FreeCalypso > hg > sipout-test-utils
changeset 12:372be50488d6
add README
author | Mychaela Falconia <falcon@freecalypso.org> |
---|---|
date | Mon, 11 Mar 2024 12:45:09 -0800 |
parents | 2cdbd574bba6 |
children | 059b79c9f0c3 |
files | README |
diffstat | 1 files changed, 27 insertions(+), 0 deletions(-) [+] |
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--- /dev/null Thu Jan 01 00:00:00 1970 +0000 +++ b/README Mon Mar 11 12:45:09 2024 -0800 @@ -0,0 +1,27 @@ +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.