SOVoIP's Feasibility

Feasibility study of SOVoIP ascertains the likelihood of its success. Current technological advancement and available standalone voice solutions have rendered the current systems trivial. Expanding businesses with extra workload confronting customers complains about the speed and quality of work businesses provide. In addition to that competitors are always looking for opportunities to grab big market share. Recent Yahoo and MSN merge is such a consequence. On the contrary SOVoIP brings all the platforms, protocols and devices into a single stage. SOVoIP’s feasibility attribute can be seen as:

· Unified model such as SOVoIP brings speed and efficiency in the business process. As people from different levels of management can access the system from anywhere, with whatever communication medium they have. That influences the customer satisfaction and ensures economical feasibility of the infrastructure.

· Unlike SIP or any other VoIP architecture SOVoIP does not require any network configuration in the client end. Only the providers need to maintain the network. Moreover its firewall friendly behavior brings simplicity to the system. Furthermore, HTTP is the most adopted protocol on the Internet. Thus universality of SOAP combined with HTTP, not only ensures interoperability between protocols but also make these protocols more open to the Internet. In addition to that QoS and security measurement ensure technical feasibility of the architecture. Finally, SOVoIP is an application layer solution. Thus adoption to future IPV6 network will not cause any problem.

· SOVoIP’s feasibility attribute, time, is perfect. SOA is maturing day by day. On the other hand SOVoIP does not require any especial hardware or software. Thus its time-to-market is predicted to beat the competition.

· Low client resource consumption and ability to cope up with changes in the architecture without client upgrade pledge extendibility and scalability. Thus organizational feasibility ensured.

SOVoIP: A New VoIP Architecture

Voice Over Internet Protocol (VoIP) is one of the most prominent communications technologies today. Consumers are already taking advantage of this cost effective solution for telephony over the Internet. The introduction of VoIP has resulted in a number of VoIP specific protocols that are not interoperable. Thus the next logical step is to make these and other existing protocols on the Internet talk to each other to bring about true convergence between voice and data networks. Naturally middleware that connects different software, platforms and protocols comes into the picture. Thus, in the research lab of The University of Melbourne, Australia, we have introduced a service oriented architecture for VoIP, Service Oriented VoIP (SOVoIP), which ensures protocol convergence while addressing many critical issues related to VoIP such as: Quality of Service (QoS), Enhanced 911 (E911), Communication Assistance for Law Enforcement Act (CALEA), NAT and firewall traversal. Comparative results endorse SOVoIP over the existing VoIP architectures.

We are going to present our first research paper in the 16th International World Wide Web Conference. You can have a look at our paper for more here.

Dave Greenfield, editor of VoIP Line and Network Computing on 26th of April, 2007 in his editor's note wrote about SOVoIP:

EDITOR'S NOTE:
Is There VoIP After SIP?

A new paper being presented at the 16th International World Wide Web Conference next month will be the first presentation on a fully blown out Web service-based telephony architecture or what I've been calling a Service Oriented Telephony Architecture (SOTA).

We've talked about SOTA as the next wave in VoIP. The combination of VoIP with Web service portability and component reuse make SOTA a compelling alternative to current VoIP approaches. Most SOTAs, such as those from Avaya or BlueNote, use some SIP process or server and don't reference any sort of standardized architecture.

This paper gets rid of the SIP infrastructure completely. It relies on SOAP to transport the telephony signaling and RTP to still carry the VoIP content. This means that the architecture should avoid the SIP's difficulties around NAT traversal not to mention its complexity.

Sure, the paper has a long way to go before its authors -- Mohammed Jubaer Arif, Shanika Karunasekera, and Santosh Kulkarni -- can present it to a standard's body. It's got an even longer way after that before it has the sophistication and breadth of SIP. But if you want a picture of where VoIP is headed take a gander at this document.

- Dave Greenfield
Editor, VoIP Line
Editor, Network Computing

Middleware is in Disarray

Middleware conference 2006, took place in Melbourne, Australia. The University of Melbourne was the organizer and my supervisor Dr. Shanika Karunasekera was general chair to the conference. Thus I had a chance to look at the current status of middleware from a close distance as a local organizer.

What is Middleware?
Middleware is a software enabling technology that connects two or more applications so that they can communicate each other. Middleware hides not only the distribution but also the complexities lie in different hardware and software level. It provides uniformity in messaging and high level interfaces to the application developers so that application can be easily composed and made interoperate. Web Services, CORBA, RMI, DCOM etc. are various kind of middleware.

Middleware are here to fulfill the above mentioned goals. However if we quote from the proceeding of the conference, it says:

“Middleware is one of those topics in computer science for which it appears difficult to reach consensus on its exact meaning”

In other words, my personal observation is that middleware community yet to come out with a proper definition for middleware. That certainly instigated disarray in the middleware world. To add to this plight, vendors those implement the middleware and the standardization bodies of these middleware are the prime source of disarray. Race in order to grab the market share among the vendors have been influencing the standardization processes. As a result same middleware comes out with different standards and implementations. Thus middleware is losing its primary goal of universality. If we look closely to CORBA’s rise and fall we will realize the reality. That reality has been revealed here by Michi Henning, who was a core part of CORBA.

Right now web services are flying high. However, this report from Network Computing describes web services are also following the same path as CORBA did with their standardization. Will web services embrace the same fate as CORBA? That’s a question to be answered. Moreover, now it seems more interesting to talk about these different standards rather than CORBA vs. Web Services.

Middleware is here with noble goals. It should exist. Hope it will march forward towards its goals smoothly. But, who is going to help it?

VoIP Protocols in a Nutshell & Skype’s Hidden Story

Voice on the Internet has been introduced by different new protocols which are broadly known as Voice Over Internet Protocol (VoIP). The functionality of VoIP has been achieved by protocols such as Session Initiation Protocol (SIP), H.323 and IAX etc. Some of them are as follows:

Skype
Skype is one of the most popular VoIP clients. Skype’s clients are self-contained and create a p2p network. Skype maintains central login server for authentication. Skype’s protocol is proprietary and messaging is encrypted. Thus it does not communicate with other protocols. Skype nodes continuously maintain UDP connections to surrounding nodes. However, TCP is used for call setup. As Skype clients need to do continuous processing, best of CPU and battery available today for mobile devices are not perfect match for Skype client. Continuous data flow of Skype clients is not encouraging for the cell phone carriers as well. Skype architecture is not suitable to provide E911 services. Skype’s voice quality is admirable. However, all the Skype users contribute to Skype’s revenue generation not only by paying for SkypeOut or SkypeIn but also by offering own resources to Skype. Skype’s heavy client uses one’s resources to serve him/her and to other users, in order to overcome NAT and firewall problem. Skype clients with public IP and enough bandwidth (known as supernode) help other users for NAT and firewall traversal. Any node in the public domain can become supernode. Users do not have control over it. However, with Skype version 3.0 one can prevent a client to become supernode. Most of the users are not aware of the supernode concept. Thus Skype has no fear of supernode shortages. Even though no network maintains are required at the client end, any solution such as Skype is not a good choice for an organization. Because with Skype an organization does not own the technology, thus have very less control. For example Skype’s price hike in the middle of budget year of an organization is to concern the management.

H.323
H.323 is an umbrella recommendation from International Telecommunication Union (ITU) that covers all aspects of multimedia communication over the IP network. It is a part of the H.32x series of protocols that describes multimedia communication over other networks such as Integrated Services Digital Network (ISDN) and Public Switched Telephone Network (PSTN). Its binary encoding makes the development harder. However, binary encoding minimizes the needs of number of bits to transport over the wire. Requirement for extra configuration such as Gatekeepers and Multipoint Control Units (MCUs) makes it complex. H.323 is more suitable to interface with PSTN than to the Internet. H.323 uses TCP as transport protocol. It also lacks user traceability in case of emergency.

H.323 APIs: ooH323c, OpenH323. To my knowledge there are not enough good APIs in JAVA (One may like to try J323 Engine) for H.323.

SIP
Session Initiation Protocol (SIP) is a client-server, text based lightweight protocol that works both on UDP and TCP. SIP was developed by Internet Engineering Task Force (IETF) to setup, modify and tear down multimedia sessions over the Internet. Similar to H.323, SIP architecture requires extra hardware and software in the network such as proxy servers, redirect servers and registration servers. However, it is more affable with other Internet protocols. SIP is not a transfer protocol like HTTP, designed to carry large amounts of data. It does not define any specific mechanism for E911 service, NAT and firewall traversal. SIP's aim for switching to a p2p architecture will increase its complexity similar to Skype. However for real voice data both H.323 and SIP use Real-Time Transport Protocol (RTP). Recent industry trend shows that H.323 is losing the race with SIP.

SIP APIs: JAIN-SIP, jSIP.

IAX
Inter-Asterisk eXchange (IAX) is a protocol used by Asterisk. Asterisk is an open source Private Branch Exchange (PBX) system from Digium. IAX is now known as IAX2, with its second version. It enables VoIP connections between Asterisk servers and IAX2 clients. Even though IAX2 is not an official standard protocol, as yet, it is well known for its less bandwidth consumption. IAX2 uses in-bound data streams i.e. both signaling and media information are transmitted by the same channel. Whereas, in SIP and H.323 both signaling and media are independent of each other. That makes IAX2 NAT friendly protocol. However, that implies that the PBX needs to separate voice from signaling. IAX2 “trunking” allows one IP packet to contain information for more than one active call. Thus minimizes the use of bandwidth.

All these standalone, different voice architectures exist each with its advantages and disadvantages.

Some Important Links :

Skype vs SIP [1, 2]
SIP vs H.323 [1, 2]
SIP vs IAX [1]

PSTN’s Fear Of Losing Revenue To VoIP: It Can Be An Illusion!

It is apparent that Voice Over Internet Protocol (VoIP) is the reality of future telephony. The cost effective VoIP solution will continue to attract more people in the year to come. However, what is happening on the other side of the coin? Can the VoIP providers achieve same amount of revenue as Public Switched Telephone Network (PSTN) providers? I believe VoIP revenue can supersede the PSTN revenue.

A Case Study:

My claim is best described with a paradigm. Bangladesh, a developing country from South Asia is yet to legalize VoIP. The state owned telephony body Bangladesh Telegraph and Telephone Board (BTTB), of Bangladesh wants people to stick with the traditional telephony in order to protect their income. In the year 2005-06 their revenue earning was 1,330 crore or 13,300 million Bangladeshi taka (tk). If we look at the graph-1, we will see their revenue has been dropping considerably in the hand of excellent cellular telephony infrastructure of Bangladesh and illegal VoIP for last few years. Now if Bangladesh legalizes VoIP and BTTB embraces VoIP, what will happen to their income generation? Certainly with cheap VoIP rate it will go down with same number of users. However, there are ways to expedite the revenue which is described in the next section.

Graph 1: BTTB’s Revenue Graph.

Graph 2: BTTB’s User Graph

If we look at the graphs (1 & 2) again we will see the number of users had been increasing but the revenue was dropping. Typically, it should be the other way around. With the increasing number of users the revenue should go up. However, this simple user-revenue relationship can be used to predict that VoIP revenue will supersede the traditional telephony.

It is explicable that the number of user will increase with VoIP. If Bangladesh had 10 million VoIP users which is 7% of the current population (147,365,352)with a monthly fixed charge of 300 Bangladeshi tk, the revenue of the 2005-06 seasons for BTTB would be 36,000 million Bangladeshi tk. This is much higher than the income of BTTB’s traditional telephony in the same year. This significant increment in the revenue scale should wash away any concern of BTTB regarding Return On Investment (ROI) of VoIP infrastructure. Currently Bangladesh only has 1.07 million land phone users. However, current cellular user expansion of 22 million users is a good indication of the demand of telephony in Bangladesh.

Brazil being listed in the list of developing countries has 42,382,000 users while her total population is 188,078,227. Look at the numbers again. Quite similar to Bangladeshi population but the number of telephony users are much higher.

In the meantime, Vonage, one of the largest VoIP providers of U.S.A has earned 261.939 million US dollar according to the 2005 calendar with total user of 2 million subscribers. However, VoIP provider such as Vonage earns not only from mere telephony but also from other sources of services that can be provided with VoIP with more ease. What are the other sources of income can be?

Other Sources of Income for a Voip Provider:

A VoIP provider can be an ISP. Thus the same organization can expect revenue not only from VoIP telephony but also from other sources of ISP’s profits flow. While eyeing the uprising trend of product support and telemarketing, it is obvious that call center business is a good choice for the VoIP providers.

Beside these Internet Protocol Television (IPTV) a digital television service delivered over the Internet is becoming one of the most glittering sources of income for the Internet community thus to the VoIP providers. One of such recent boom is Joost from Skype.

Moreover, existing VoIP providers are already exploring other sources of income. For example, Vonage’s enhanced 411 dialing, it charges 99cents per call. User can come to know about weather, horoscope and many others regularly appended services from 411. As VoIP protocols can be affable with all other existing Internet protocols, VoIP provider can come out with more and more lucrative services from the Internet to the telephony. For example, few years back nobody had thought of Google’s AdSense which can fill up your pocket with other’s hits.

Finally, it is worthwhile saying advertisement can provide a huge income to the VoIP providers. If we look at the picture of the VoIP telephone set, the main difference we will notice between this particular phone set and any traditional phone set is the screen. It is perfectly suitable to show an advertisement in the screen when a user talking over the telephone. It is necessarily will not hamper the telephony quality. When the phone is not in use, the advertisement can be loaded from the central server into the local memory and can be played later. More trial and error will bring perfection to such system. If we look into the television and telephone usage of U.S.A, we will see more then 90 percent of the people are using both. Thus the telephony advertisement must make a good inroad for explosive revenue for the VoIP provider. Moreover, I see such a system is a speedy version of telemarketing.

However, some of the statistics in this writing are loosely coupled but not groundless. It is a positive indication, especially, to those countries that are yet to legalize VoIP. Obviously some other partial economical factors have played their role in the mentioned examples. In spite of all these, it is rewarding to march forward with VoIP which does not necessarily mean immediate replacement of PSTN. Moreover, local investment in Internet or VoIP infrastructure can bring global market in your doorsteps.