03-01-2013, 12:53 PM
Voice over Internet Protocol: A Technical Overview
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Abstract
Voice over IP (VOIP) is popular communication technology. First of all I try to explain
what VoIP is. Then I discuss main issues associated with VoIP. Protocols and standards that exist
today to make the VOIP products from different vendors to interoperate are also discussed. The
main focus is on H.323 and SIP (Session Initiation Protocol), which are the signaling protocols.
We also discuss few advantages of VoIP.
Introduction
Voice over IP (VoIP, or voice over Internet Protocol) commonly refers to the
communication protocols, technologies, methodologies, and transmission techniques involved
in the delivery of voice communications and multimedia sessions over Internet Protocol (IP)
networks, such as the Internet. It uses the Internet Protocol (IP) to transmit voice as packets
over an IP network. As we know voice is an analog signal, here the voice signal is digitized,
compressed and converted to IP packets and then transmitted over the IP network. On the
receiving side, similar steps (usually in the reverse order) such as reception of the IP packets,
decoding of the packets and digital-to-analog conversion to reproduce the original voice stream
are executed. Basic VoIP access usually allows you to call others who are also receiving calls
over the internet. Interconnected VoIP services also allow you to make and receive calls to and
from traditional landline numbers, usually for a service fee. Some VoIP services require a
computer or a dedicated VoIP phone, while others allow you to use your landline phone to
place VoIP calls through a special adapter. One of the main motivations for Internet telephony
is the very low cost involved. Some other motivations are:
• Demand for multimedia communication.
• Demand for integration of voice and data networks.
Other terms commonly associated with VoIP are IP telephony, Internet telephony, voice
over broadband (VoBB), broadband telephony, IP communications, and broadband phone.
Internet telephony refers to the transmission of voice, fax, SMS, and voice-message via the
Internet, rather than the public switched telephone network (PSTN). Even though IP telephony
and VoIP are used interchangeably, IP telephony refers to all use of IP protocols for voice
communication by digital telephony systems, while VoIP is one technology used by IP telephony
to transport phone calls.
History
Voice over IP, a commercial realization of the experimental Network Voice Protocol
invented for the ARPANET. Network Voice Protocol (NVP) developed by Danny Cohen and
others to carry real time voice over Arpanet in 1973. In 1974 Network Voice Protocol (NVP) first
tested over Arpanet in August 1974, carrying 16k CVSD encoded voice – first implementation of
Voice over IP. First Voice over IP application (Freeware for Linux) is implemented in the year
1994.
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 as Skype have built closed networks for private user bases, offering the benefit
of free calls and convenience, while denying their users the ability to call out to other networks.
This has severely limited the ability 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 complete departure from the architecture of the legacy networks.
These solutions typically allow arbitrary and dynamic interconnection between any two
domains on the Internet whenever a user wishes to place a call.
Main Issues
For VOIP to become popular, some key issues need to be resolved. Some of these issues
stem from the fact that IP was designed for transporting data while some issues have arisen
because the vendors are not conforming to the standards.
The key issues are discussed below
• Quality of Service (QOS): Communication on the IP network is inherently less reliable in
contrast to the circuit-switched public telephone network, as it does not provide a
network-based mechanism to ensure that data packets are not lost, and are delivered in
sequential order. Also it does not provide real time guarantees but only provides best
effort service. To ensure good quality of voice, we can use either Echo Cancellation,
Packet Prioritization (giving higher priority to voice packets) or Forward Error
Correction.
• Interoperability: In a public network environment, products from different vendors need
to operate with each other if voice over IP is to become common among users. To
achieve interoperability, standards are being devised and the most common standard
for VOIP is the H.323 standard.
• Security: This problem exists because in the Internet, anyone can capture the packets
meant for someone else. Some security can be provided by using encryption and
tunneling. The common tunneling protocol used is Layer 2 Tunneling protocol and the
common encryption mechanism used is Secure Sockets Layer (SSL). Voice over Secure IP
(VoSIP) is also used to make it secure.
• Integration with Public Switched Telephone Network (PSTN): While Internet telephony is
being introduced, it will need to work in conjunction with PSTN for a few years. We
need to make the PSTN and IP telephony network appear as a single network to the
users of this service.
Protocol
VoIP systems employ session control protocols to control the set-up and tear-down of
calls as well as audio codecs which encode speech allowing transmission over an IP network as
digital audio via an audio stream. The choice of codec varies between different
implementations of VoIP depending 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 u-law and 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 8kbit/s each way called G.729, and many others.
Multipoint Control Units:
A MCU is responsible for managing multipoint conferences and is
composed of two logical entities referred to as the Multipoint Controller (MC) and the
Multipoint Processor (MP). In more practical terms, an MCU is a conference bridge not unlike
the conference bridges used in the PSTN today. The most significant difference, however, is
that H.323 MCUs might be capable of mixing or switching video, in addition to the normal audio
mixing done by a traditional conference bridge. Some MCUs also provide multipoint data
collaboration capabilities. What this means to the end user is that, by placing a video call into
an H.323 MCU, the user might be able to see all of the other participants in the conference, not
only hear their voices.
Session Initiation Protocol (SIP)
This is the IETF’s standard for establishing VOIP connections. It is an application layer
control protocol for creating, modifying and terminating sessions with one or more
participants. The architecture of SIP is similar to that of HTTP (client-server protocol). Requests
are generated by the client and sent to the server. The server processes the requests and then
sends a response to the client. A request and the responses for that request make a
transaction. SIP has INVITE and ACK messages which define the process of opening a reliable
channel over which call control messages may be passed. SIP makes minimal assumptions
about the underlying transport protocol. This protocol itself provides reliability and does not
depend on TCP for reliability. SIP depends on the Session Description Protocol (SDP) for carrying
out the negotiation for codec identification. SIP supports session descriptions that allow
participants to agree on a set of compatible media types. It also supports user mobility by
proxying and redirecting requests to the user’s current location.