[asterisk-dev] Feature request: SIP tx/rx gain

[Manuel Wenger]

Hi everyone,
there's a feature we are missing from chan_sip: the possibility to
adjust a SIP peer's/user's TX/RX gain.

The reason for this is that we have an upstream PSTN-to-SIP provider
which converts their SS7 links directly to SIP without reducing gain on
the line. The result is that PSTN calls are much louder than regular
SIP-to-SIP calls. We would like to add, say, "txgain=-10" and
"rxgain=-10" to the SIP peer configuration, so that all audio
coming/going from/to that peer will be made "quieter" (or louder). I
reckon that there would be some DSP programming involved in this, if I'm
not mistaken...

The PSTN provider is using a softswitch where rx/tx gain can only be
adjusted on an SS7 trunk basis. The trunks are shared among several
customers, therefore they won't adjust the gain for us.

Is there anyone who would be willing to create this feature? Is this
something for a bounty? Or should we plain forget about it?

Thanks
-Manuel