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with analog trunked radio voice networks and with Police Dispatch Centers using radio-over-IP (RoIP)
VoIP deployments require the provisioning of explicit priority servicing for VoIP (bearer stream) traffic
and a guaranteed bandwidth service for call-signaling traffic. Voice quality is directly affected by all three
Quality of Service (QoS) factors: packet loss, latency (delay), and jitter (the variance in latency).
For IP voice traffic to be transmitted over an Ethernet LAN, the QoS recommendations for voice service
Voice traffic should be marked per IETF RFC 3246.
Loss (dropped packets) should be no more than 1 percent.
One-way latency (mouth to ear) should be no more than 150 msec, although in many situations 200
msec latency should be tested for acceptability.
Average one-way jitter should be targeted at less than 30 msec.
A range of up to 320 kbps of guaranteed priority bandwidth should be provisioned per voice call
(depending on the sampling rate, the VoIP codec, and layer 2 media overhead).
For high bandwidth applications, and especially when video imagery is transmitted over a network, the
system design should include means for prioritizing traffic and controlling utilization to prevent the network
from being overwhelmed.
Multicasting is an essential means because it enables traffic to be sent to a group of identified users
simultaneously using a single transmission. The alternative, unicasting, requires that a server discretely
transmit individual video feeds to each client.
IP video multicasting reduces the bandwidth required on the backbone or core of the network. As multicast
data travels through the network, it continues to propagate; for this reason, the load from multicasting is
felt primarily at the edge, which scales in capacity as more users are added to the network. The ability to
source a single stream that can service hundreds and even thousands of users significantly decreases
infrastructure and operating costs for content providers, and is essential for scaling video distribution.
In computer networking, multicast is the delivery of a message or information to a group of destination
computers simultaneously in a single transmission from the source. Copies are automatically created in
other network elements, such as routers, but only when the topology of the network requires it.
Multicast is most commonly implemented in IP multicast, which is often employed in Internet Protocol (IP)
applications of streaming media and Internet television. In IP multicast the implementation of the multicast
concept occurs at the IP routing level, where routers create optimal distribution paths for datagrams sent to
a multicast destination address.
At the Data Link Layer, multicast describes one-to-many distribution such as Ethernet multicast addressing,
Asynchronous Transfer Mode (ATM), point-to-multipoint virtual circuits (P2MP) or Infiniband multicast.
IP multicast is a technique for one-to-many communication over an IP infrastructure in a network. It scales
to a larger receiver population by not requiring prior knowledge of who or how many receivers there are.
Multicast uses network infrastructure efficiently by requiring the source to send a packet only once, even
if it needs to be delivered to a large number of receivers. The nodes in the network take care of replicating
the packet to reach multiple receivers only when necessary.
The most common transport layer protocol to use multicast addressing is User Datagram Protocol (UDP).
By its nature, UDP is not reliable—messages may be lost or delivered out of order. Reliable multicast
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