In IP telephony?
Does your voice actually get changed? or does the software actually replicate your exact voice pattern? I am learning about Professor Nyquists theorem and find it so amazing!
Public Comments
- IP Telephony is exactly the same as normal telephony in this respect. Everybody's voice sounds a little different thru the phone, but that's more down to the limitations and restrictions of the actual equipment in use. This is why voices now generally sound so much better than the old recordings you sometimes hear of very old telephone conversations, because the equipment being used is so much more advanced. All the software does is to replicate the exact same sequence it recorded at your end of the conversation. If the encoding equipment is not quite up to the same standard as the replication equipment, then you'll have problems. But if the encoding software is better than the decoding software, you STILL won't get better sound. In other words, it will always be governed by the lowest common denominator on the circuit. Scots
- Standard PSTN telephones transmit your voice as analog signals and thus fairly accurately replicate your voice. However, there is some lost information due to the fact that most telephone systems are optimize for voice frequencies in the range of 300-3000 Hz. Thus, higher and lower frequencies are lost. IP telephony, digitizes your voice using various digital sampling techniques and algorithms. How well your voice is replicated over IP-phones is largely dependent on the Codec used to sample and compress your voice. G.711u/a is considered the best standardized codec for your voice reproduction with minimal loss. Other codecs, such as G.729a highly compress the digital signal and thus some loss of audio quality is lost at the expense of reduced bandwidth usage. So, when using IP-telephony, there is always some loss of original signal - how much depends primarily on the codec used to convert the signal into IP packets and back again.
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