A POTS line *should* have predictable latency *during* any particular call.
This is because they operate over circuit-switched networks where the
connection/route is established at call initiation and remains the same
throughout the duration of the call. Essentially, a defined data path is
created through the telephone network for the duration of the call. There is no
guarantee that the next call, even to the same destination, will be completed
through the exact same path so the latency is not predictable (although will
likely be very similar) from call to call.
On the Internet, which is a packet-switched network, each data packet
(generally not more than about 1,500 bytes per packet) is routed separately, so
each *packet* may take a different path to the destination. Per-packet latency
is also affected by network load and buffering in routers. This means that
latency *during* an Internet session may vary on a packet-by-packet basis and
isn't necessarily predictable within the duration of a "call". In practice
things are usually pretty consistent though. Keep in mind also that packets may
arrive out of order, and packets will likely take completely different paths
(and even different service provider networks) in each direction to/from the
50ms, BTW, is about the time it takes to travel from Chicago to Los Angeles via
typical backbone network links. I have a gigabit link at work from Southfield
(outside Detroit) to Los Angeles that is around 55ms latency round trip.
One last thing: pretty much ALL phone lines are now digital. This is especially
true for links between phone exchanges (and any 100 mile link is sure to be
between exchanges). There are very few analog trunks left in service. The
telephone network standard is for 8k samples/sec, 8 bit samples. Such circuits
are transported as 64kb/s data streams without compression. Some systems use
in-band signalling that "robs" one bit from each byte resulting in 7 bit
effective samples and a 56kb/s data stream for the analog content.
Jim, I remember the old wide-band phone lines one could special order for
remote broadcast setups. A lot of that is done through the Internet now. Crazy
new world! :-) Many event venues now provide "house bandwidth" for this
purpose. Back when I was working in a studio I remember setting up delay lines
to compensate for processing delays on other busses, now the internal DSP in
the board does it automatically. It's really amazing what some of the new gear
can do without outboard boxes!
Sent from my iPad
> On Mar 18, 2015, at 2:50 PM, Jim Brown <firstname.lastname@example.org> wrote:
>> On Wed,3/18/2015 11:06 AM, Richard (Rick) Karlquist wrote:
>> There is a very straightforward fix for this latency.
>> Simply use a POTS telephone line with a phone patch.
>> Receive audio comes in, and CW tones go out. At
>> the remote site, a tone decoder generates key closures.
>> I measured 50 ms latency on a 100 mile phone connection.
>> And you never get audio glitches.
> Good system, BUT -- don't count on that 50 msec being a constant. Telco's
> routing may change at any time. And I suspect it may be digital in their
> system. Why do I say this? Roughly 25 years ago, I had a part time gig doing
> remotes for WGN Radio for their farm show, which included a live jazz combo.
> The location rotated around their rather large listening area (50kW on 720
> kHz). We did the gig on two dialup POTS lines using Dutch system that
> dibandsplitting to put something like 100 - 2700 Hz on one line and 2700-5400
> on the other. It was not unusual for these lines to have slightly different
> latencies, and the system had the ability to delay one channel to put them in
> 73, Jim K9YC
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