Telephony | Synthflow

Telephony connects your voice agents to the phone network, so they can place and receive calls on real phone numbers. You can buy a number in Synthflow, bring your own Twilio account, or connect your existing carrier or PBX over SIP.

Setup options

Option	Use it when	Your numbers live in	Start here
Buy a number	You want to test an agent or run a simple setup.	Synthflow	Buy a number
Twilio	Your team already runs calls and SMS on Twilio.	Your Twilio account	Twilio
SIP trunk or PBX	You need to connect an existing carrier, PBX, or contact center.	Your carrier or PBX	Connect over SIP

We recommend starting with a purchased number to validate your call flows. Purchased numbers cover the U.S., Canada, and Australia. For production, move to Twilio or SIP so you keep control of your numbers, regional rules, and configuration.

SIP trunking

Enterprise Feature: SIP/PBX integration requires an Enterprise plan subscription. Contact our sales team to enable this feature.

SIP trunking lets you keep your current provider or PBX and route calls through Synthflow agents. You do not need to upgrade or change your existing phone system. We adapt transport, codecs, DTMF, and authentication to your environment.

Providers fall into two support tiers, listed in the telephony integrations overview:

Tier 1 providers, such as Twilio, Telnyx, RingCentral RingCX, Vonage, 8x8, Zoom, and Five9, have native integrations that we test and support.
Tier 2 providers, such as Dialpad, 3CX, Aircall, Genesys Cloud, and Talkdesk, have been deployed with customers but are not regression-tested.

Any other system that supports standard SIP, including Asterisk, Cisco, and Avaya, can connect with the settings in the integration matrix.

Infrastructure and regions

We run our own Session Border Controllers (SBCs) and media infrastructure, and manage routing and capacity in-house.

Synthflow operates three regions: Global, US, and EU. Your SIP trunk must point to the region that matches your workspace’s data region. The region addresses list the signaling and media IPs to allow in your firewall.

We can also deploy a point of presence closer to your callers. The LATAM case study shows the results for one customer.

Call quality

For traffic that stays within a region, we target a round-trip time under 100 ms and a MOS above 4.2. Results depend on your access network, peering, and carrier path.

Call quality defines each metric, explains how to measure it on your network, and covers regional deployments.

FAQ

Can I get a SIP trace or PCAP for a call?

Yes. Open the call in Logs and select the Telephony tab to see the signaling diagram and download the PCAP. See SIP Call Ladder.

Why did a call fail?

Open the failed call in Logs to see its error message, then look it up in Telephony Errors for the cause and fix.

Is there a limit on how many calls run at once?

Yes. Each workspace has a concurrent call limit and a call attempts per second (CAPS) limit. See Concurrency for how to find your limits and what happens when you exceed them.