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
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.