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PSTN to SIP calling


If your network runs over the PSTN and you are in the process of peering with us, please take a moment to ensure the following requirements are met:

  • Gateway must have session inactivity timer to prevent runaway calls due to lost signal
  • SIP (RFC 3261) compliance
  • Gateway must understand DTMF events via SIP INFO method
  • Recommended Gateway: Cisco AS5350
  • Minimum Codec support: g.711 (a & u), GSM, ILBC
  • Optional: Additional Codec support: g.723, g.726, g.729
  • Support for TCP and UDP transport protocols
  • Periodic or realtime feed of numbers on your network (flat file or lookup directory)
  • Optional: End-to-end IP based transport for efficiency Internet
  • Optional: Support Transport Layer Security (TLS) for secure communications

Meeting these requirements will ensure successful implemenation of your SIP gateway and interoperability between our networks.



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Home > network > peering > calls