search

x25today / voipwardialer

A Voip Wardialer for the phreaking of 2020
51 stars 11 forks source link

Optimization testing: Ensure pjsua and asterisk sample voice at 8khz #3

Open fpietrosanti opened 4 years ago

fpietrosanti commented 4 years ago

Modem for Analog Line does sample voice at 8khz.

We must ensure that whatever we do within PJSUA and Asterisk is set to work at audio 8khz sampling, including codec of both parties and conference bridge https://dsp.stackexchange.com/questions/22107/why-is-telephone-audio-sampled-at-8-khz

This may also impact the way we do conference media in PJSIP CONFERENCE CALL: https://www.pjsip.org/docs/1.12/pjmedia/docs/html/group__PJMEDIA__CONF.htm#gga95e1f5fd21e35d6317cc218844c2e0baa980479c5fecd3ab1cdc93c795190cc8a

That accept those parameters:

PJMEDIA_CONF_NO_MIC | Disable audio streams from the microphone device. PJMEDIA_CONF_NO_DEVICE | Do not create sound device. PJMEDIA_CONF_SMALL_FILTER | Use small filter table when resampling PJMEDIA_CONF_USE_LINEAR | Use linear resampling instead of filter based.