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Network Requirements
Internet Group Management Protocol
IGMP (Internet Group Management Protocol) can be used to control where multicast is allowed to propagate. When a console
on the subnet is expected to be continually operational, multicast must be active for that subnet at all times.
Network Performance
Networks should perform well under any loading conditions. The default audio delay is 120ms, plus any delay added by the
network. While delay alone does not cause issues, variable delay (jitter) does. Jitter in a network cannot exceed the
maximum packet buffer of any individual product buffer. For example, the IP-223 can handle approximately 600ms of
network jitter.
REFERENCE:Refer to the individual product manuals for these specifications.
NOTE: Losing more than 5% of the total packets transmitted compromises audio quality and system performance.
Optimally, packet loss should be less than 1%.
TCP/IP and UDP/IP
TCP/IP (Transmission Control Protocol/Internet Protocol) is the best-known protocol for use in computer communications. It
is the basis for communications on the Internet and World Wide Web. It is a guaranteed method of transferring data between
two (2) computers. Being guaranteed means for every packet of information transferred from one (1)computer to another an
acknowledgement packet is returned. Additional handshaking is utilized from the outset of the data communications to ensure
both ends of the connection. Because of this guaranteed communications and its implementation utilizing handshaking (no
other method is available), TCP/IP adds a great deal of overhead to data communications is not desirable for audio traffic over
a network. This is where UDP/IP finds its acceptance.
UDP/IP (Universal Datagram Protocol/Internet Protocol) has its existed as long as TCP/IP as an unreliable method of data
communications. The term unreliable should not be thought of as a problem for audio communications over a network
connection. UDP allows for a computer to send a packet of data to another computer without the handshaking sequence
required within TCP/IP. Because of this, the computer sending the packet has no confirmation the packet arrived at its
destination. While the loss of packets can be a problem, it generally is accounted for when the UDP application is developed.
In the case of VoIP, the loss of a packet, which only contains 10-40ms of audio, is not a problem, as the human ear generally
ignores the small chunk of lost audio. The largest single factor in the loss of UDP/IP packets is network design and loading.
UDP applications use algorithms which makes the loss of information the largest single factor in UDP/IP network design and
loading. As long as a network is well designed with capacity for all of its chartered requirements, packet loss can be a
non-issue. Because of its lower overhead and its ability to Multicast, UDP/IP is the protocol of choice for VoIP development.
Multicast UDP/IP
Multicast is an extension to UDP/IP. It enables one (1) computer to broadcast data packets to multiple recipients. This is an
ideal model for radio communications when multiple people need to monitor the audio. A single VoIP connected radio is setup
to broadcast Multicast VoIP packets when receiving audio. Since the Multicast packets can be received by any interested party,
all consoles monitoring the audio can receive and decode the packets for playback. In addition to simplifying monitoring of
audio traffic by multiple listeners, Multicast also greatly reduces the bandwidth requirement on the network. Instead of having
to regenerate the received audio into a UDP/IP data stream to each individual monitor, which uses the bandwidth times the
number of monitoring consoles, a single data stream is generated and monitored by all.
Implementation of a Multicast protocol requires a few things for seamless use on a network. First, clients must all support the
protocol. This is accepted as given since all Telex products utilize Multicast for audio communications. Second, consider if the
network infrastructure supports Multicast.
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