VOIP (VOICE OVER IP)
Voice over IP (VOIP) uses
the Internet Protocol (IP) to transmit voice as packets over an IP network.
Using VOIP protocols, voice communications can be achieved on any IP network
regardless of whether it is Internet, Intranet or Local Area Network (LAN).
Voice over Internet Protocol (VoIP) is a methodology and group of technologies for the
delivery of voice communications and multimedia sessions over Internet
Protocol (IP)
networks, such as the Internet. Other terms commonly associated with VoIP are IP telephony, Internet telephony, voice over broadband (VoBB), broadband telephony, IP communications, and broadband phone service.
The term Internet
telephony specifically refers
to the provisioning of communications services (voice, fax, SMS, voice-messaging)
over the public Internet,
rather than via the public switched telephone network (PSTN). The steps and principles
involved in originating VoIP telephone calls are similar to traditional digital telephony and involve signaling, channel setup,
digitization of the analog voice signals, and encoding. Instead of being
transmitted over a circuit-switched network, however, the digital information
is packetized, and transmission occurs as Internet
Protocol (IP) packets
over a packet-switched network. Such transmission
entails careful considerations about resource management different from time-division multiplexing (TDM) networks.
Early providers of voice over IP services offered business models
and technical solutions that mirrored the architecture of the legacy telephone
network. Second-generation providers, such asSkype, have built closed
networks for private user bases, offering the benefit of free calls and
convenience while potentially charging for access to other communication
networks, such as the PSTN. This has limited the freedom of users to
mix-and-match third-party hardware and software. Third-generation providers,
such as Google Talk,
have adopted the concept of federated VoIP – which is a departure from the
architecture of the legacy networks. These solutions typically allow dynamic
interconnection between users on any two domains on the Internet when a user
wishes to place a call.
VoIP systems employ session control and signaling protocols to
control the signaling, set-up, and tear-down of calls. They transport audio
streams over IP networks using special media delivery protocols that encode voice,
audio, video with audio codecs,
and video codecs as Digital audio by streaming media.
Various codecs exist that optimize the media stream based on application
requirements and network bandwidth; some implementations rely on narrowband and compressed
speech, while others support high fidelity stereo codecs. Some popular codecs
include μ-lawand a-law versions of G.711, G.722, which is a
high-fidelity codec marketed as HD Voice by Polycom,
a popular open source voice codec known as iLBC, a codec that only
uses 8 kbit/s each way called G.729, and many others.
VoIP is available on many smartphones,
personal computers, and on Internet access devices. Calls and SMS text messages
may be sent over 3G or Wi-Fi.
PSTN
and mobile network providers
It
is becoming increasingly common for telecommunications providers to use VoIP
telephony over dedicated and public IP networks to connect switching centers
and to interconnect with other telephony network providers; this is often
referred to as "IP backhaul.
PSTN integration
The Media VoIP Gateway connects the digital
media stream, so as to complete creating the path for voice as well as data
media. It includes the interface for connecting the standard PSTN networks with
the ATM and Inter Protocol networks. The Ethernet interfaces are also included
in the modern systems, which are specially designed to link calls that are
passed via the VoIP.
E.164 is a global FGFnumbering standard for
both the PSTN and PLMN. Most VoIP implementations support E.164 to
allow calls to be routed to and from VoIP subscribers and the PSTN/PLMN. VoIP implementations can also allow
other identification techniques to be used. For example, Skype allows
subscribers to choose "Skype names" (usernames) whereas SIP
implementations can use URIs similar to email addresses. Often VoIP
implementations employ methods of translating non-E.164 identifiers to E.164
numbers and vice-versa, such as the Skype-In service provided by Skype and the ENUM service in IMS and SIP.
Echo can also be an issue for PSTN
integration. Common causes of echo include impedance
mismatches in analog
circuitry and acoustic coupling of the transmit and receive signal at the
receiving end.
Number portability
Local
number portability (LNP)
and Mobile
number portability (MNP)
also impact VoIP business. In November 2007, the Federal Communications Commission in the United States released an order
extending number portability obligations to interconnected VoIP providers and
carriers that support VoIP providers. Number portability is a service that
allows a subscriber to select a new telephone carrier without requiring a new
number to be issued. Typically, it is the responsibility of the former carrier
to "map" the old number to the undisclosed number assigned by the new
carrier. This is achieved by maintaining a database of numbers. A dialed number
is initially received by the original carrier and quickly rerouted to the new
carrier. Multiple porting references must be maintained even if the subscriber
returns to the original carrier. The FCC mandates carrier compliance with these
consumer-protection stipulations.
A voice call originating in the VoIP
environment also faces challenges to reach its destination if the number is
routed to a mobile phone number on a traditional mobile carrier. VoIP has been
identified in the past as a Least
Cost Routing (LCR)
system, which is based on checking the destination of each telephone call as it
is made, and then sending the call via the network that will cost the customer
the least. This rating is subject to some debate
given the complexity of call routing created by number portability. With GSM number
portability now in place, LCR providers can no longer rely on using the network
root prefix to determine how to route a call. Instead, they must now determine
the actual network of every number before routing the call.
Therefore, VoIP solutions also need to handle
MNP when routing a voice call. In countries without a central database, like
the UK, it might be necessary to query the GSM network
about which home network a mobile phone number belongs to. As the popularity of
VoIP increases in the enterprise markets because of least
cost routing options,
it needs to provide a certain level of reliability when handling calls.
MNP checks are important to assure that this
quality of service is met. Handling MNP lookups before routing a call provides
some assurance that the voice call will actually work,
Emergency calls
A telephone connected to a land line has a direct relationship between a
telephone number and a physical location, which is maintained by the telephone
company and available to emergency responders via the national emergency
response service centers in form of emergency subscriber lists. When an
emergency call is received by a center the location is automatically determined
from its databases and displayed on the operator console.
In IP telephony, no such direct link between
location and communications end point exists. Even a provider having hardware
infrastructure, such as a DSL provider, may only know the approximate location
of the device, based on the IP address allocated to the network router and the
known service address. However, some ISPs do not track the automatic assignment
of IP addresses to customer equipment.
IP communication provides for device
mobility. For example, a residential broadband connection may be used as a link
to a virtual
private network of a
corporate entity, in which case the IP address being used for customer
communications may belong to the enterprise, not being the network address of
the residential ISP. Such off-premise
extensions may appear
as part of an upstream IP PBX. On mobile devices, e.g., a 3G handset or USB
wireless broadband adapter, the IP address has no relationship with any physical
location known to the telephony service provider, since a mobile user could be
anywhere in a region with network coverage, even roaming via another cellular
company.
At the VoIP level, a phone or gateway may
identify itself with a Session Initiation Protocol (SIP)
registrar by its account credentials. In such cases, the Internet telephony service provider (ITSP) only knows that a particular
user's equipment is active. Service providers often provide emergency response
services by agreement with the user who registers a physical location and
agrees that emergency services are only provided to that address if an
emergency number is called from the IP device.
Such emergency services are provided by VoIP
vendors in the United States by a system called Enhanced 911 (E911), based on the Wireless
Communications and Public Safety Act of 1999. The VoIP E911 emergency-calling
system associates a physical address with the calling party's telephone number.
All VoIP providers that provide access to the public switched telephone network
are required to implement E911, a service for which the subscriber may
be charged. However, end-customer participation in E911 is not mandatory and
customers may opt-out of the service.
The VoIP E911 system is based on a static
table lookup. Unlike in cellular phones, where the location of an E911 call can
be traced using assisted GPS or other methods, the VoIP E911
information is only accurate so long as subscribers, who have the legal
responsibility, are diligent in keeping their emergency address information
current.
VoIP Technology and Glossary
VoIP stands for Voice
over IP (Internet Protocol), a variety of methods for establishing two-way
multi-media communications over the Internet or other IP-based packet switched
networks. Although VoIP systems are capable of some unique functions (for
example: video conferencing, instant messaging, and multicasting), this
appendix concentrates on the ways in which VoIP can be used to replicate the
voice conversation functionality of the public switched telephone network
(PSTN).
There are several
competing approaches to implementing VoIP. Each makes use of a variety of
protocols to handle signaling, data transfer, and other tasks. To help describe
the similarities and differences between these approaches, consider the
following simplified description of a telephone call under VoIP:
- Caller picks up the phone (his terminal), hears a dial
tone and dials a destination number.
- Destination number is mapped to a
destination IP address.
- Call setup routines are invoked,
handled by signaling
protocols. Depending on the VoIP standard in use, this may involve a
device (or function) known as a Gateway,
and may also involve a Gatekeeper.
- Destination phone generates a
ring, the called party picks up the phone, and a two-way conversation is
established.
- Data is moved between the two
endpoints using a media
protocol, the Real-time Transport Protocol (RTP). A codec(coder/decoder) is
used to convert the sound of each caller's voice to digital data, then
back to analog audio signals at the other end.
- Conversation ends and the call is
torn down. Again, this involves the signaling
protocols appropriate to
the particular implementation of VoIP, along with any Gateway or Gatekeeper functions.
VoIP Protocols
Like every other aspect
of Internet communications, VoIP has evolved rapidly since its introduction in
1995, and continues to evolve today. The standards show the influence of their
creators: the traditional telecommunications players, the Internet community,
and the communications equipment manufacturers such as Cisco and 3Com.
H.323 Developed
by the International Telecommunications Union (ITU) and the Internet
Engineering Task Force (IETF)
systems, by off-loading some functions to a Call Manager.
Each of these approaches
involves the use of multiple protocols. In the sections below, we split these
software tools into three groups: Signaling protocols, Media protocols, and
Codecs. The media protocols (RTP and RTCP) are common to all types of VoIP, and
the codecs are also widely used. The principle distinction between one VoIP
setup and another is their use of signaling protocols and related devices or
functions, such as Gateways and Gatekeepers.
Signaling protocols
In VoIP communication,
the signaling that controls the conversation is distinct from the actual stream
of data carrying the voice content of the conversation. The principle families
of VoIP signaling protocols are described briefly below.
Note: The data streams of VoIP are
carried in connectionless UDP packets. Many setups use UDP for signaling also,
but some require the connection-oriented TCP instead, and few permit either TCP
or UDP for signaling.
H.323 protocols suite
H.323 is an ITU-T
standard that provides multimedia video conferencing, voice, and data
capability for use over packet-switched networks. It is the most widely
deployed VoIP protocol in enterprise and carrier markets.
- H.225.0 defines
the call signaling between endpoints and the Gatekeeper
- H.225.0 Annex G
and H.501 define the procedures and protocol for communication within and
between Peer Elements
- H.245 is the
protocol used to control establishment and closure of media channels
within the context of a call and to perform conference control
- H.460.x is a
series of version-independent extensions to the base H.323 protocol
- T.120 specifies
how to do data conferencing
- T.38 defines
how to relay fax signals
- V.150.1 defines
how to relay modem signals
- H.235 defines
security within H.323 systems
- X.680 defines
the ASN.1 syntax used by the Recommendations
- X.691 defines
the Packed Encoding Rules (PER) used to encode messages for transmission
on the network
MGCP
Media Gateway Control
Protocol is used for controlling telephony gateways from external call control
elements called media gateway controllers or call agents. A telephony gateway
is a network element that provides conversion between the audio signals carried
on telephone circuits and data packets carried over the Internet or over other
packet networks.
MEGACO (H.248)
Media Gateway Control
protocol (H.248) is used between elements of a physically decomposed multimedia
gateway. This protocol creates a general framework suitable for gateways,
multipoint control units (MCUs) and interactive voice response units (IVRs).
SGCP
Simple Gateway Control
Protocol (SGCP) is used to control telephony gateways from external call
control elements.
SIP
Session Initiation
Protocol (SIP) is used to initiate VoIP connections. SIP provides the necessary
protocol mechanisms so that the end user systems and proxy servers can provide
different services such as call forwarding, called and calling number
identification, and caller and called authentication. See IETF RFC 2543.
SKINNY (SCCP)
As a generic computing
term, "skinny" refers to a device with fewer features or functions
than the common or "fat" version of the same device. In VoIP, SKINNY
is a proprietary Cisco system intended to allow skinny clients to communicate
with H.323 VoIP systems, by placing most of the required H.323 processing
capabilities in an intervening device called a Call Manager. The skinny client
and the Call Manager use a simple messaging set called Skinny Client Control
Protocol (SCCP) to communicate with each other over TCP/IP. SKINNY systems use
a proxy for the H.225 and H.245 signalling, and use RTP/UDP/IP for audio.
Media protocols
RTP and RTCP (RFC
3550) are used to transmit media such as audio and video over IP networks.
RTP and RTCP are carried in UDP packets.
RTP
The Real-time Transport
Protocol (RTP) provides end-to-end network transport functions suitable for
applications transmitting real-time data such as audio, video or simulation
data, over multicast or unicast network services. RTP does not address resource
reservation and does not guarantee quality-of-service for real-time services.
The data transport is augmented by a control protocol (RTCP) to allow
monitoring of the data delivery in a manner scalable to large multicast
networks, and to provide minimal control and identification functionality. RTP
and RTCP are designed to be independent of the underlying transport and network
layers. The protocol supports the use of RTP-level translators and mixers.
RTCP
The RTP Control Protocol
(RTCP) is based on the periodic transmission of control packets to all
participants in the session, using the same distribution mechanism as the data
packets. The underlying protocol must provide multiplexing of the data and
control packets, for example using separate port numbers with UDP.
Codecs
A codec (coder/decoder)
handles the conversion of analog signals to digital form, and back again. VoIP
systems may use any of a wide variety of codecs for voice, video, or both. In
VoIP, the codec used is often referred to as the encoding method or the payload
type for the RTP packet. Codec designers seek to optimize among three primary
factors: the speed of the encoding/decoding operations (packetization delay),
the quality and fidelity of sound and/or video signal, and the size of the
resulting encoded data stream. In Table J.1, note that the Data Rate column refers to the compressed
(encoded) data, while the Bandwidth column describes the uncompressed
audio data equivalent delivered by the codec.
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