delay

Posts Tagged ‘delay’

postheadericon Are You VoIP Ready? – Latency

Every IT staff member knows that when you ping something, in addition to confirming that a device is connected to the network, the reply will give you the round trip time in milliseconds from the device you are pinging. A ping is a type of ICMP packet (along with the commonly used trace route command) that you can use to determine just how much delay your packets will experience from point A to point B and back. All of the graphs you see in this article are latency based and bandwidth availability – or lack thereof – has to be extrapolated from that.
Realistically, there is no way to know how much bandwidth some one has by simply looking at their connection from the outside  unless you are
sitting in the Central office looking directly at the connection. That said, there are several things that latency will tell you. I usually try to get a week s worth of data before I m comfortable with the circuit – if there is a problem, it will likely show up within that sample.
The magic latency number I like to see when testing for VoIP usability between sites is 80ms or lower. Another term that is directly related to latency is jitter . Jitter is caused when packets leaving a source in a certain order and spacing, arrive at the destination in the same order
(usually) but with different spacing. It is essentially the difference in latency time from one packet to the next. When jitter is high – anything over 15% variance between samples – it usually points to bandwidth problem. To get an idea of the impact of high jitter, imagine the sound of an
audio CD that is played while alternating between pause and fast-forward. The garbled sound is characteristic of jitter.
If you are going to network offices within the same city, you should see ping times of around 30ms or less and jitter under 5ms. The further across the country you go, the higher the latency tends to get. As of this article, a typical ping time from Houston to Los Angeles is
between 54ms and 67ms. Surprisingly, latency from Houston to Hong Kong is only 62ms to 87ms. I was in Switzerland not too long ago and the ping time was only 75ms from Bern to Los Angeles. My point is that, while geographical distance is an issue, it is not going to be the
determining factor of whether your VoIP project is going to work or not. Things that will affect whether your voice packets will get there in reasonable time or not is QOS (which stands for Quality of Service), the ISP or Carrier, bandwidth, hardware and hardware configuration.
To get an idea of what mis-configured hardware looks like, take a look at Figure 3. The IT manager had just moved into a new facility and users were complaining to him about slow Internet speed. As you can see, the latency is pretty good but the amount of dropped packets was
very high. The poor guy spent the better part of 3 weeks arguing with the carrier about the problem. Their contention was that the source of the problem was at his end as they showed everything good when they tested up to the NIU. They also ran diagnostics on the router – that
they supplied – and that also produced nothing. Technically, they were right except for one important setting that would not show up on any diagnostic. The Clock Source  setting for the router was configured as Internal  – i.e. it was referencing itself as a clock source – instead of
clocking off the network (sometimes referred to as Recovered  mode). For the most part, it worked but any time the router or carrier s clock drifted slightly, packets would be dropped.