[asterisk-dev] Silence detection on Dial application (mainly with SIP)

Alex Balashov abalashov at evaristesys.com
Tue Jul 14 07:07:57 CDT 2009


Nir Simionovich wrote:

>   On callback based systems, I've noticed that their exists an issue 
> with disconnecting the call when
> there is silence on the line. Asterisk is fairly capable of detecting 
> the absence of RTP on SIP, however,
> if RTP is still existant between the nodes, Asterisk doesn't disconnect 
> - although the RTP is just silence.

If something like this were implemented, it would not be particular to 
Dial(), but rather a feature of the way RTP streams are processed in 
general--although, perhaps, the feature could be enabled on a given call 
leg by a flag passed to Dial().

This is computationally expensive, since Packet2Packet bridging would 
have to be done away with and a certain sample of waveform would need to 
be reconstructed (and buffered, and saved persistently across multiple 
packets--20 ms packetisation duration won't cut it) by Asterisk, much as 
is done with in-band DTMF detection.  It is also difficult and unlikely 
to work well given the ubiquity of background noise - whether real or a 
figment of something in the transmission path (i.e. static, buzz) - that 
has practically infinite manifestations;  in this respect, AMD would 
seem to be easier from a mathematical perspective.

Then again, people said the same thing about AMD and, instead, it 
sort-of works.

Still, there's a fundamental fallacy in all this for as long as we have 
analog lines and/or crappy variable bit-rate codecs on cell phones 
and/or all the other acoustic artifices of the PSTN.  If every channel 
consisted of pristine, noise-canceled HD voice, it might stand a chance.

When John Sculley (the Pepsi guy, later Apple CEO) was pushing the 
development of the Newton PDA, he failed to appreciate a fact known to 
almost every programmer and computer science student:  Handwriting 
recognition is impossible.  Not "some kind of handwriting recognition," 
but rather the kind his "vision" aimed for.

-- Alex

-- 
Alex Balashov
Evariste Systems
Web     : http://www.evaristesys.com/
Tel     : (+1) (678) 954-0670
Direct  : (+1) (678) 954-0671



More information about the asterisk-dev mailing list