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

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.

PING Ted Stevens

[Rob T Firefly]

This can only mean twice as much material filling up the tubes.

Re:Doubling? hardly

[jdmicklos]

The only real advantage to adding in "unused" octaves is in order to transmit overtones. Overtones shape the sound you can hear even though they may not be hear directly. Think about it as if you were to have a G note at 120 dB playing in an octave that you couldn't hear. It would still cause all things around with a fundamental frequency that is a "G" to vibrate as well as color certain audible noises.

Only a slight improvement

[riflemann]

Actually, I've used Asterisk to pass through 24KHz Speex encoded audio - very impressive sound quality, but only works when the SIP channel is client to client.

In theory a SIP server doesn't need to know all of the codecs a client supports - the clients themselves negotiate any compatible protocol.

Of course, if the sip server puts itself in the path (such as when it needs to pass through to PSTN or firewalled clients), then 8KHz is the (till now) maximum supported rate.