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

Please get the rest of the telcomms to follow.

[rob_squared]

Everything else is stuck at 8khz, so unless your call uses this service end-to-end, there's going to be a downconversion if you're calling someone on a land line. And you'll be stuck with 8khz if you get any calls from someone not on this service.

Still, its a good piece of news, onward and upwards.

*crosses fingers* Please nobody mention video phones. *crosses fingers*

Re:Please get the rest of the telcomms to follow.

[joe 155]

I'll agree to some extent that this is good news, but my friend, 16khz is a lot of packages which will have to be squeezed through the current pipes which I recieve my internets through; so this will make the speed of internets go down to no faster than the standard post. Did you know the other day I got an internet which had been sent on Friday! *mubbles to self*...

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.

Re:So what?

[xachen]

If you find a codec that does 44kHz stereo, FreeSWITCH will do this. It has no hard limit in it and is variable to any rate! This is just awesome!

Define: IVR

[theGreater]

Google gives the definition of IVR as Interactive Voice Response.

So I knew what one was, I just didn't know there was a TLA for them. This inane personal revelation brought to you by the captcha "accuse".

-theGreater.

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.

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.

PING Ted Stevens

[Rob T Firefly]

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

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.

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!