Re: [OT] skype
>
>
>
>---- Original Message ----
>From: tzafrir@cohens.org.il
>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" <dotancohen@gmail.com> 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 | tzafrir@jabber.org | VIM is
>>http://tzafrir.org.il | | a Mutt's
>>tzafrir@cohens.org.il | | 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
>>
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