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

[Sergey Okhapkin]

Yes, they accept SIP signalling from a preset IPs, but RTP can be redirected 
after the call is accepted. All carriers I use support SIP reinvites.

On Thursday 29 March 2007 08:24, Loic Didelot wrote:
> 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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