> > > >---- Original Message ---- >From: [EMAIL PROTECTED] >To: debian-user@lists.debian.org >Subject: Re: [OT] skype >Date: Fri, 26 Sep 2008 20:52:05 +0000 > >>On Fri, Sep 26, 2008 at 04:34:13PM +0100, Brad Rogers wrote: >>> On Fri, 26 Sep 2008 18:05:09 +0300 >>> "Dotan Cohen" <[EMAIL PROTECTED]> wrote: >>> >>> Hello Dotan, >>> >>> Ah, I said "for masochists" because it's going to be painfully >slow over >>> a DU account. Skype has (fairly) high bandwidth requirements and >>> therefore the modem may not be able to cope. The result being >dropped >>> packets, and consequently, the inability to understand what's >being >>> said. >> >>The codec Skype uses marginally fits within the bandwidth of a modem >>connection. Likewise will speex. >> >>Modem connections also have a relatively high latency. I seem to >recal >>that the hop of a modem has a delay of some 100ms. This is something >you >>might even notice. But latency isn't that a problem with VoIP >>connections. >> >>A worse problem with VoIP, that tends to accompany latency, is >jitter: >>how much different packets have different time. An ideal VoIP stream >>would have the exact latency and thus no cracks. >> >>-- >>Tzafrir Cohen | [EMAIL PROTECTED] | VIM is >>http://tzafrir.org.il | | a Mutt's >>[EMAIL PROTECTED] | | best >>ICQ# 16849754 | | friend >> >>>Most VOIP systems (I don't know about Skype specifically) have a >>>relatively large receive buffer to smooth-out the jitter. This results >>>in a fixed, but usually acceptable delay in reconstructing the speech >>>samples. OTOH if a packet out of the buffer is required before the packet >>>has arrived, most systems will interpolate (e.g. predict that the missing >>>samples are the "same" as the current or the "average" of the last >>>correctly received and the "next" samples after the missing packet). >>>Larry >> >>-- >>To UNSUBSCRIBE, email to [EMAIL PROTECTED] >>with a subject of "unsubscribe". Trouble? Contact [EMAIL PROTECTED] >ebian.org >> >> >>
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