Support Agentforce Voice with SIP

When transmitting voice calls to Agentforce agents, customers can select from multiple types of communication protocols: Public Switched Telephone Network (PSTN), Session Initiation Protocol (SIP), or Dynamic Voice Routing. To fully support customers who want to implement and use Agentforce Voice with SIP protocol, you must become a verified Agentforce Voice with SIP partner.

If you’re a verified Agentforce Voice with SIP partner, to get started, your customers can start setting up Agentforce Voice with SIP by referring to the Create a Voice-Enabled Service Agent Help page.

If you’re not an Agentforce Voice with SIP partner, perform the steps in this document to obtain verification.

Becoming a Verified Agentforce Voice with SIP Partner

The flowchart represents a high-level flow of the Agentforce Voice with SIP setup process for the Contact Center as a Service (CCaaS) partner. The process starts when you contact your Salesforce team with a request to support Agentforce Voice with SIP and ends when the Salesforce team endorses you as a verified Agentforce Voice with SIP partner.

Flowchart representing the steps that CCaaS partners perform to become Agentforce Voice with SIP partners.

Step 1: Perform Initial Assessment

Review and complete the SIP support and header requirements before you begin the Agentforce Voice with SIP partner-verification process.

SIP Support Requirements

To ensure a successful SIP integration, verify that you meet the support requirements.

Understand the connectivity standards required to establish a SIP trunk.

SIP over TLS (mTLS isn’t supported)
Media over SRTP
Static signaling and media IP addresses
Codecs Supported: G.711 PCMU and PCMA
SIP INVITE header fields

Perform the verification steps to ensure your environment is prepared.

Verify that real-time communication systems are set up to support the SIP standard.
Verify that user-to-user information (UUI) headers with Hex encoding are supported.
Verify that you have the full public address range by regions for allowlisting.

Provide your partner name and the customer’s data center location.

Partner name, such as Genesys
Customer data center location, such as London

UUI and SIP Standard Header Fields and Values

Gather the values for the UUI and SIP header fields.

To transfer an incoming call to an Agentforce agent using SIP, Salesforce uses UUI headers to pass call context and authentication data over SIP trunks. The UUI header carries key parameters that Salesforce uses to route the call. Alternatively, if custom SIP headers are enabled, Salesforce can use X-headers for those key parameters instead.

During this setup process, you’ll use the header fields to perform basic SIP connectivity and end-to-end verification tests.

For a list of UUI and SIP header fields, refer to Configure Inbound Call Transfers When Using SIP.

Custom SIP Headers and SIP REFER

Determine whether you plan to support X-headers in SIP invite requests and the SIP REFER protocol. Custom headers are used to send contextual data such as customer intent, language preference, or authentication information to an Agentforce agent. They are also used to support SIP REFER to handle disconnects, if this is a requirement for your customers.

For more information, refer to Include Custom SIP Headers and REFER Support. For a list of alternative X-headers to pass key parameters instead of using the UUI header, refer to Configure Inbound Call Transfers When Using SIP.

Step 2: Gather SIP Configuration Details

To collect the information that Salesforce requires to start the SIP integration process, complete the SIP Signaling Configuration Form.

Make a copy of the SIP Signaling Configuration Form and complete it.
Send the completed document to your Salesforce team, who then submits a request to the development team to integrate you as an Agentforce Voice with SIP partner.
Wait for your Salesforce team to confirm that the integration has completed so that you can move on to the next step.

Step 3: Perform Basic SIP Connectivity and End-To-End Verification Tests

After your Salesforce team confirms that the integration has completed, perform basic SIP connectivity and end-to-end verification tests.

Connectivity Test

Use the UUI header (in Hex code) to initiate a test call to the Salesforce Session Border Controller (SBC).
Sample UUI Header
1{"orgId":"echo-test-org","scrt2Domain":"afv-wss-echo:7442"}
To confirm that connectivity is successful, after you initiate the call, listen for the ECHO.

To verify that signaling works as expected, perform basic call tests and verify the SIP traces at both ends.

The sample SIP INVITE message is used to initiate a user-to-user call.

1INVITE sip:+11060321122@10.10.1.148:5060;user=phone SIP/2.0
2Via: SIP/2.0/UDP 10.41.152.80:5060;branch=z9hG4bK10Bf15a9a98700aeb9c
3From: <sip:+919247XX8773@afv.thb-perf06.unified-engagement1.perf2-uswest2.aws.sfdc.is>;tag=gK103154c4
4To: <sip:+11060321122@afv.thb-perf06.unified-engagement1.perf2-uswest2.aws.sfdc.is>;transport=TLS
5Call-ID: 608213684_125667659@10.41.152.80
6CSeq: 850640 INVITE
7Max-Forwards: 68
8Allow: INVITE, ACK, CANCEL, BYE, REGISTER, REFER, INFO, SUBSCRIBE, NOTIFY, PRACK, UPDATE, OPTIONS, MESSAGE, PUBLISH
9Contact: <sip:+919247XX8773@10.41.152.80:5060>
10User-to-User: 7B226F72674964223A226563686F2D746573742D6F7267222C227363727432446F6D61696E223A226166762D7773732D6563686F3A37343432227D;encoding=hex
11User-Agent: VineProx-v2.3
12Supported: timer, 100rel, precondition, replaces
13Session-Expires: 1800
14Min-SE: 90
15Content-Length:   235
16Content-Disposition: session; handling=required
17Content-Type: application/sdp
18
19, msgLen = 1504.
20324 12092025 042639.296880:1.01.00.22619.Info    .SIPCM: SipFeSocketSendMsg - msg = v=0
21o=Sonus_UAC 520733 642062 IN IP4 10.22.88.80
22s=SIP Media Capabilities
23c=IN IP4 10.22.88.80
24t=0 0
25m=audio 1072 RTP/AVP 0 101
26a=rtpmap:0 PCMU/8000
27a=rtpmap:101 telephone-event/8000
28a=fmtp:101 0-15
29a=sendrecv
30a=maxptime:20
31, msgLen = 224.

Repeat the steps for each mapped region that you support.

End-To-End Testing

Configure a Salesforce org with partner telephony fully set up.
Set up Agentforce Voice with SIP. For the steps, see the Create a Voice-Enabled Service Agent Help page and then the Connect a Service Agent to Partner Telephony Help page.
Perform an inbound call that enters your partner telephony system.
Transfer the call to Salesforce via SIP.
After the call connects to the Agentforce Voice agent, escalate the call to a service rep.
Return the call back to your partner telephony system and route the call to the service rep inside Salesforce via Salesforce Voice. Verify that the Agentforce Voice Call record is linked to the Voice Call routed to the service rep.

Step 4: Final Review with Salesforce

Schedule a call with Salesforce to review your end-to-end setup. To facilitate this review, share documentation of your setup, such as a recording. The documentation should explain:

Your telephony provider configuration for Agentforce Voice with SIP
The flow used to transfer the call to an agent and escalate the call to a rep
How you link related calls to pass call context

Step 5: Publish Supported Agentforce Voice Regions

After you’ve thoroughly tested the connection, wait for Salesforce to notify you that you’re endorsed as a supported Agentforce Voice with SIP partner.

Congratulations. As a verified Agentforce Voice with SIP partner, you can work with your customers to implement Agentforce Voice with SIP in their orgs. Your customers can refer to the Create a Voice-Enabled Service Agent Help page to get started with Agentforce Voice.