| VoIP
Basics
When engineering a satellite network
capable of effectively handling VoIP, there are several
important elements that require consideration.
Latency: There is
approximately 280 msec of one-way propagation delay due
to the location of the Geo Stationary orbit and the speed
of light. Regardless of the satellite product, this propagation
delay must be considered and overcome. Today, numerous
overseas calls originated in the United States are actually
transmitted as VoIP over satellite calls, particularly
if provided by the smaller long distance carriers.
Jitter: Quantifies
the effects of network delays on packets arriving at the
receiver. Packets transmitted at equal intervals from
the transmitting gateway arrive at the receiving gateway
at irregular intervals. Excessive jitter has the effect
of making speech choppy and difficult to understand. Jitter
is calculated based on the inter-arrival time of successive
packets. For high-quality voice, the average inter-arrival
time at the receiver should be nearly equal to the inter-packet
gaps at the transmitter and the standard deviation should
be low. Jitter buffers (packet buffers that hold incoming
packets for a specified amount of time) are used to counteract
the effects of network fluctuations and create a smooth
packet flow at the receiving end.
Packet Loss: Packet
loss or packet corruption will cause degradation of voice
quality. Since all of the voice traffic is UDP/IP and
would not be retransmitted (like in the case of TCP/IP)
all packets would be completely lost if the packet becomes
lost or corrupted. It is extremely important to have very
low Bit Error Rates (BER) to ensure low or no corruption
or loss.
QoS and Traffic Prioritization:
Packet switched networks are subject to congestion as
typical data traffic is bursty. Congested networks can
wreak havoc on a VoIP call with delayed, dropped,or packets
that are out of sequence. It is a necessity to have QoS
and Prioritization in order to guarantee delivery of VoIP
traffic through congested links.
Compression Technologies:
There are many encoding schemes that have been
standardized for voice. The most dominant standard in
the industry is the G.729 codec. G.729 encoding requires
8Kbps of bandwidth, but because of the overhead associated
with IP/UDP/RTP headers, the actual bandwidth needed is
between 16 and 18Kbps (depending on the equipment vendor
and configuration). With Compress RTP (cRTP) the total
bandwidth requirement per call will drop to about 10Kbps.
There are other standards, such as G.723 that takes only
5.3Kbps for the voice payload.
Required bandwidth per VoIP
call: To design a network properly, one would
need to know the amount of bandwidth required per VoIP
call, the number of concurrent calls, and the duration
of the call. Bandwidth required per VoIP call will depend
the encoding standard used, header compression, and payload
size. For example, with G.729(b), payload of 40 bytes,
and no header compression, a VoIP call would take about
16Kbps of bandwidth. With header compression this would
require 10Kbps of bandwidth. |
