Telephony Overview
Telephony powers your agent’s ability to make and receive phone calls. This section explains available options with links to setup guides for each.
What telephony enables
- Inbound and outbound calling from your voice agent.
- Number presentation and caller ID policies.
- Optional SMS when supported by your provider.
In‑house telephony and regional deployment
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.
Regional points of presence (PoPs)
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.
Synthflow Benchmark
We aim for the following performance and quality metrics:
- Round‑trip latency: sub‑100 ms when traffic remains regional.
- Call quality: MOS > 4.2 and premium‑range R‑Factor under healthy network conditions.
- Stability: engineered for burst traffic, with attention to jitter and packet loss budgets.
Actual results depend on access networks, peering, and carrier paths. When needed, we collaborate on routing and codec choices to meet your SLA goals.
How to measure
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.
Options of deployment
- 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. Learn about our SIP integrations or see how to connect your own system.
Tip
Start with a test number to validate call flows. Move to Twilio or SIP/PBX for scale and control.
Integrate without re‑architecting
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. Follow our guide on how to connect your phone system for typical settings and next steps.
FAQs
Do you operate your own telephony infrastructure?
Yes. We run our own SBCs and media infrastructure and manage capacity, routing, and changes in‑house.
Can you deploy infrastructure in my region?
In many cases, yes. We can place points of presence near your users to reduce latency and jitter. See the LATAM regionalization case study.
Are there servers in Europe?
Very soon.
Will this work with our existing PBX or carrier SIP?
Yes. We interoperate with existing SIP systems and trunks without a rip‑and‑replace. Start by connecting your phone system or checking all our SIP integrations.
What performance should we expect?
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.
Do you provide SIP traces for troubleshooting?
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.