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remove mod_isac and references #252

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6dc4b56
remove mod_isac and references
May 4, 2025
850a7ff
Merge branch 'main' into issue205/remove-module-mod_isac
hey-august May 23, 2025
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4 changes: 1 addition & 3 deletions docs/FreeSWITCH-Explained/Codecs-and-Media/Audio-Codecs_1048675.mdx
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Expand Up @@ -22,9 +22,7 @@ The following codecs can be used when setting codec\_string and absolute\_codec\
* opus@8000h@100i - Opus 8khz using 100 ms ptime
* opus@8000h@120i - Opus 8khz using 120 ms ptime
* provided by [mod\_opus](../Modules/mod_opus_6586850.mdx#about)
* iSAC
* provided by [mod\_isac](../Modules/mod_isac_6586969.mdx#about)
* CODEC2 2550bps) 8000hz 20ms
* CODEC2 2550bps) 8000hz 20ms
* provided by: [mod\_codec2](../Modules/mod_codec2_6587354.mdx#about)
* SILK Skype Audio codec.
* Provided by mod\_silk.
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19 changes: 0 additions & 19 deletions docs/FreeSWITCH-Explained/Modules/mod_isac_6586969.mdx
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This file was deleted.

2 changes: 1 addition & 1 deletion ...ITCH-Explained/Troubleshooting-Debugging/Packet-Capture/VoIPmonitor_7143574.mdx
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Expand Up @@ -7,7 +7,7 @@

From [http://www.voipmonitor.org](http://www.voipmonitor.org/)

VoIPmonitor is open source network packet sniffer with **commercial frontend** for SIP RTP RTCP and SKINNY(SCCP) VoIP protocols running on linux. VoIPmonitor is designed to analyze quality of VoIP call based on network parameters - delay variation and packet loss according to ITU-T G.107 E-model which predicts quality on MOS scale. Calls with all relevant statistics are saved to MySQL or ODBC database. Optionally each call can be saved to pcap file with either only SIP protocol or SIP/RTP/RTCP/T.38/udptl protocols. VoIPmonitor can also decode speech and play it over the commercial WEB GUI or save it to disk as WAV. Supported codecs are G.711 alaw/ulaw and commercial plugins supports G.722 G.729a G.723 iLBC Speex GSM Silk iSAC. VoIPmonitor is also able to convert T.38 FAX to PDF.
VoIPmonitor is open source network packet sniffer with **commercial frontend** for SIP RTP RTCP and SKINNY(SCCP) VoIP protocols running on linux. VoIPmonitor is designed to analyze quality of VoIP call based on network parameters - delay variation and packet loss according to ITU-T G.107 E-model which predicts quality on MOS scale. Calls with all relevant statistics are saved to MySQL or ODBC database. Optionally each call can be saved to pcap file with either only SIP protocol or SIP/RTP/RTCP/T.38/udptl protocols. VoIPmonitor can also decode speech and play it over the commercial WEB GUI or save it to disk as WAV. Supported codecs are G.711 alaw/ulaw and commercial plugins supports G.722 G.729a G.723 iLBC Speex GSM Silk. VoIPmonitor is also able to convert T.38 FAX to PDF.

## Sniffer Installation

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1 change: 0 additions & 1 deletion provisioning/nginx/redirects.map
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Expand Up @@ -1569,7 +1569,6 @@
/freeswitch/confluence-to-docs-redirector/display/FREESWITCH/mod_httapi /freeswitch/FreeSWITCH-Explained/Modules/mod_httapi_3966423;
/freeswitch/confluence-to-docs-redirector/display/FREESWITCH/mod_http_cache /freeswitch/FreeSWITCH-Explained/Modules/mod_http_cache_6587452;
/freeswitch/confluence-to-docs-redirector/display/FREESWITCH/mod_iLBC /freeswitch/FreeSWITCH-Explained/Modules/mod_iLBC_3375542;
/freeswitch/confluence-to-docs-redirector/display/FREESWITCH/mod_isac /freeswitch/FreeSWITCH-Explained/Modules/mod_isac_6586969;
/freeswitch/confluence-to-docs-redirector/display/FREESWITCH/mod_janus /freeswitch/FreeSWITCH-Explained/Modules/mod_janus_20709557;
/freeswitch/confluence-to-docs-redirector/display/FREESWITCH/mod_java /freeswitch/FreeSWITCH-Explained/Modules/mod_java_3966491;
/freeswitch/confluence-to-docs-redirector/display/FREESWITCH/mod_json_cdr /freeswitch/FreeSWITCH-Explained/Modules/mod_json_cdr_4653124;
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