIONS is also enabled on the peer device, since this affects monitoring accuracy.

Enable [1]: Toggles keepalive and capability query behavior on or off.
Email Notifications [2]: At least one email address that receives alerts when the trunk or an endpoint fails. Admins press Enter or Space to add more than one address.

OPTIONS Interval [3]: The frequency, in seconds, at which keepalive probes are sent. Default: 60 seconds.
Acceptable Response Codes [4]: Defines which SIP responses count as a healthy endpoint, for example, any SIP response (4xx–6xx) or specific codes only. Default: any SIP response.
Consecutive Failures Before Marking Trunk Down [5]: The number of missed responses required before the trunk is flagged offline. Default: 3.
Consecutive Successes Before Marking Trunk Up [6]: The number of successful probe responses required to restore the trunk to operational status. Default: 1.

Security and Authentication

Talkdesk uses a security-by-default approach: Signaling is always encrypted over TLS, and media is always encrypted using SRTP. TLS connection setup requires certificate-based authentication, and Talkdesk strictly enforces the configured TLS version on all incoming and outgoing connection attempts.

Select the TLS version:1.2 or 1.3 [1].
Select the SRTP cipher [2].

Under “Certificates" [3], review the following:

Outbound connections (Talkdesk to the third-party network): The certificate chain presented by the third-party network must resolve to a well-known Certificate Authority (CA). The third-party network must present all identity and intermediate certificates up to the root CA during the TLS handshake.
Inbound connections (the third-party network to Talkdesk): Admins download the CA certificate used to sign Talkdesk's identity and intermediate certificates. Some third-party devices, such as SBCs, require this certificate to be preconfigured before they can authenticate Talkdesk.

Advanced Settings

Under “Media settings” [1], select the audio codec: PCMA (G.711 A-law), PCMU (G.711 μ-law), or both. At least one codec must be selected.
Under “Header extensions” [2], configure the following as needed:

Calling number presentation preference: When enabled, the E.164-formatted calling number is presented in outbound requests.
Remote-Party-ID (RPID): Controls whether the RPID header is included in outbound INVITE and OPTIONS messages.
P-Asserted-Identity (PAID): Controls whether the PAID header is included in outbound SIP messages.
Display name handling: Select Passthrough to forward the originating display name as-is, or Override to specify a custom display name (up to 50 characters) for all outgoing requests. When using Override, Admins must enter a valid SIP URI user portion, up to 50 characters.

Configuring Call Routing

Call routing for SIP Interconnect relies on Call Patterns configured on the Talkdesk Voice Platform.

Outbound Call Routing

Navigate to Admin [1] > Call Patterns [2] > Custom Pattern [3] and click Create pattern [4].

Enter a dial string sequence [5] (for example, 5XXX, +91*, or 345XX), optionally enter a description [6], and select SIP Trunk from the “Type” drop-down [7]. Select a primary SIP trunk [8].

Assign the pattern to a Site, user, or device so end users and devices gain access to route calls over the trunk.

Note: Failover routing is coming soon, allowing Admins to select secondary and tertiary trunks for added routing resilience.

Inbound Call Routing

On the third-party environment, configure all dialog-forming SIP requests sent to Talkdesk (SIP INVITE and SIP OPTIONS) to include the unique Talkdesk-generated trunk identifier, visible in the trunk's Basic Information section. This identifier must be presented as either a Request-URI parameter or a custom header.

Note: Formatting all inbound requests with the trunk identifier is a mandatory requirement. This identifier can be presented on SIP requests sent to Talkdesk in one of the following forms:
a) A Request URI parameter.
b) A custom header.

Below are examples of either scenario:

Unique identifier as a Request URI parameter

INVITE sip:+15551234567@us.peering.sip.talkdeskapp.com;transport=tls;trunk-id=35386492-b106-4bf0-92ae-4c4d2adca5a9 SIP/2.0

Via: SIP/2.0/TLS 192.0.2.1:5061;branch=z9hG4bK77ef4c2312983

Max-Forwards: 70

From: <sip:+15550000001@192.0.2.1>;tag=9fxced76sl

To: <sip:+15551234567@us.peering.sip.talkdeskapp.com>

Call-ID: 3848276298220188511@192.0.2.1

CSeq: 1 INVITE

Contact: <sip:+15550000001@192.0.2.1:5061;transport=tls>

Content-Type: application/sdp

Content-Length: 0

Unique identifier as a custom header

INVITE sip:+15551234567@us.peering.sip.talkdeskapp.com;transport=tls SIP/2.0

Via: SIP/2.0/TLS 192.0.2.1:5061;branch=z9hG4bK77ef4c2312983

Max-Forwards: 70

From: <sip:+15550000001@192.0.2.1>;tag=9fxced76sl

To: <sip:+15551234567@us.peering.sip.talkdeskapp.com>

Call-ID: 3848276298220188511@192.0.2.1

CSeq: 1 INVITE

Contact: <sip:+15550000001@192.0.2.1:5061;transport=tls>

X-Trunk-Id: 35386492-b106-4bf0-92ae-4c4d2adca5a9

Content-Type: application/sdp

Content-Length: 0

On the SIP trunk's Patterns tab, assign the Number Manipulation and Custom Patterns that should apply to incoming calls. Incoming numbers are checked, in order, against Number Manipulation Patterns, extensions, Custom Patterns, and finally PSTN dialing.

For example, a trunk configured with a Number Manipulation pattern of +1408* (Replace, value ${1}) and a Custom Pattern of +1* (Blocked) processes an incoming call from +14085550192 as follows: the Number Manipulation pattern strips the +1408 prefix, producing 5550192; that value no longer matches +1*, so the Blocked Custom Pattern does not apply, and the call proceeds. A call from +16505550100 does not match the Number Manipulation pattern, so it reaches the Custom Pattern step unchanged, matches +1*, and is blocked.

Setting Up a Microsoft Teams SIP Trunk

Admins set up SIP trunking with a Microsoft Teams environment to enable calling between Talkdesk and Microsoft Teams. After selecting Microsoft Teams as the trunk type, Admins select the call routing option that matches their deployment: Direct Routing or Carrier Routing.

Note: Currently, only Microsoft Direct Routing calls are supported through this setup flow. Carrier Routing requires the involvement of Talkdesk Customer Care.