[Elecraft] How reliable an internet connection is needed for A K3 remote to work well?

[Per-Tore Aasestrand]

Is QoS well implemented in IPv4?

Per-Tore / LA7NO

On 18 July 2014 11:22, David Woolley <forums at david-woolley.me.uk> wrote:
> It is standard to use UDP (RTP) over VoIP for the reasons given by Iain.
> Over a corporate network, VoIP traffic should have a QoS tagging on the IP
> packets which causes routers to prioritise it. VoIP over the internet has
> always been done for cost, not quality reasons, as the whole concept behind
> IP networks is at conflict with constant rate traffic; the telephone
> industry devised ATM as a packet network for that application (although they
> are now moving to IP, because voice is no longer the dominant bandwidth user
> - but I am sure they will prioritise their voice traffic).
>
> RTP has a marker bit which indicates a safe place to dump a latency buffer's
> contents.  Conceivably setting this during tuning would be a good idea.  If
> the remote operation protocol doesn't user RTP, someone has been
> re-inventing the wheel.
>
> As someone mentioned WiFi.  It is generally accepted, in the VoIP world,
> that WiFi and VoIP don't mix because WiFi introduces additional latency.  I
> believe it also does link level retransmission which, means latency can be
> particularly bad if you don't have ideal conditions.
>
> --
> David Woolley
> Owner K2 06123
>
> [ Top quoted through list policy, not preference. ]
>
> On 16/07/14 20:35, iain macdonnell - N6ML wrote:
>
>> The flip-side is that use of a "reliable" protocol, such as TCP, which
>> detects and retransmits dropped packets, causes increasing latency
>> over time (the more packets get retransmitted, the further behind
>> "real time" you get). For something like a "real time" audio stream,
>> it generally better to just accept the packet-loss. The problem of
>> increasing latency can affect UDP too - some types of unreliable links
>> cause a sequence of packets to get queued, then all transmitted in a
>> burst.
>
>
>> It's a tricky problem area. My personal software
>> solution uses UDP and a moderately-sized buffer, and when the buffer
>> builds up to a point where the latency is more than I like, I click a
>> button to dump the contents of the buffer, and return me to
>> low-latency.
>
>
>
>
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