VoIP Calls Double In Quality

anthm writes "From Newsforge and LinuxPR
FreeSWITCH, an open source soft-switch and IVR platform, have announced that they can support 16khz audio calls thus doubling the potential voice quality. They have had successful tests with a conference bridge, a pass-through SIP call and an IVR that reads RSS news feeds with the Cepstral Text-To-Speech Engine."

Voip-Info.org has a good list of business VoIP providers.

Comments

Good Work

[kasgoku]

good work there, but all you need is to get the message across. its not like u r singing on the phone and need good voice quality. just do what's needed.

So what?

[Spazmania]

So what? If you're going to up the sampling rate why not go directly to 44khz stereo (CD quality audio) and be done with it? Jumping from the telephony industry standard 8khz to 16 khz is thoroughly uninspired.

Doubling? hardly

[MacBoy]

I fail to see how adding one additional octave of frequency response to the 6 or 7 currently available, can be called "doubling" the quality.

Marketing BS

[jheath314]

This "improvement" is idiotic. The thing which most limits the quality of a VoIP call is delay and jitter, NOT the sampling rate. Guaranteeing the quality of a telephone conversation over the internet is tricky because the internet was originally designed for best-effort packet delivery, with no guarantees on packet delay, sequence, or even (at the network layer) delivery.

If anything, this feature reduces end-to-end quality by doubling the amount of data being sent down the pipe, as you'd need to buffer more data at the same transmission speed to correct for jitter. Brillant!