Telephony Overview
What telephony enables in Synthflow, with options at a glance.
Telephony powers your agent’s ability to make and receive phone calls. This section explains available options with links to setup guides for each.
Synthflow operates its own enterprise communications stack with an in‑house telephony team. We run our own Session Border Controllers (SBCs) and media infrastructure and are not tied to third‑party vendor footprints. This gives us direct control over routing, capacity, change windows, and incident response.
We can deploy or position infrastructure close to where your traffic originates to minimize latency and jitter. For example, we recently deployed a LATAM PoP for a customer handling high concurrent call volumes. See the case study in LATAM regions for concrete outcomes and methodology. This deployment model is fully repeatable for other regions and compatible with any SIP‑capable system.
We aim for the following performance and quality metrics:
Actual results depend on access networks, peering, and carrier paths. When needed, we collaborate on routing and codec choices to meet your SLA goals.
To validate these benchmarks or troubleshoot quality issues, see the measurement methodology for detailed steps on collecting RTT, jitter, packet loss, and MOS data from your infrastructure.
At a high level:
use Buy a number for quick testing or simpler setups
use Twilio when your team already operates in Twilio
use SIP/PBX when you need to connect existing carrier or phone-system infrastructure
Twilio integration — Use your own Twilio account for calls and SMS. Best for teams already standardized on Twilio. More information here.
Buy a number — Purchase a number in Synthflow for quick evaluation. Learn how to purchase a phone number here.
Connect your phone system (SIP/PBX) — Bring your carrier or PBX via SIP trunking for Enterprise deployments. We maintain native integrations with Twilio, Telnyx, RingCentral, and Vonage, and have community-validated integrations for several other providers. See the SIP Integration overview for the full tier breakdown, or go straight to how to connect your own system.
Start with a test number to validate call flows. Move to Twilio or SIP/PBX for scale and control.
Enterprise Feature: SIP/PBX integration requires an Enterprise plan subscription. Contact our sales team to enable this feature.
We interoperate with common enterprise platforms (Asterisk, Cisco, Avaya, Genesys, and others) as well as carrier SIP trunks. We adapt transport, codecs, DTMF, and auth to fit your environment — no upgrades or changes to your existing phone system are required.
Not all providers receive the same level of validation and ongoing support. Check the SIP Integration overview to see which tier your provider falls into before starting setup.
Follow our guide on how to connect your phone system for typical settings and next steps.
Common telephony questions usually come down to:
Yes. We run our own SBCs and media infrastructure and manage capacity, routing, and changes in‑house.
In many cases, yes. We can place points of presence near your users to reduce latency and jitter. See the LATAM regionalization case study.
Very soon.
Yes. We interoperate with existing SIP systems and trunks without a rip‑and‑replace — no upgrades or changes to your current phone system are needed. Check the SIP Integration overview to see support tiers for specific providers, or go to connect your phone system for setup steps.
We target sub‑100 ms RTT for regional traffic and MOS > 4.2 under healthy network conditions. Results depend on network paths and peering; we can test and tune with you.
We will provide PCAP (SIP trace) of signaling and RTP captures in call logs to help with debugging and analysis. This capability is coming soon.