This is probably a bit trivial for this forum, but for a college report I’d be grateful if someone could tell me the audio frequency range transmitted in a typical VoIP or PSTN transmission. I’ve read that the old analogue telephones worked at 300 – 3400 Hz where most of the information in speech is concentrated, but can’t seem to find anything quoted for VoIP or PSTN.
All vendors still base it on the same ranges. They dont want to do unnecessary sampling and would rather keep down the sample rate.
Its all based on Nyquists theorum, that you sample at a rate twice the frequency.
This is why TDM is based on 64k channels. Voice is sampled between the frequencies of 0-4000 Hz. 4000 Hz *2 = 8000 samples and 8 bits per sample = 64 kbps per channel.
Rommel gave you the basic spill on analog freq usages but also remember that many AD/DA chip mfgs choose slightly different specs.
Much of the old equipment used less than 3.5K bandwidth and thus the T-1’s could use 56K channels and have some out-of-band signaling, supervisory, and error detection taking place on the diff between 56K-64K (8k left over per channel). Much of the old analog transmission systems also clipped drastically when doing OOS-QC’s 3Khz. They used many non-linear devices and their QC’s tolerated big db losses on the fringes. These losses were desired for analog multiplexing and called guard bands.
Back to what is the standard now…, You posted it. Much of the mic and speaker final end devices don’t have a good freq. response past your posted values. They were good enough for Sprint’s “pin drop” commercial campaign in the ’80s, though.
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