|
VoIP Frequently Asked Question (FAQ) sheet
What is VoIP?
Voice over IP or VoIP is the transport of digitized speech traffic over an IP based network
instead of over a traditional telephone network. The speech may be part of a real -time
conversation or a non-real-time transaction such as voice mail. Speech is digitized by any
of a large number of standard or proprietary voice coding schemes; however, compatible
coding is required at both ends of the connection. The IP network may be the public
Internet or a private IP -based network. Voice transport service could be phone-to-phone,
computer-to-phone or computer-to-computer. Phone connections require an interface to
the Public Switched Telephone Network (PSTN) via a gateway. The presence of a
Gatekeeper is optional.
What voice compression or coding does VoIP use?
There are three main voice compression schemes used in VoIP networks. G.711 is the
same compression used on traditional wired telephone networks and used 64 kbps
coding. G.729 compresses voice to an 8 kbps rate but still delivers near toll quality voice
over most networks. G.723 compresses voice at either 6.3 kbps or 5.3 kbps, but has the
lowest quality of the three compression algorithms mentioned here.
What signaling protocol (for setting up calls) is used?
VoIP networks typically use one of two signaling methods for setting up and tearing down
calls. The most widely deployed is referred to as H.323, which is an international standard
designed to deliver multimedia services over wired networks. Multimedia services include
voice, video, and data. H.323 is related with and depends on other protocols to deliver
these services. A second protocol is SIP, which is an Internet Protocol compliant
signaling standard set by the IETF.
What is a VoIP Gateway?
In H.323, an endpoint on the network that supports real -time communication between
other endpoints or terminals that have dissimilar capabilities. This includes supporting
voice communication between terminals on a packet, e.g., Internet Protocol (IP) network
and terminals on a circuit (e.g., Public Switched Telephone Network (PSTN) network.
Gateways translate protocols between telephony and packet networks to deliver voice and
similar services.
What is a VoIP Gatekeeper?
In H.323, provides address translation, bandwidth control, and access control to a
network of VoIP terminals and gateways. The network of all elements (gateways,
gatekeepers, VoIP terminals) under control of a gatekeeper is defined as an H.323 Zone.
What are the critical factors in delivering good quality voice services on IP
networks?
There are many factors that contribute to voice quality. The three main factors are
latency, jitter, and voice coding or compression.
What is Latency?
Latency refers to the delays encountered in delivering voice packets from the originating
to terminating end of a voice call. Delays are contributed by the voice coding algorithms,
packetization of voice packets, equipment used in the delivery of these packets over an
Internet Protocol network, and ability to prioritize voice packets through such networks.
What is Jitter?
Jitter is the variations in the delays delivering voice packets from originating to terminating
ends.
What equipment is needed in a VoIP system?
Generally, a VoIP system needs a network gateway, premises gateway, and gatekeeper.
The size of business served will dictate the type of premises gateway used. For example,
small or home-based businesses could be well served by a premises gateway that
provides a couple of analog telephone ports and a LAN port to provide simultaneous voice
and data services. A medium sized business could be well served by a premises gateway
that provides up to eight analog telephony ports. These ports could be used for traditional
analog telephones or a small PBX. Larger enterprises would be better served by a
premises gateway that provides T1 / E1 services.
What types of network topologies can be supported?
There are two general topologies, point- to-point and point to multipoint. Enterprise
customers are likely to employ point-to-point topologies while service providers are likely
to employ point to multipoint. Wi-LAN Inc. can support both.
Is billing supported on VoIP systems?
RADIUS/Billing in VoIP systems varies widely. The most common system for collecting
billing information in VoIP networks is a RADIUS server. Originally developed to manage
access in dial-up networks they have evolved to collect call information, which is uploaded
to a billing system.
What is RADIUS?
The RADIUS server allows centralized administrative control over remote access to a
network service. For example, a RADIUS server will typically hold the database of user
logins and passwords, and will be queried when a user dials into an Internet Service
Provider or a corporate network. Some RADIUS servers collect usage information, which
can be used for billing purposes.
Can regular phones be used? Most premises gateways support the use of analog telephones and Fax machines.
Depending on the VoIP network equipment, vertical features can be provided;
conferencing, call forwarding, call waiting, calling number ID, among others.
Can VoIP phones be used?
VoIP phones, although significantly more expensive, are supported on VoIP networks
using WiLAN radios.
Can I send a Fax over a VoIP system?
Wi-LAN VoIP networks support Group 3 Fax over Internet Protocol services. To do this,
both the premises and network Gateways need to support T.38 protocols. T.38 defines an
Internet facsimile protocol consisting of messages and data exchanged between Facsimile
gateways connected via an IP network. Facsimile gateways must exist at both ends of the
IP network.
What is Packet Saver Technology?
PacketSaver™ Technology saves bandwidth, and bandwidth can be an expensive. By
configuring the Gateways with PacketSaver, many (up to 30) IP phone calls share the
same IP Packet Header (which is the largest chunk of a VoIP call). By combining
PacketSaver and other integrated technologies, you can reduce the amount of IP
Bandwidth used by over 50%! At 100% capacity on an E1 trunk, that means using
192Kbps instead of 397Kbps (Kilobits per second).
Are separate Gateway and Gatekeeper needed for Wi-LAN’s VoIP networks?
Most other VoIP Gateways are only equipped to turn voice signal into IP Packets and
start the call setup process. Wi-LAN’s VoIP solutions don’t stop there. Integrated within
every Gateway is the ability to run Gateway, Gatekeeper and Border Element processes.
The Gatekeeper’s function is to match together Phone Numbers on your network with the
coinciding IP addresses of where each Phone Number terminates. The Border Element’s
function is to communicate with each Gatekeeper and to furnish updates of the matching
Phone Numbers & IP Addresses to all of other Gatekeepers dynamically as changes are
made.
What are the differences between FXO and FXS ports?
A FXS port is a transmission equipment interface that emulates the line-side interface of a
switching system such as a PSTN Central Office and can be connected directly to a
telephone set. A FXO port is a transmission equipment interface that emulates subscriber
equipment such as a telephone or a key system and passes standard signaling indications
to a remote location. Premises Gateways provide FXS ports to which the end user
connects an analog telephone or key telephone system.
What is VoIP?
Voice over IP or VoIP is the transport of digitized speech traffic over an IP based network instead of over a traditional telephone network. The speech may be part of a real -time conversation or a non-real-time transaction such as voice mail. Speech is digitized by any of a large number of standard or proprietary voice coding schemes; however, compatible coding is required at both ends of the connection. The IP network may be the public Internet or a private IP -based network. Voice transport service could be phone-to-phone, computer-to-phone or computer-to-computer. Phone connections require an interface to the Public Switched Telephone Network (PSTN) via a gateway.
What equipment is needed in a VoIP system?
Generally, a VoIP system needs on premises IP-PBX. The size of business served will dictate the type of premises IP-PBX used. For example, small or home-based businesses could be well served by equipment that provides a couple of analog telephone ports and a LAN port to provide simultaneous voice and data services. A medium sized business could be well served by a premises IP-PBX that provides up to eight analog telephony ports. These ports could be used for traditional analog telephones. Larger enterprises would be better served by a premises IP-PBX that provides T1 / E1 services.
What types of network topologies can be supported?
There are two general topologies, point- to-point and point to multipoint. Enterprise customers are likely to employ point-to-point topologies while service providers are likely to employ point to multipoint.
Can VoIP phones be used?
VoIP phones, although slightly more expensive, are supported on LAN networks using ethernet connections.
Can I send a Fax over a VoIP system?
Wi-LAN VoIP networks support Group 3 Fax over Internet Protocol services. To do this,both the premises and network Gateways need to support T.38 protocols. T.38 defines an Internet facsimile protocol consisting of messages and data exchanged between Facsimile gateways connected via an IP network. Facsimile gateways must exist at both ends of the IP network.
What voice compression or coding does VoIP use?
There are three main voice compression schemes used in VoIP networks. G.711 is thesame compression used on traditional wired telephone networks and used 64 kbps coding. G.729 compresses voice to an 8 kbps rate but still delivers near toll quality voice over most networks. G.723 compresses voice at either 6.3 kbps or 5.3 kbps, but has the lowest quality of the three compression algorithms mentioned here.
What is Latency?
Latency refers to the delays encountered in delivering voice packets from the originatingto terminating end of a voice call. Delays are contributed by the voice coding algorithms, packetization of voice packets, equipment used in the delivery of these packets over an Internet Protocol network, and ability to prioritize voice packets through such networks.
What is Jitter?
Jitter is the variations in the delays delivering voice packets from originating to terminatingends.
What are the differences between FXO and FXS ports?
A FXS port is a transmission equipment interface that emulates the line-side interface of aswitching system such as a PSTN Central Office and can be connected directly to a telephone set. A FXO port is a transmission equipment interface that emulates subscriber equipment such as a telephone or a key system and passes standard signaling indications to a remote location. Premises IP-PBX provides FXS ports to which the end user connects an analog telephone or key telephone system.
|
FAQ