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*...

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

[anthm]

Yes, you are correct. The benefit comes when both ends of the call are using a 16khz device. Situations where you are connecting to the PSTN would obviously be better suited at 8khz. The media description on the pstn gateway would advertise only 8k so the client would know better than to operate at a higher frequency.

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?

[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!

Re:So what?

[cdrudge]

Yeah, but the on hold music sounds great!

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.

Re:Doubling? hardly

[slyvren]

Actually 8 bit to 16 bit is far greater than double quality. The quality essentially doubles everytime you add a bit.

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.

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!

Re:Marketing BS

[anthm]

FYI: 20ms of 16khz audio (the typical size of 1 RTP packet) encoded with the Speex Codec http://www.speex.org/ is 43 bytes. 20ms of 8khz audio encoded with the Speex Codec http://www.speex.org/ is 29 bytes which is only 1.4 times as big as it's 8khz counterpart. 20ms of 8khz g711 is 160 bytes so with speex at 16khz, you can still fit 3 calls in the same amount of bandwidth that it takes for one 8khz call. The biggest overhead in VoIP is the various headers on each RTP packet per level of encapsulation, not the size of the payload.

Because it covers almost all of the human voice

[Sycraft-fu]

Our voices don't have that wide a frequency range, there's little up in the high frequencies. A voice sample recorded at 22kHz (11kHz frequency range) is very hard to distinguish from one recorded at 44kHz (22kHz frequency range). In fact you'd need to be using a fairly good mic to really get much of the higher frequencies anyhow. 8kHz works since F1 and F2 (the frequencies of the first two peaks in the harmonic curve) fall under 4kHz for essentially all speakers. F1 and F2 are what we primarly use to determine vowel sounds and thus are what's realy relivant. Well with an increase to 16kHz you get F3 and even F4 which leads to pretty natural sound as far as most listeners are concerned. Past that, there's just not a whole lot that affects your perception of speech.

The reason for chosing 16kHz is probably simply that it's twice what you have before. Thus if you are interfacing with an old system that doesn't support it, just discard every other sample, no sample rate conversion needed (which is CPU intensive).