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

[Nic Bellamy]

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?
>   
I've actually written two different patches for this to solve pretty 
much the same problem:

1. A SIP-channel only patch that works for alaw/ulaw (a bit of a kludge)

2. A technology independent patch that works for all codecs[1] that adds 
a SetGains(txgain,rxgain) function for adjusting incoming channels, and 
adds a V(txgain:rxgain) flag to Dial() for outgoing channels (much less 
of a kludge).

Currently both patches are against the 1.2.x branch.

Want me to send one/both through to you? Or should I wait until you 
offer a bounty... >;-)

Cheers,
    Nic.

[1] Although it'll add a transcode step to linear PCM and back in the 
middle, which if you're using expensive codecs will chew extra CPU and 
add delay (and if you're using G.729, will chew through your channel 
licenses).

-- 
Nic Bellamy,
Head Of Engineering, Vadacom Ltd - http://www.vadacom.co.nz/