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

[Loic Didelot]

Normally when sending traffic to a carrier, asterisk is in the RTP audio
path as carriers limit the access to a defined range of IP addresses.

We work with many carriers and none of them is accepting traffic from
uknown IP addresses. So their switch communicates with our asterisk for
the SIP signalling and the RTP audio. But with decent carriers this
feature is not needed anyway.

Best regards,
Loic Didelot.

On Thu, 2007-03-29 at 08:10 -0400, Sergey Okhapkin wrote:
> 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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-- 
Loic DIDELOT (CTO)
voipGATE S.A.
Tel: +352 20 200 223
Fax: +352 20 200 923
E-mail: ldidelot at voipgate.com
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