Problems that may affect a video
call
The path along the network between video end points,
or from your sites video system to the MCU, has a significant impact on videoconferencing
performance. Network packets do not necessarily take the shortest path from
one location to another; routers determine which path is taken. A router
examines the destination address of the packet and then calculates where
to send it. Every pass through a router is called a "hop." Because a calculation
is involved, even though it occurs at very high speed, every hop adds a bit
of delay to the total time required to transit the entire path. Excessive
network hops can degrade video performance. ITS Video Network Services
recommends that video signals take no more than six hops to ensure their
quality. A trace can determine the number of hops on your connection. If
an independent ISP is involved there can be an excessive number of hops on
your sites circuit. Usually with independent ISPs there is no way to
guarantee that you will be provided the minimum recommended number of hops.
There are five fundamental network problems that affect IP video. They are
bandwidth, packet loss, latency, jitter and policies.
Bandwidth – There must be enough space in a network path
for all of your data, video, and voice packets to get through unimpeded.
For a rough idea of scale, a typical ISDN videoconference uses around 128-384Kbps
(kilobits per second). IP-based H.323 video systems can use the same
amount of bandwidth or more. A typical IP video system can connect
at between 128 – 768 Kbps. The bandwidth required for a given videoconferencing
speed is higher on IP networks than on ISDN networks, because of the overhead
packet requirements of TCP/IP. 20% overhead is required for a video
call, so for example a 384 Kbps IP video will actually use about 460 Kbps
of bandwidth. This overhead space along with the number of video conferencing
units located at your site is important when planning your sites bandwidth
requirements.
Different clients are sensitive to discrepancies in bandwidth symmetry in
different ways depending on whether the bandwidth restriction is incoming
or outgoing. In most cases, only video frame rate is affected, though some
clients may drop the video or even the call all together.
Packet loss is when packets fail to arrive correctly,
arriving too late to be useful, or do not arrive at all. This can be
because of insufficient bandwidth along the path (when congestion occurs,
routers will drop packets), or perhaps errors in transmission. Errors occur
most commonly on wireless links such as microwave, satellite or local wireless
Ethernet. They can however also occur on copper and even fiber links. Packet
loss results in effects such as "tiling" within the video window, missing
pieces or blank areas within the video window, and/or disruptions in audio.
Latency is the time delay between an event occurring
and the remote end seeing it. Latency is introduced both by the encoding/decoding
process, and hence depends on the equipment used, and also by the time it
takes packets to traverse the network. There is little you can usually do
to change the network latency, on any large scale, beyond getting directly
involved with a carrier or a research network.
Excessive latency increases the chances of people "talking over one another"
because they don't realize that the person at the other end has started speaking
too. This is less significant in calls with less than 50ms of network latency.
It can become very troublesome in calls with more than 150ms. Another problem
is that the latency for the audio and video may be different, and hence lip
movements don’t appear synchronized with the audio. This is a function of
both the terminal and the network, and can vary dramatically — some products
try to compensate for it. You must experiment to see if it is an issue for
your applications.
Jitter is the random variation in latency due to reasons
such as competing processes running on the terminal (for example on your
desktop PC), other traffic temporarily blocking the path through routers
along the way, or even the network path changing during a videoconference.
This random variation is one of several things that can cause packets to
arrive out of order from their transmitted order. Jitter results in uneven
and unpredictable quality within a videoconference and the end station client
will try to compensate for it by buffering the traffic up to some finite
time, before playing it out to you. This increases the latency even further.
Policies are introduced by things like firewalls and
network address translation (NAT) devices that are generally used to try
to hide or protect network elements from the wider Internet. H.323 uses dynamically
allocated ports and is thus not very firewall-friendly.
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