| Note: Flyfone VoIP
refers to FlyFone International,
a privately held company based in Los Angeles, CA, is an international
provider of Voice-over-IP (VoIP) services. Flyfone.com does not have a
website, so they may not be in business anymore. (Nov-2008)
VoIP.com is a site to get VoIP
service.
Voice over Internet Protocol (VoIP)
is a general term for a family of transmission technologies for delivery
of voice communications over the Internet or other packet-switched
networks.
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Online VoIP
Conventional phones are connected directly to telephone company phone
lines, which in the event of a power failure are kept functioning by
backup generators or batteries located at the telephone exchange.
However, IP Phones and the IP
infrastructure connect to (routers and servers), which typically
depend on the availability of mains electricity or another locally
generated power source. Therefore, most VoIP networks and the
supporting routers and servers are also on widely available and
relatively inexpensive uninterrupted power supply (UPS) systems to
maintain electricity during a power outage for a predetermined length
of time.
The amount of time typically ranges from
as little as an hour and up from there, depending on the quality of
the UPS unit and the power draw and characteristics of the
communications equipment.
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Voice travels over
the Internet in packets in the same manner as data. So when you talk over
an IP network your conversation is broken up into small packets. These
voice and data packets travel over the same network with a fixed
bandwidth. This system is more prone to congestion and DoS attacks than
traditional circuit switched systems.
Other terms frequently encountered and synonymous with VoIP are IP
telephony and Internet telephony, as well as voice over broadband,
broadband telephony, and broadband phone, when the network connectivity is
available over broadband Internet access.
VoIP systems usually interface with the traditional public switched
telephone network (PSTN) to allow for transparent phone communications
worldwide.
VoIP can be a benefit for reducing communication and infrastructure costs
by routing phone calls over existing data networks and avoiding duplicate
network systems. Skype and Vonage are notable service provider examples
that have achieved widespread user and customer acceptance and market
penetration.
Voice-over-IP systems carry telephony speech as digital audio, typically
reduced in data rate using speech data compression techniques, packetized
in small units of typically tens of milliseconds of speech, and
encapsulated in a packet stream over IP.
Some broadband
connections may have less than desirable quality. Where IP packets are
lost or delayed at any point in the network between VoIP users, there will
be a momentary drop-out of voice. This is more noticeable in highly
congested networks and/or where there are long distances between end
points. Technology has improved the reliability and voice quality over
time and will continue to improve VoIP performance.
It has been suggested to rely on the packetized nature of media in VoIP
communications and transmit the stream of packets from the source
phone to the destination phone simultaneously across different routes
(multi-path routing). In such a way, temporary failures have less impact
on the communication quality. In capillary routing it has been suggested
to use at the packet level Fountain codes or particularly raptor codes for
transmitting extra redundant packets making the communication more
reliable.
A number of protocols have been defined to support the reporting of QoS/QoE
for VoIP calls. These include RTCP XR (RFC3611), SIP RTCP Summary Reports,
H.460.9 Annex B (for H.323), H.248.30 and MGCP extensions. The RFC3611
VoIP Metrics block is generated by an IP phone or gateway during a live
call and contains information on packet loss rate, packet discard rate
(due to jitter), packet loss/discard burst metrics (burst length/density,
gap length/density), network delay, end system delay, signal / noise /
echo level, MOS scores and R factors and configuration information related
to the jitter buffer.
RFC3611 VoIP metrics reports are exchanged between IP endpoints on an
occasional basis during a call, and an end of call message sent via SIP
RTCP Summary Report or one of the other signaling protocol extensions.
RFC3611 VoIP metrics reports are intended to support real time feedback
related to QoS problems, the exchange of information between the endpoints
for improved call quality calculation and a variety of other applications.
About Internet VoIP |
Flyfone VoIP | Flyfone.com | VoIP Service Providers | VoIP.com | wwwVoIP
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