Arduino Projects   |   Raspberry Pi   |   Electronic Circuits   |   AVR   |   PIC   |   8051   |   Electronic Projects

VOIP - Voice over Internet Protocol

Written By: 

Anshul Thakur

 

Written words often betray emotions. What was intended is often mistaken for something drastically obtuse. Voice conversations are different, unless the emotions and expressions betray each other of course. Nevertheless, phone calls have always been the lifeline of most people from all walks of life. They are way faster than any inland mail and give a real sense of interpersonal conversation; distances seem to reduce to zero. Despite the advances of the internet into the life of a common man, phones still hold great importance. Now what if internet, living up to its reputation of being adaptable and innovative offered phone calls and that too at rates much lower than any PSTN service provider can offer? It did, and the offering was soon a phenomenon. Voice over Internet Protocol or VoIP as it is popularly known.   
A Representational Image of Voice Over Internet Protocol or VOIP
 

Fig. 1: A Representational Image of Voice Over Internet Protocol or VOIP

What is VoIP?
VoIP is a whole gamut of technologies, protocols, techniques and methodologies applied over an IP network to facilitate voice and multimedia traffic flow. It is more of a generic term which can mean a hobbyist project on a P2P basis, or a full-fledged PSTN replacement. It involves a sequence of well laid out steps like signalling channel and media channel set up, digitization, encoding, compression, packetization and hop-by-hop transmission over packet based IP networks as independent entities before reaching the destination.
 
History
Before we get to know how VoIP works, it would be imperative to know about two key technologies that made it possible in the first place. These technologies were nothing short of phenomenon when they were invented. The first phenomenon was the ‘Telephone’, back in 1870’s. Invented by Graham Bell and Elisha Gray. The first telephone exchange came up in 1878. Almon Strowger improved upon the switching technology and gave the electromechanical Strowger exchange that bore the yoke of telecom switching for more than 100 years. A paper titled ‘A Mathematical Theory of Communication’ by Dr. Claude Shannon in 1943 introduced the concept of conveying information using binary codes. This helped AT&T create ‘Touch Tone’ dialling in 1963 which ushered a new era of digital switching. The second phenomenon was the ‘Internet’ which started off in around 1968 with ARPANET.
 
In the 70’s, time shared computer networks evolved and companies owning these networks started to rent them out during evenings and weekends when the company offices were closed. Certain online service companies like Prodigy and AOL evolved from this model and offered premium services like email through dial up by 1978. Tim Berners Lee and his group at CERN laid the foundations of the ‘www’ with ‘http’, ‘URL’ and ‘HTML’. Earlier Dr. Vincent Cerf had invented the TCP/IP, which still holds prominence in packet based networks. Then in 1995 came the first VoIP product into the market, ‘Internet Phone’. A small company named Vocaltec Inc. in Israel invented it. It was a softphone, an application software that offered PC to PC calls. By 1998, PC to Phone and Phone to Phone VoIP had come into the market. The calls were offered for free, but the dialler had to listen to advertisements before the call was processed. A few IP switch manufacturers included VoIP switching capabilities in their equipment by 1998. By 2003, VoIP calls comprised of 25% of all calls. VoIP as we know of today, has evolved into a much more mature technology.
 
Flavours offered for VoIP service:
IP Phones: It is used in enterprises and institutions as Private Branch Exchanges as a suitable replacement for EPABX systems. It still forms the least obvious form of VoIP technology.
Software VoIP: Popular software like Skype forms this segment. These have been extensively used in the past global recession to curtail call costs. These can further be of three types: Web based service, instant voice and video messenger clients and web conferencing suites.
Analog Telephone Adapter: It is used in between the customer premises equipment and the network to add the IP network functionality to Analog phones.
 

VoIP Architecture

VoIP Architecture
A generic architecture of the VoIP network and how various signalling and voice information traverses it can be depicted in the figure below.
 
 A Figure Depicting General Architecture of VOIP with Various Signaling and Voice Information
 

Fig. 2: A Figure Depicting General Architecture of VOIP with Various Signaling and Voice Information

 

VoIP has many protocols to choose from for every type of signalling involved. For example, Device Control Protocols like H.248 (more popularly known as Megaco), Media Gateway Control Protocol (MGCP), NCP, Real-time Transport Protocol (RTP); Access Service Signalling protocols like Session Initiation Protocol (SIP) and H.323; Network Service Signalling Protocols like SIP, SIP-T, CMSS, BICC etc. The most popular among these are Megaco and RTP for Device Control and SIP for Signalling. Proprietary software like Skype use P2P based protocols.
 
The main components of the contemporary VoIP systems are as highlighted below.
Call Agent/ SIP Server: It is located in the service provider’s network for call logic and control functions. It maintains call states for every call in the network. Additional functionalities like Caller ID etc. can be added and the call details necessary for billing are also provided by the same. A SIP server is similar to Call Agent, in a SIP based networks and routes the SIP requests. A SIP Client is the terminating or originating SIP server rather than the forwarding server. It can convert MGCP or H.248 protocols for call setup. These are also termed as Media Gateway Controllers, softswitches and call controllers depending on the network topology and features used.
Service Broker: Placed on the edge of the service provider’s network, it is responsible for service distribution, control and coordination between application servers, media servers and call agents.
Application Server: It provides service logic and execution for one or more applications which are otherwise not hosted on the call agents like Voice Mail etc.
Media Server: Often referred to as Announcement server, it is responsible for playing announcements, codec transcoding, tone generation and detection, IVR etc.
Signalling Gateway: It is the gateway between the call agent signalling like SIP and SS7 based PSTN networks or between different packets based carrier domains allowing communication between various service providers.
Trunking Gateway: Gateway between IP networks working on H.248 or MGCP protocols and TDM based PSTN for transcoding of packet based voice into TDM networks.
Access Gateway and Subscriber Gateway: These are meant for providing compatibility support for POTS (Plain old telephone system). The difference between the two is the capacity the two can handle with the latter offering support for very few subscribers.
Access Concentrator: These act as terminations for service providers over WAN links like DSLAMs for DSL links and CMTS in Cable networks.
Bandwidth Manager: Caters to providing and maintaining the required QoS pre user on a per call basis.
Edge Routers: These are responsible for routing traffic onto the carrier backbone network.
 

Advantages

Advantage over other technologies
But after having gone through a brief review of this technology, one would wonder why anyone would want to switch from the PSTN network to VoIP. Why is it touted as the next generation of telephony? The first reason is cost saving. Internet has spread to most parts of the world where PSTN has existed and with the world steadily moving towards Broadband access for all, faster internet is being made available at cheaper rates.
 
In case of laying down of a separate network for telephony and internet, the initial infrastructure costs involved are too large and the pace of development is slow. But with a unified architecture for both, there are cost saving involved for both the entrepreneur and the customer. Packet switching is considered best for bursty data, and yet it is being used for voice traffic due to the availability of large bandwidths thanks to advancements made in fields like Passive Optical Networks. 
 
VoIP offers many other advantages too. For example, additional services like call conference which are available at premium rates to customers on TDM based lines can be offered free. Since there is no dedicated path initially, the concept of geographical distances is virtually eliminated. This has drastically reduced long haul communication costs. Non-VoIP networks do not offer location independence for customers whereas to use VoIP, the only requirement is a fast internet connection and the service can be availed from any part of the world, let alone the country restrictions. Moreover, the communication backbone is unified with email support, CRM support and other web systems.
 
A Graph Diagram Illustrating Increased Volume of VOIP Traffic Among TDM Customers
 

Fig. 3: A Graph Diagram Illustrating Increased Volume of VOIP Traffic Among TDM Customers

Current & Future

Current Scenario & Future
The promise of VoIP has been alluring to most telephone service providers throughout the world. PSTN providers have started to use VoIP telephony to connect switching centers and with other networks. This is called ‘IP Backhaul’. Corporates are migrating from copper based networks to VoIP systems. In 2008, 80% of all the PABX lines laid were VoIP based. The prices to add more extensions to the exchange are much less in VoIP systems than traditional systems. By the elimination of the concept of geographical distances, BPOs have flourished. Various companies from western parts of the world have set up call-centres in exotic places like South Africa, Philippines and India to provide day and night support to the customers in the parent country. More than any technological impact, VoIP has had a cultural and societal impact. For example, setting up of call-centres in other countries has improved job opportunities in those places, but has sent a large workforce out of employment in the native country. The cultural shifts in countries where services are outsourced are also facing a cultural shift with increased exposure to other cultures.
 
VoIP holds much promise for everyone. But there are a certain issues which hinder its wide scale adoption as an equivalent to PSTN Network. The foremost is Quality of Service. Since it used packet networks in comparison to the PSTN’s circuit switched model, packet delivery and sequencing is not ensured. 3 main causes of degrading QoS are Latency, Jitter and packet loss. Routers generally work on first serve first serve basis and may introduce latencies beyond permissible limits. There are path based delays which cannot be controlled much. Like any other IP networks, it is susceptible to DoS attacks and service theft. Unlike PSTN phones, these require local mains power to be active or else the services cannot be availed. The routing of emergency calls like the 911 number to the nearest point is difficult in this case as the subscriber may be anywhere in the world using that IP. In the absence of any legal ruling in case of IP networks, the lawful interception of voice packets is a little difficult to tackle. Unlike PSTN networks where a warrant is required to eavesdrop on conversation, the case is not the same in VoIP. In countries where regulation is weak, VoIP is either restricted like in India (where retail of only long distance VoIP is allowed and gateways may not be established in the country), or is completely banned, like in Ethiopia. Routing to other networks faces problems with features like number portability in place. Another issue is NAT and Firewall traversal for packet delivery.
 
However, many of these problems can be rectified to a certain extent, like using buffers to avoid jitter and prioritizing voice packets as delay-sensitive to reduce latencies. VoIP networks have been developed taking in the good points from both Asynchronous Transfer Mode (ATM) networks and IP networks and employ Virtual Circuit Network Numbers to route the packets in a sequence over the hops. This takes care of sequencing and prioritizing of packets and also reduces the overhead costs in bytes that would have to be used in totally IP based systems. Further a number of protocols are in place for QoS reporting in real-time like RTCP Extended Report, SIP RTCP summary reports which are exchanged occasionally to support active feedback. Security over VoIP is being upholstered using techniques like Secure VoIP (SVoIP) or Voice over Secure IP (VoSIP). The trend is catching in and fast. Now many smartphones have started to support VoIP services based on SIP protocols. As the winds flowing in favour of VoIP, it can be easily said that it is the direction of the future.

 

Comments

This article is very very good...

the article is very good to read.... and my request is that i have planned to my final year engineering project on voip phones .. can you suggest some ideas for that ... and help me to done the project that i prefer to do .. mobile phones is my area of interest and voip phones is my field of intrest..