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

[Sergey Okhapkin]

In many cases asterisk is not in RTP audio path and those settings will not 
work.

On Thursday 29 March 2007 08:04, Manuel Wenger wrote:
> 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
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