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SIP Configuration

Connect PBXs, softswitches and SIP phones to the platform.

The platform speaks standard SIP over UDP, TCP and TLS. Trunks authenticate by credentials (registration) or by allow-listed IP. This page covers the values every PBX needs; the console generates system-specific guides for Asterisk, FreePBX, 3CX and FreeSWITCH.

Connection details

sip-endpoints.txt

Signalling   sip.gtalk2voip.com
  UDP/TCP    port 5060
  TLS        port 5061  (TLS 1.2+)

Media (RTP)  20000-40000/udp
  Codecs     Opus, G.722, G.711 (a/u), telephone-event
  SRTP       optional on UDP/TCP trunks, required on TLS

Registration trunks

Use the trunk ID as the username and the trunk secret as the password. Re-registration interval is 300 seconds; the platform answers OPTIONS pings, so keep-alives from your side are optional.

IP authentication trunks

  • Add each signalling IP of your PBX in the console. CIDR ranges are supported.
  • Calls from unlisted IPs are rejected with 403, and the attempt is logged on the trunk's security tab.
  • Set your PBX to send RFC 2833 DTMF (telephone-event); in-band DTMF is not detected on compressed codecs.

Common issues

  • One-way audio usually means your firewall blocks the RTP return range. Open 20000-40000/udp from our media IPs.
  • Registration flaps behind NAT: lower your router's UDP timeout below 300 seconds or switch the trunk to TCP.
  • 403 on calls that used to work: check whether your PBX's public IP changed and update the allow-list.