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VoIP Calls Double In Quality

Posted by Hemos on from the sounding-better-and-better dept.

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.

4 of 116 comments (clear)

  1. Please get the rest of the telcomms to follow. by rob_squared · · Score: 3, Informative

    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*

    --
    I don't get it.
  2. Define: IVR by theGreater · · Score: 3, Informative

    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.

  3. Re:Doubling? hardly by jdmicklos · · Score: 5, Informative

    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.

    --
    -Jon
  4. Only a slight improvement by riflemann · · Score: 4, Informative

    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.

    --
    Sparks:Gadget:Beer Maker