Choose Your Communication Protocol

Use one of these protocols to route calls to an agent.

Public Switched Telephone Network (PSTN)

With PSTN, you use a permanent, fixed phone number to transfer incoming calls to your agent. Configure your partner contact center to claim this number and route calls to it.

• Regional availability: US and Canada only
• Call travels over: Phone network
• Call context passed to Salesforce: None. Salesforce identifies the org from the static number. See Connect Related Voice Calls When Using PSTN.
• Partner requirement: Any partner contact center

Choose PSTN for US or Canada setups that don't require partner-side API calls. PSTN relies on Caller ID, so concurrent calls might not link context reliably. For more reliable context linking, use a different communication protocol.

For details, see Configure the Channel and Telephony Settings for PSTN.

Session Initiation Protocol (SIP)

Your telephony partner sends calls directly to Salesforce over the internet, using a SIP trunk. The SIP message includes the call context in its header, eliminating the need for a Salesforce phone number. Configure your partner contact center to transfer inbound calls to the agent using this SIP address.

• Regional availability: Global
• Call travels over: Internet (IP)
• Call context passed to Salesforce: Full context via SIP headers. See Connect Related Voice Calls When Using SIP or Dynamic Routing.
• Partner requirement: Requires integration with a vetted partner. For a list of supported partners, see Agentforce Voice with SIP Regional Availability Chart.

Choose SIP for a direct connection with low call-setup latency. Routing calls over an encrypted IP connection (SIP-over-TLS) bypasses PSTN carriers, avoiding per-country routing variability and translation issues. SIP requires a vetted telephony partner and an IP-level SIP trunk between the partner's SBC and Salesforce's SBC.

For details, see Configure the Channel and Telephony Settings for SIP.

Dynamic Routing

Salesforce dynamically allocates a routable phone number via an API request. Use this number to transfer calls to the agent.

• Regional availability: Global (where supported)
• Call travels over: Hybrid (phone network + internet)
• Call context passed to Salesforce: Full context via the Voice Context Registry. See Connect Related Voice Calls When Using SIP or Dynamic Routing.
• Partner requirement: Any partner contact center that can make an API call

Choose Dynamic Routing to support callers outside the US and Canada without additional telecom licensing or specific partner restrictions.

Note Agentforce Voice with Dynamic Routing doesn't support voice call recording.

For details, see Configure Telephony Vendor Settings When Using Dynamic Routing.

Real-time Streaming

Your telephony partner streams live audio directly to Salesforce over a secure WebSocket connection. The WebSocket handshake carries real-time audio, allowing Salesforce to stream bidirectional audio directly with the voice-enabled agent. You configure your partner contact center to stream inbound call audio to the Agentforce Voice media gateway over WebSocket.

• Regional availability: Global
• Call travels over: Internet (WebSocket)
• Call context passed to Salesforce: To pass the context to Salesforce, see Connect Related Voice Calls When Using Real-Time Streaming.
• Partner requirement: Supported by Amazon Connect.

Choose Real-time Streaming if your partner contact center is Amazon Connect and you want to use direct, low-latency bidirectional audio streaming with Agentforce Voice without requiring PSTN or SIP call transfers.

For details, see Configure Channel and Telephony Settings for Real-Time Streaming.

Comparison Summary of Communication Protocols

Review this comparison table to choose the right communication protocol for transferring incoming calls to agents.