FreeCalypso > hg > sipout-test-utils
comparison README @ 12:372be50488d6
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| author | Mychaela Falconia <falcon@freecalypso.org> |
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| date | Mon, 11 Mar 2024 12:45:09 -0800 |
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| 11:2cdbd574bba6 | 12:372be50488d6 |
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| 1 This repository contains a collection of programs that need to be run under | |
| 2 the following somewhat constraining conditions: | |
| 3 | |
| 4 1) The developer-operator running these programs needs to be someone with an | |
| 5 active account at BulkVS, AnveoDirect or some other similar provider of SIP | |
| 6 trunk access to PSTN; | |
| 7 | |
| 8 2) The test programs in this collection need to be run on a machine with a | |
| 9 direct, non-NAT'ed connection to public Internet, one whose static & public | |
| 10 IP address is whitelisted with the service provider from the previous bullet | |
| 11 point as a customer host authorized to send outgoing calls to PSTN on the | |
| 12 customer's account. | |
| 13 | |
| 14 Each of sipout-test-* utilities in this collection is a command line tool that | |
| 15 generates an outbound call on PSTN (G.711 codecs only, either PCMU or PCMA) via | |
| 16 one of the just-described SIP trunk providers. The breakdown is as follows: | |
| 17 | |
| 18 * sipout-test-voice is intended for testing voice calls and the in-band signals | |
| 19 for GSM 08.62 or 3GPP TS 28.062 TFO; | |
| 20 | |
| 21 * sipout-test-fsk generates a 300 baud FSK modem call, either Bell103 or V.21; | |
| 22 | |
| 23 * sipout-test-v22 generates a V.22(bis) modem call at either 1200 or 2400 bps. | |
| 24 | |
| 25 These programs use SpanDSP library for V-series modulations (required for | |
| 26 compilation) and themwi-rtp-mgr (required at run time) to get a properly aligned | |
| 27 (even/odd) RTP+RTCP port pair for talking to PSTN-via-SIP. |