A Peering Agreement with SIPphone
Enabling your company to offer free global internet calling
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.
