Voice Over Internet

Protocol
A VoIP Overview
 What is VoIP
• Voice Over Internet Protocol is the assembling of
voice into IP data. This data can then be transmitted
over an IP network to an addressable ( IP address)
destination
• Voice calls look like data calls
• VoIP calls are packet switched and analog calls are
circuit switched
• Packet switched is data that is can be routed through
different routes on a network to reach a destination
• Circuit switched is a connection where a physical
path is dedicated between two end points
 Why VoIP
• A major implementor of VoIP is to bypass
long distance calls by transporting calls
over an IP network
• VoIP allows communications on existing IP
networks without adding additional lines or
bandwidth
• Wireless adjuncts such as POS scanners
and wireless handsets can roam and still
be connected to the main office via IP
 How does it work?
• Analog voice is received by a CODEC (coder / de-coder)
• Complex circuitry then samples the voice and compares the voice to a
codebook. This codebook is used by both the coders and decoders.
• Once sampled and matched with the codebook the voice is now
presented as data
• Some codebooks are of different sizes and have a different fidelity
• These codebooks are “recommended standards” that copies voice
samples into different sizes
• Codec standards:
– G.711 changes voice into data but uses the same amount of space as an analog
method. Uses 64K
– G.729 Uses a large codebook but sounds good at 1/8 of G.711. Uses about 8K
– G.729A Codebook smaller than G.729
– G.729B Almost the same as G.729A but has silence suppression (most English
conversations are silent 50% of the time and silence does not need to be
transmitted)
– G.723 Very small codebook. Provides poor quality and uses 5.3-6.5K
 Voice to IP Data
• Once voice is sampled and replaced by code the voice can then be
formed into data and transported.
• The usual transport method uses the IP (Internet Protocol) standard.
• IP uses many layers associated with the OSI model to form, control,
administer, and transport data.
• Most VoIP calls use a recommended standard that covers voice, video
and data over IP networks. This standard is called H.323
• H.323 encompasses many sub-standards that control setup, signaling,
transport and terminations
• H.323 has migrated to several version with additional features and
addendum's added to each version. H.323 is a developing standard
which is continually being enhanced
• Other standards such as SIP and MGCP are also being developed but
are not widely deployed
 Voice to IP
Analog
Voice

CODEC

Compress

Create Packet

Add Headers

Digital Network
 IP to Voice
Digital Network

Process Header

Decompress

CODEC

Analog Voice
 Leading Technology?
• Not really. H.323 standard recommendations were
developed for IP network video transport in 1996
• In 1998 significant revisions were made to standardize the
use of voice codec’s
• This “standardization” prompted growth in the VoIP
developers arena and the hope for a “new” revolutionary
product
• VoIP has been around for many years in proprietary
formats with gateways and “piggy backing” on existing
data networks.
• An example of legacy VoIP products are the MICOM
Communications Corporation’s Network Integration
products.
 VoIP Quality?
• VoIP Quality of Service (QoS) factors involve latency (delay),
packet loss and jitter (deviation or displacement of data)
• VoIP can provide a better quality of service (QoS) over a private
Intranet or a managed LAN
• Multiple router hops , marginal bandwidth and security firewalls
can destroy the QoS of a VoIP call
• To maintain high QoS, VoIP LAN management requires a
Gatekeeper
• Gatekeepers give voice packets a priority over data packets.
This is needed because voice packets are not re-tried in the
event of an error on a VoIP call
• Gatekeepers perform admissions control (security), bandwidth
control and call control signaling
 VoIP and the Internet
• VoIP can be done over the Internet
• Quality will suffer depending on ISP and call packet routing
• VoIP call packets routed through different routers cause the
routers to “learn” about the route the VoIP packet must take
• This “learning” causes the routers to re-send the setup
message every time a different router is selected and causes
latency
• Latency is increased by each device that traffic must pass
through. If devices such as routers are very busy the latency
through them can adversely affect voice quality
• Internet VoIP call packets may be routed through different
routers during the call
 Internet Topology
Access Edge Router Edge Router Access
Router ISP POP/ NAP POP/ NAP Router ISP

Customer Customer
Router Router

Core Routers/
SONET
VoIP
VoIP Access Access Gateway
Gateway Routers Routers

Telephone
 VoIP and the Intranet
• An unmanaged LAN (no gatekeeper) can provide
an acceptable QoS
• Available bandwidth and data traffic must be
considered
• Reduced cost of long distance has eliminated
private voice networks
• Many companies still have private data networks
• VoIP calls would pass through “known” routers
• VoIP allows connectivity to main and branch
office applications over existing data networks
 Intranet Topology
Internet ISP
Firewall

Router
Router
VoIP Gateway VoIP Gateway
Intranet
LAN
LAN

PC
PC

PC
PC
 VoIP Today
• VoIP has been a non-starter in the enterprise (private
systems) market due to the plunge in toll costs
• A large number of voice calls today pass over VoIP
connections
• End users are unaware of this underlying transport
that they are using
• The absence QoS controls and standards deter
adoption
• Inconsistent and sometimes unreliable service limits
VoIP to private networks
 What's Next for VoIP?
• More calls can be transported using VoIP making it a
natural transport for long haul services
• Traditional voice calls use 64 kilobits
• VoIP calls using G.729 use 8 kilobits
• As the technology become more prevalent the cost will
decrease and VoIP will prevail because of economic
advantages
• More efficient networking with voice and data
• When the H.323 standard becomes more stable, Web, e-
mail, fax or phone calls can be bundled together with
seamless CTI integration.
 What’s Our Position?
• Samsung Telecommunications America believes in
adopting new and emerging technologies
• STA has implemented VoIP standards into our
products
• STA’s approach is to provide a “bridged” solution
• The “bridged” solution will provide the quality,
investment retention and features of our existing
products but will allow the ability to add-on VoIP
without wholesale system replacement