FreeCalypso > hg > sipout-test-utils
view README @ 12:372be50488d6
add README
author | Mychaela Falconia <falcon@freecalypso.org> |
---|---|
date | Mon, 11 Mar 2024 12:45:09 -0800 |
parents | |
children |
line wrap: on
line source
This repository contains a collection of programs that need to be run under the following somewhat constraining conditions: 1) The developer-operator running these programs needs to be someone with an active account at BulkVS, AnveoDirect or some other similar provider of SIP trunk access to PSTN; 2) The test programs in this collection need to be run on a machine with a direct, non-NAT'ed connection to public Internet, one whose static & public IP address is whitelisted with the service provider from the previous bullet point as a customer host authorized to send outgoing calls to PSTN on the customer's account. Each of sipout-test-* utilities in this collection is a command line tool that generates an outbound call on PSTN (G.711 codecs only, either PCMU or PCMA) via one of the just-described SIP trunk providers. The breakdown is as follows: * sipout-test-voice is intended for testing voice calls and the in-band signals for GSM 08.62 or 3GPP TS 28.062 TFO; * sipout-test-fsk generates a 300 baud FSK modem call, either Bell103 or V.21; * sipout-test-v22 generates a V.22(bis) modem call at either 1200 or 2400 bps. These programs use SpanDSP library for V-series modulations (required for compilation) and themwi-rtp-mgr (required at run time) to get a properly aligned (even/odd) RTP+RTCP port pair for talking to PSTN-via-SIP.