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On this page
  • Prerequisites
  • Typical trunk settings
  • NAT and failover
  • Integration matrix
  • Importing your numbers
  • Secure calls
  • Secure signaling requirements
  • Secure media requirements
  • Enabling secure calls
  • Enabling secure media
LaunchTelephony

Connect your phone system (SIP/PBX)

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Synthflow SIP trunking network guidelines

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Connect Synthflow to your PBX or carrier via SIP trunking.

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

Before starting, check the SIP Integration overview to see which support tier your provider falls into. Tier 1 providers (Twilio, Telnyx, RingCentral, Vonage) have native, actively tested integrations. Tier 2 and other providers may require Professional Services involvement.

This guide is intended for system administrators and technical staff with experience configuring SIP trunks and PBX integrations. If you’re new to SIP trunking, consider starting with our simpler options like Twilio integration or buying a number directly.

Prerequisites

  • Public IP(s) or SBC hostname for signaling/media.
  • SIP credentials or IP‑ACLs.
  • Desired codecs and DTMF method.
  • Firewall rules to allow our SBC addresses and ports.
  • Enterprise account with custom number import access.

Note
We provide SIP endpoints and credentials in the Synthflow Console after you request access.

Typical trunk settings

  • Transport: UDP or TCP.
  • Codecs: G.711 μ‑law/A‑law. Opus.
  • DTMF: RFC 2833 (RTP events). SIP INFO.
  • Registration: Static trunk (IP‑based) or SIP REGISTER (digest auth).
  • NAT: Symmetric NAT supported with media pinholes and rport.

Tip
Start with G.711 for maximum compatibility across systems.

NAT and failover

  • Configure SIP timers and keep‑alives to maintain NAT bindings.
  • Provide multiple destination IPs or DNS SRV for failover when available.
  • Set reasonable retry intervals to avoid storming.

Integration matrix

PlatformSIP transport (UDP/TCP)DTMF (RFC2833 / SIP INFO)Registration (Static/REGISTER)NAT traversalCodecs (G.711 / Opus)
AsteriskUDP, TCPRFC2833, SIP INFOStatic, REGISTERYesG.711; Opus
Cisco (CUCM/CUBE)UDP, TCPRFC2833Static, REGISTERYesG.711; Opus
AvayaUDP, TCPRFC2833StaticYesG.711
GenesysUDP, TCPRFC2833StaticYesG.711; Opus
Generic SIP trunkUDP, TCPRFC2833, SIP INFOStatic, REGISTERYesG.711; Opus

Some features depend on your specific version and configuration. For provider-specific setup guides and support tiers, see the SIP Integration overview.

Importing your numbers

Once you have configured your SIP trunk or PBX integration with the settings above, you need to import your phone numbers into Synthflow so we can route calls correctly.

The import process is done through the Synthflow portal and requires you to specify:

  • Your phone number in E.164 format
  • SIP domain and authentication credentials
  • Outbound proxy settings (if using non‑standard ports or mid‑registrar)
For detailed step‑by‑step instructions and field‑level explanations, see Custom phone numbers.

Tip
Have your SIP credentials, domain, and trunk configuration details ready before starting the import process.

Secure calls

If you require additional security for your calls, the following features are available:

  • Secure signaling: Protects any sensitive data you might be exchanging via SIP sessions, such as custom headers.
  • Secure media: Encrypts call audio between your carrier/PBX and Synthflow Edge when SRTP is successfully negotiated. You must use secure signaling to enable secure media.

Secure signaling requirements

  • Your carrier or PBX must be provisioned with a valid certificate signed by a trusted authority.
  • Your carrier or PBX must support TLSv1.2+.

Secure media requirements

  • Your carrier or PBX must support SRTP-DTLS or SRTP-SDES.
  • Your carrier or PBX must support OSRTP (RFC8643).

Enabling secure calls

Inbound calls

Point the SIP trunk to Synthflow’s secure SIP server port using TLS-secured URI: sip:sip.synthflow.ai:32682;transport=tls.

Outbound calls

The secure call feature requires the use of custom phone numbers. Follow the instructions for adding custom phone numbers and configure the outbound proxy with the SIP URI of the secure endpoint of your carrier or PBX.

Enabling secure media

For calls that use secure signaling, Synthflow will always attempt to negotiate encrypted media (SRTP). If SRTP cannot be negotiated with your carrier or PBX, media will fall back to non-encrypted RTP while signaling remains encrypted. This means media encryption is best-effort and is not guaranteed unless your carrier or PBX is configured to require SRTP and reject non-encrypted RTP.