Skip to content

New: BD, IN and PK Voice OTP capacity now live, test routes issued same day.

Request Test Route

SIP / RTP

SIP interconnect

A SIP interconnect for voice termination, Voice OTP and TTS, with an engineer on the call for interop.

  • SIP over UDP or TCP
  • RTP, G.711 A-law / µ-law and G.729
  • Static IP or digest
Get test credentials
SMPP 3.4 binds HTTP API REST + JSON SIP / IP AUTH trunks ONE ACCOUNT one balance LIVE
At a glance SIP over UDP or TCP RTP, G.711 A-law / µ-law and G.729 Static IP or digest RFC 2833

Voice services connect over standard SIP. Signalling runs over UDP or TCP, media over RTP, with G.711 and G.729 negotiated at setup.

Interop first, minutes second

Every new trunk starts with an interop session with one of our engineers on the call. Codec negotiation, DTMF handling and CLI presentation are confirmed against your actual equipment rather than assumed from a specification sheet, and then test minutes run on the trunk you will use in production.

  • Signalling SIP over UDP or TCP
  • Media RTP, G.711 A-law / µ-law and G.729
  • Authentication Static IP or digest
  • DTMF RFC 2833
  • Capacity Concurrent channels agreed per account

Get credentials

Credentials are issued with your test account, on the same routes production will use.

Request access

Start in 24 hours

Test a live route before you commit a single dollar

Send us your destinations and traffic profile. We return a costed route plan and a free test account, usually the same business day.

Request Test Route Talk to routing team
Chat on WhatsApp