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.

Re:Good Work

[Tychon]

But for those of us with a bit of trouble hearing, or when speaking with a person that has a thick and or foreign accent, that extra quality is the difference between a conversation and a stream of "What'd you say?"

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.

Re:So what?

[cdrudge]

Yeah, but the on hold music sounds great!

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.

Big Whoopie

[jmorris42]

The problem isn't making a software based IVR system or even a softswitch run at a better rate. Now find me a SIP phone that runs at anything other than 8Khz. No, I'm not talking about a F/OSS softphone, but a real hardphone. They have the minimum DSP power the manufacturers can get away with to support 8Khz. Now find me a PRI that can interface with it. For now that is still an issue.

Skype has been running their softphones at higher than 8Khz/8bit so their softswitch obviously was the first widely deployed one to leave 64kbit max quality behind.

Yes, someday all telephony (except legacy telco stuff that will never change, which will be a shrinking market) will offer higher quality audio and an option for video. But not for a few more years until the saturation of next gen telephony products gets better.

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!