PC to Phone (PC-Phone) VoIP technology refers to using your Internet-
connected computer to make voice conversations to virtually any conventional
telephone in the world (cellular, land-line, sat phone, etc.).By doing so, you pay
only a fraction of what the traditional telephone companies charge for similar
services (i.e., Phone to Phone). Regardless of whether you access the Internet
through a high speed connection (such as cable or ADSL) or through a
conventional dial-up connection, PC-Phone technology can save you a
significant amount of money on your long-distance telephone calls. VoIP calls are
routed through the Internet in a method known as 'packet switching'. Essentially,
packet switching of voice transmissions enables the carriage of multiple
conversations using the same amount of bandwidth that a typical 'circuit
switched' call would use.
In a typical PC to Phone call, voice input is received by your personal
computer (using a microphone), translated into a data 'packet', then routed from
your computer through the Internet to as close as possible to the destination you
are trying to call. Once your 'packet' of data reaches the destination locality, it is
switched back to the local 'traditional' telephone network and routed to the
telephone of the person whom you are calling. Routing and switching of your PC
to Phone call takes place transparently and quickly. In fact, the chances are high
that the person whom you are calling would have no idea that you are talking to
them using your computer as the voice quality and data transmission times are
very similar to those achieved by circuit switched networks.
TABLE OF CONTENTS:
Content Page No.
2. Review of relevant literature………………………………………………………..4
3. Project Design and process………………………………………………………...6
b. Encoding….. …………………………………………………………………..11
d. Gateway Control………………………………………………………………13
4. Management plan and Organization……………………………………………..13
1. a. Purpose:
The purpose of our project is:
• To make a call to a phone with the help of a PC in a cost effective manner
• To better understand the practical usage of Voice over Internet Protocol
1. b. Goals:
The goal of our VoIP implementation is to achieve:
• Significant savings in network maintenance and operation costs
• Send voice transmissions over IP-based data network
• Provide the efficiency of a packet-switched network while rivaling the
quality of a circuit-switched network
1. c. Objective:
The objective is to utilize open, flexible and distributed implementation of
PSTN-type services using IP based signaling, routing, protocol and interface
technologies. To achieve this, it is necessary to change the mindset of those
responsible for the design and operations of traditional voice services networks.
Furthermore, one has to be ready to face the challenging problems of achieving
reliability, availability, quality of service.
2. Review of relevant literature:
The following are some of the research work done in this area of VoIP:
1) “Voice over IP Signaling: H.323 and Beyond” Hong Liu and Petros Mouchtaris,
Telcordia Technologies IEEE communications, October 2000
This paper deals with Signaling, which has been one of the key areas of
Voice over IP (VoIP) technologies since inception. H.323 was the key protocol
that allowed interoperability of VoIP products and moved the industry away from
the initial proprietary solutions. Once the VoIP industry started maturing, some
limitations of H.323 came to light. In this article we provide an overview of H.323,
describe its capabilities, and discuss how its limitations are being addressed
using the concept of gateway decomposition. We also discuss how H.323 can
coexist with other protocols such as MGCP, H.248, and SIP which are attracting
a lot of interest in the VoIP industry today.
2) “Implementation of QoS-Provisioning system for voice over IP”
Shengquan Wang; Zhibin Mai; Magnussen, W.; Dong Xuan; Wei Zhao
Real-Time and Embedded Technology and Applications Symposium, 2002.
Proceedings.Eighth IEEE Volume, Issue, 2002 Page(s): 266 - 275
Digital Object Identifier 10.1109/RTTAS.2002.1137402
This paper addresses issues related to implementing Voice-over-IP (VoIP)
services in packet switching networks. VoIP has been identified as a critical real-
time application in the network QoS research community and has been
implemented in commercial products. To provide competent quality of service for
VoIP, the call admission control (CAC) mechanism has to be introduced to
prevent packet losing and over-queuing. Several well-designed CAC
mechanisms, such as the Site- Utilization-Based CAC and the link-utilization-
based CAC mechanisms are in place. However, the existing commercial VoIP
systems have not been able to adequately apply and support these CAC
mechanisms, and hence unable to provide QoS guarantees to VoIP. They have
designed and implemented a QoS-Provisioning system that can be seamlessly
integrated to the existing VoIP system to overcome its weakness in offering QoS
guarantees. As a result, the system has been realized at Internet2 Voice over IP
Testbed in Texas A&M University.
3) ”Voice over IP - considerations for a next generation architecture
Hillenbrand”, M. Gotze, J. Muller, P. Dept. of Comput. Sci., Univ. of
Voice over IP is on its way to becoming an alternative to the classical
telephony system because more and more Voice over IP providers offer their
solutions on the Internet and try to increase their clientele. But current voice over
IP technology is still in its commercial and technological infancy and has not yet
proven to be worldwide accessible, usable and scalable as the classical
telephony system has proven in the past 100 years. In this paper, architecture for
a next generation Voice over IP model will be outlined and discussed. The main
focus lies on interoperability between different Voice over IP providers as well as
dependability and robustness.
4) ”A Multi-Signaling Protocol Architecture for Voice over IP Terminal”, Anoop
Kumar K. and Tanu Malhotra, Wireline Systems Group, R&D Centre, IEEE
This paper describes novel multi-signaling protocol architecture of a VoIP
endpoint in an IP packet network. The endpoint is a single channel IP phone or
gateway, but can handle H.323, SIP or MGCP calls simultaneously to/from
different endpoints. The other endpoints could be using any of the signaling
protocols. The endpoint is registered to a Media Gateway Controller, SIP server
and a H.323 gatekeeper, possibly with the same number. The details of the
design including the state machines of the different call control modules are
described. The advantage of such architecture is reduced delay in call setup
between two endpoints that otherwise use different signaling protocols. The
architecture has been implemented and verified on one of the company’s IP
Phone chip, which has an ARM7 and a DSP core. An example of a simple
handling of a second call from a different protocol endpoint, when the first call is
in progress, has also been given.
3. PROJECT DESIGN AND PROCESS:
The process involved in making a call from a PC to a phone through
internet is illustrated in the following figure.
Here, a terminal adapter now connects the handset in your house to your
broadband Internet connection. The terminal adapter acts as a translator,
converting the handset signals into VoIP signals (in other words, it takes the
analog voice and converts it to a digital signal). For example, when you lift the
handset, instead of the central office recognizing that your phone is off hook, the
terminal adapter translates it to a message sent to the broadband phone provider
that you want to place a call.
From that point, the signaling is similar to the PSTN example previously
described, except at each step, the terminal adapter is translating your phone
handset actions into digital messages that are being sent over your broadband
Internet connection to the broadband phone service provider's softswitch. The
softswitch takes care of routing your call, just like the central office would. Notice
that in this case, the call is still routed through the PSTN to the person you are
calling. This is done by a gateway between the broadband phone service and the
The overall technology requirements of an Internet Protocol (IP) telephony
solution can be split into four categories:
4) Gateway control
3. a. Signaling:
The purpose of the signaling protocol is to create and manage
connections between endpoints, as well as the calls themselves.
Some of signaling protocols are:-
2) SIP (Session Initiation Protocol) & etc.
SIP is a control (or signaling) protocol similar to HTTP. It is a protocol that
can set up and tear down any type of session. SIP call control uses SDP to
describe the details of the call (i.e., audio, video, a shared application, codec
type, size of packets, etc.). SIP uses a URI to identify a logical destination, not an
IP address. The address could be a nickname, an e-mail address (e.g.,
sip:firstname.lastname@example.org), or a telephone number. In addition to setting
up a phone call, SIP can notify users of events, such as “I am online,” “a person
entered the room,” or “e-mail has arrived.” SIP can also be used to send instant
text messages. SIP allows the easy addition of new services by third parties.
Microsoft has included a SIP stack in Windows XP, its latest desktop operating
system, and it has a definite schedule for rolling out a new .NET server API that
is the successor to the Windows 2000 server. Since SIP will support intelligent
devices that need little application support from the network as well as
unintelligent devices that need a lot of support from the network, we have an
opportunity analogous to the transition from shared computers to personal
computers. In the 1960s and 1970s, we used dumb terminals to access
applications on a mainframe computer shared by many hundreds of users.
Starting in the 1980s, we began to use sophisticated applications on a PC,
but we were also able to use the PC as a communications terminal to gain
access to applications and databases on shared computers (servers) in the
network. SIP hosts with various degrees of sophistication will perform some
functions locally while allowing us to access applications in the network. SIP is
different from H.323 in this regard. Whereas the H.323 model requires
application interaction through call control, SIP users can interact directly with
applications. SIP can be used to create new services in addition to replicating
traditional telephone services. Presence and instant messaging is an example of
a new type of service that can use SIP. There are several popular instant-
messaging systems that allow users to create buddy lists and convey status to
other member of the buddy list. Status messages can show that one is talking on
the phone, or in an important meeting, out to lunch, or available to talk. The
members of the buddy list can use these “presence” status messages to choose
an appropriate time to make a phone call, rather than interrupting at an
inopportune time. Several leading suppliers of instant messaging software have
committed to converting their systems to the use of SIP.
Table describes some of the types of services that can be offered using
SIP. Using a client–server model, SIP defines logical entities that may be
implemented separately or together in the same product. Clients send SIP
requests, whereas servers accept SIP requests, execute the requested methods,
The SIP specification defines six request methods:
• REGISTER allows either the user or a third party to register contact information
with a SIP server.
• INVITE initiates the call signaling sequence.
• ACK and CANCEL support session setup.
• BYE terminates a session.
• OPTIONS queries a server about its capabilities.
SIP session setup with one proxy server
The SIP protocol is structured into four layers and has six categories of
responses. Some of the important SIP functional entities are listed below.
• User agent performs the functions of both a user agent client, which
initiates a SIP request, and a user agent server, which contacts the user
when a SIP request is received and returns a response on behalf of the
• SIP proxy acts as both a SIP client and a SIP server in making SIP
requests on behalf of other SIP clients. A SIP proxy server may be either
stateful or stateless. A proxy server must be stateful to support TCP, or to
support a variety of services. However, a stateless proxy server scales
better (supports higher call volumes).
• Registrar is a SIP server that receives, authenticates and accepts
REGISTER requests from SIP clients. It may be collocated with a SIP
• Location server stores user information in a database and helps determine
where (to what IP address) to send a request. It may also be collocated
with a SIP proxy server
• Redirect server is stateless. It responds to a SIP request with an address
where the request originator can contact the desired entity directly. It does
not accept calls or initiate its own requests.
SIP supports five facets of establishing and terminating multimedia
• User location: determination of the end system to be used for
• User capabilities: determination of the media and media parameters to be
• User availability: determining the called party’s willingness to engage in
• Call setup: “ringing,” establishing call parameters at both called and calling
• Call handling: including transfer and termination of calls.
SIP can also initiate multiparty calls using a multipoint control unit MCU or a
Fully - meshed interconnection instead of a multicast. Gateways that connect
PSTN parties can also use SIP to set up calls between them. The protocol is
designed as part of the overall Internet Engineering Task Force (IETF)
multimedia data control architecture. It incorporates many protocols, for example
Resource Reservation Protocol (RSVP) and Real-Time Transport Protocol
(RTP), for proper functionality and operation.
3. b. Encoding:
Next, when the conversation commences, the analog signal produced by
the human voice needs to be encoded in a digital format suitable for transmission
across an IP network. The encoding is done using a voice coder (vocoder)-also
referred to as a codec (coding/decoding) is to use the analog signal (human
speech) and transform and compress it into digital data.
An efficient voice encoding and decoding mechanism is vital for using the
packet-switched technology. A number of factors must be taken into account
including bandwidth usage, silence compression, intellectual property, look-
ahead and frame size, resilience to loss, layered coding, and fixed-point vs.
floating point digital signal processor (DSPs). The bit-rate of available
narrowband vocoders ranges from 1.2 to 64 kbps, with an inevitable effect on the
quality of the restituted voice. There is ordinarily, but not always, a trade-off
between voice quality and bandwidth used. Using today’s most efficient vocoder
allows quasi-toll quality bandwidth usage to be as low as 5 kbps. Toll quality is
recognized as the standard of a long-distance PSTN call. As newer and more
sophisticated algorithms are developed, this bit-rate will decrease. This will
permit more samples to be squeezed more efficiently while minimally sacrificing
quality, if at all. The algorithmic delay introduced by a coding/decoding sequence
is the frame length plus the look-ahead size. A vocoder with a small frame length
has a shorter delay than one with a longer frame length, but it introduces a larger
overhead. Most implementations choose to send multiple frames per packet.
Thus, the real frame length to take into account is the sum of all frames stacked
in a single IP packet. The smaller the frame size, the more frames in an IP
packet; thereby, there is minimal influence on latency.
Since the early 1990s, the ITU has forged ahead from the 64 kbps G.711
to the more recent G.723.1 specification that consumes merely one-twelfth of
that bandwidth. This bandwidth savings commonly comes at the cost of lower
quality and robustness to hostile network environments. Given the inevitable
increase in the average user’s bandwidth over time, perhaps this effort would be
better directed at improving quality first, then addressing bandwidth.
3. c. Transport:
Once signaling and encoding occur, the IP network itself must then ensure
that the real-time conversation is transported across the available media in a
manner that produces acceptable voice quality using Real-time Transport
Protocol (RTP) and Real - Time Control Protocol (RTCP) are utilized to move the
voice packets. Media streams are packetized according to a predefined format.
RTP provides delivery monitoring of its payload types through sequencing and
time stamping. RTCP offers insight on the performance and behavior of the
media stream, such as voice stream jitter. RTP and RTCP are intended to be
independent of the signaling protocol, encoding schemes, and network layers
Typical Internet applications use TCP/IP, whereas VoIP uses
RTP/UDP/IP. Although IP is a connectionless best effort network
communications protocol, TCP is a reliable transport protocol that uses
acknowledgments and retransmission to ensure packet receipt. Used together,
TCP/IP is a reliable connection-oriented network communications protocol suite.
TCP has a rate adjustment feature that increases the transmission rate when the
network is uncongested, but quickly reduces the transmission rate when the
originating host does not receive positive acknowledgments from the destination
host. TCP/IP is not suitable for real-time communications, such as speech
transmission, because the acknowledgment/retransmission feature would lead to
excessive delays. UDP provides unreliable connectionless delivery service using
IP to transport messages between end points in an internet. RTP, used in
conjunction with UDP, provides end-to-end network transport functions for
applications transmitting real-time data, such as audio and video, over unicast
and multicast network services. RTP does not reserve resources and does not
guarantee quality of service. A companion protocol RTCP does allow monitoring
of a link, but most VoIP applications offer a continuous stream of RTP/UDP/IP
packets without regard to packet loss or delay in reaching the receiver.
3. d. Gateway control:
Finally, it may be necessary for the IP telephony system to be converted
by a gateway to another format-either for interoperation with a different IP-based
multimedia scheme or because the call is being placed onto the PSTN. The IETF
standard Media Gateway Control Protocol (MGCP) is a merger between the
Internet Protocol Device Control and the Simple Gateway Control Protocol.
4. Management plan and Organization:
Module No. Work Assigned weeks
1 Detailed study of SIP and codec 13th-20th October
2 Implementation of SIP 20thOctober-3th November
3 Implementation of codec 4th -17th November
4 Integration of modules+GUI 17th-24th November
5 Documentation and Testing 24th November – 1st
We intend to give the user which has the following features:
• A user friendly softphone
• User authentication
• Able to call either a home phone or a mobile with Internet
Some of the additional features which we may include are as follows:
• Chat application
• PC-to-PC calling
• Web cam
• Audio conferencing
6. a. Bibliography
1. Bur Goody, “Voice over Internet Protocol (VoIP)”, PROCEEDINGS OF THE
IEEE, VOL. 90, NO. 9, SEPTEMBER 2002
2. Princy Mehta ,Sanjay Udani,”Voice over Internet Protocol-sounding good on
the internet”,IEEE potentials,2001
3. Shengquan Wang, Zhibin Mai, “Design and Implementation of QoS-
Provisioning System for Voice over IP”, IEEE TRANSACTIONS ON
PARALLEL AND DISTRIBUTED SYSTEMS, VOL. 17, NO. 3, MARCH 2006
4. Bhumip Khasnabish, “Implementing Voice over IP”, Wiley publications,2003