Mobile VoIP

Mobile VoIP or 'mobile voice over Internet Protocol' is an extension of the voice over IP technology and service. It puts wings on the classic approach of VoIP.

Mobile VoIP is more than Voice over WiFi or VoWiFi. Using any broadband IP-capable wireless network connection mobile VoIP will be an application over other networks such as EVDO rev A (which is synchronously high speed - both high speed up and down), HSDPA or potentially WiMax. Mobile VoIP will enable further economic and mobility tradeoffs. For example, Voice over WiFi offers free service but is only available within the coverage area of the WiFi Access Point. High speed services from mobile operators using EVDO rev A or HSPDA with probably have better audio quality and capabilities for metropolitan-wide coverage including fast handoffs from mobile base station to another, yet it will cost more than the typical WiFi-based VoIP service.

By mid-2006, there are an estimated 70 million users of Skype - a PC to PC service for voice communications over the Internet Protocol and some 20 million users of gateway-to-gateway voice over IP services such as Vonage, and there are a billion users of mobile phone users around the world.

Each of the endpoints in any VoIP service is in various ways, a computer:

- The gateway that Vonage users plug their home phones into.

- The PC (of course) that runs the Skype client.

- and, depending on the capabilities of the specific mobile model, the mobile phone in your pocket.

Mobile VoIP will become an important service in the coming years as device manufacturers exploit more powerful processors and less costly memory to meet the users' needs for ever-more 'power in their pocket'. Smartphones in mid-2006 are capable of sending and receiving email, browse the web (albeit at low rates) and in some cases watch TV.

The challenge for the mobile operator industry is to deliver the benefits and innovations of IP without losing control of the network service. Users like the Internet to be free and high speed without extra charges for visiting specific sites versus other sites. Delivering mobile VoIP is a service that challenges the most valuable service in the telecommunications industry - voice - and threatens (or promises, depending on your views) the pace of innovation in the global communications industry.

Mobile VoIP took a significant step forward in the summer of 2006 when Nokia included not only a SIP (Session Initiation Protocol) stack but a VoIP client in their new E-series dual-mode WiFi handsets (E60, E61, E70). The E-series handsets are aimed at enterprise buyers, and significantly Nokia have announced their intention to do the same thing for consumer handsets by launching the N80 Internet edition in November 2006. In theory these handsets only require the settings to be populated into the user interface, however this process requires nearly 200 key clicks so very few unsupported users succeed in using the handsets for VoIP. The first mobile VoIP operator to launch on these handsets is the UK-based Truphone, using an OTA (Over The Air) provisioning method: users send a text to the relevant Truphone number in each country, which sends a text back containing a link which then downloads and runs a provisioning wizard.

Enterprise open source VoIP with Asterisk

A lot of people want to know if it is possible to build an enterprise grade open source VoIP solution and if there are any benefits to it. The answer is that it is not only possible, but there are special features that are exclusive Asterisk which is an open source IP telephony platform. Asterisk not only serves as an IP call signaling server (sometimes referred to as an IP PBX), but it also serves as a tradition TDM/analog PBX and seamlessly bridges the gap between the two worlds.

In Figure A, we have an enterprise class hybrid IP/TDM/analog telephony solution. Note that there is no such thing as a "pure" IP solution because there is no way to avoid analog devices and the analog telephony POTS (Plain Old Telephone Service) world. At some point a company has to support an analog Fax machine or an analog telephone port for some thing. The only question is how many analog phones versus how many IP phones do you use and who gets which phones.

In my "Open Source VoIP" illustration, the solution is made up of several key components. They are:

IP PBX (Asterisk servers)

Figure A
Voicemail and Fax servers
Ethernet to T1 Bridge
Channel banks
Ethernet Switch segments (thick tubes labeled LAN)
Analog phones
Analog faxes
CAT-5 Ethernet wiring (black)
T1 wiring (red)
Standard analog phone cabling (green)
Router/Firewall devices
IP Phones
Computers with software IP phones
PDAs with software IP phones
Telco cloud

IP PBX (Asterisk servers)
PBX is a very generic term that usually indicates some kind of central box that controls all the telephones. The term IP PBX is a very loose term and usually means there is some kind of call signaling server that sets up and negotiates IP to IP phone or IP to analog POTS (Plain Old Telephone Service) calls. The PBX also manages the phone system in general. In our particular illustration, we have two redundant Asterisk servers that act as the IP PBX solution. The server hardware for Asterisk in this scenario could be a generic "white box" 1U server costing $1000 to $2500 or a name brand server from an IBM, Dell, or HP costing $1500 to $5000. That's the beauty of Asterisk software, is that it can run on commodity x86 Intel or AMD hardware and the cost savings are enormous compared to proprietary PBX systems that cost 10 times more.
Voicemail and Fax servers
Voicemail and Fax server functionality can be rolled in to a single Asterisk server. The beauty of this approach is that a commodity server will have massive storage capacity compared to a proprietary voice mail solution. Voice mails can be emailed. There are smaller Internet based telephone companies that use Asterisk to host production voice mail today. Asterisk has the following voicemail features:
Visual Indicator for Message Waiting
Stutter Dial tone for Message Waiting
Voicemail to email
Voicemail Groups
Web Voicemail Interface
Ethernet to T1 Bridge
An Ethernet to T1 Bridge such as the RedFone foneBRIDGE can be used to link multiple servers to 4 T1 devices. Even though a slightly cheaper quad T1 PCI card can be used directly in an Asterisk server, it forces you to commit those T1 resources to that particular server. The foneBRIDGE allows you the flexibility to use those T1 resources on multiple servers and be more robust in a server failure scenario. T1 interfaces connect to the Telephone Company or Channel Banks that can break out to many analog devices. The foneBRIDGE has costs ~$2200 which costs more than a Quad T1 PCI adapter but it isn't locked to any specific server. If more than 4 T1s are desired, additional foneBRIDGE devices can be added. Asterisk's parent company Digium makes PCI T1 and E1 adapters which are natively supported in Asterisk.
Channel banks
Channel banks allow you to break a T1 line in to 24 independent FXS or FXO ports. So if you bought 1 foneBRIDGE that used up a T1 to connect to the Telco and used 3 T1 connections to link up to 3 24-port Channel Banks, you would be able to serve 72 analog telephone or fax devices. Here is a great resource that breaks down some of the Channel Bank solutions and vendors. 24 port Channel Banks range in pricing from $700 to $1500. Considering the cost of Power-Over-Ethernet, powered FXS analog ports are about half the price of a powered IP phone port. If you don't need so many analog ports and you're not interested in building redundant Asterisk servers, you can skip the foneBRIDGE and the big Channel Bank by going with a USB device like the Astribank 8.
An FXS port on a Channel Bank allows you to plug an analog phone in to it and the FXS will supply power to the phone. An FXO port on a Channel Bank allows uplink to a telephone company which is providing you an FXS port in the form of an RJ-11 jack. Either of the following connection scenarios are valid.
Phone (FXO) - (FXS) Channel bank (FXO) - (FXS) Telco
Phone (FXO) - (FXS) Telco
Ethernet Switch segments
The thick tubes illustrated in the drawing represent distinct Ethernet segments. This could be a physically separate switch or an isolated VLAN on a switch. The internal LAN segment and the VoIP segment are separated by a router/firewall device. Having the firewall is highly recommended if you don't want your phones and phone system hacked or infected.
The TDMoE (Time Division Multiplexing over Ethernet) LAN segment is for PBX-Voicemail and PBX-Channel Bank communications. In the past, TDM communications between these devices used expensive T1 cards, but this has been vastly improved by sending TDM communications over inexpensive Ethernet.
Analog phones
Analog phones are the regular phones we've been using for a hundred years. They are simple, inexpensive, and well understood. They don't have as many features as an IP phone, but they do a great basic job for many people. An analog phone has an FXO RJ-11 port that connects directly to a power-providing FXS port on a Channel Bank or provided by the Telco on the wall socket.
Analog faxes
While server based faxing has revolutionized faxing in the office place, there are times that a simple fax machine is still needed. A fax machine like the analog phone has an FXO RJ-11 port that connects directly to a power-providing FXS port on a Channel Bank or provided by the Telco on the wall socket.
CAT-5 Ethernet wiring
CAT-5, CAT-5e, or CAT-6 cabling are used for 10-Base T, 100-Base T, or even gigabit Ethernet. CAT-5 is the most generic Ethernet cabling.
T1 wiring (red)
T1 wiring is used to connect T1 ports between PBXs, Channel Banks, and the Telco. A T1 carries 24 standard telephone channels which is why a single T1 port can be split in to 24 individual FXS or FXO ports or a combination of FXO/FXS ports. A T1 connected to a Telco can carry 24 simultaneous calls.
Standard analog phone cabling
This is usually just a pair of copper cabling with RJ-11 connectors on both ends. These cables carry enough power to feed a simple analog phone. These cables are used between the Telco, the analog phone, the analog fax, and the Channel Bank.
Router/Firewall devices
A router connects multiple Ethernet segments. Having an integrated Firewall or at least a simple router ACL can limit the traffic flow between two Ethernet segments. For security reasons, it's always a good idea to restrict traffic flow to a minimum.
IP Phones
IP phones are actually special purpose dedicated computers that run VoIP software. IP phones cost anywhere from $70 for a very basic unit to $700 units that have large color LCDs for customer applications. IP phones can be powered via Ethernet with the 802.3af POE (Power over Ethernet) standard or Cisco's proprietary POE for Cisco IP phones or a separate AC Adapter. Adding POE to an Ethernet Switch costs an extra ~$50 per port over the cost of a regular Ethernet Switch. Here is a good list of vendors that offer IP phones.
Computers with software IP phones
Software based IP phones or "softphones" run on general purpose computers and are very similar in function to dedicated IP Phones. Softphone prices range from free to $50 and here is a good list of softphones.
PDAs with software IP phones
PDA based softphones are identical to PC based software phones only they run on small PDAs instead of full blown laptops and desktop computers.
Telco cloud
The Telco cloud represents any generic telephone company. Companies can connect to a Telco over a simple single-channel pair of copper cabling via the RJ-11 jack on the wall or they can trunk 24 channels over a single T1 line. This is still the most common approach in Telco connectivity.
There is a new breed of telephone companies that operate over the Internet and let you connect directly to them using a cheaper Internet connection. One unique benefit of Asterisk is the fact that it has its own IAX (Inter Asterisk Exchange) protocol which allows the efficient multiplexing of multiple VoIP channels. This allows you to multiplex over 180 VoIP conversations over a single T1 Internet or Frame Relay connection whereas a normal T1 can only carry 23 voice channels.
by George Ou

Wiretapping VoIP?

VoIP has no wires. It isn’t connected to the Public Safety Answering Point System (911) and a local operator using switchboard technology can’t trace it. It is a completely different infrastructure than PSTN- the Public Switched Telephone Network. Yet, the FCC and the FBI would like to apply the same rules to VoIP as it does to every other telephone service.

What would Alexander Graham Bell say? The monopoly is over for the telephone industry- so I thought- yet it appears that there is still a monopoly on the infrastructure of what condones a telephone system.

VoIP is Voice over Internet Protocol. It is not the same technology in any way, shape, or form as your local telephone service. Using your Internet High Speed connection and transmitting audio in the form of “packets” over your IP to another computer, VoIP user, or a standard telephone on the PSTN network conduct it.

VoIP technology has existed for some time, but is now being recognized by the mainstream as a viable alternative to standard telephone service. You may wonder why some would even want to leave behind the telephones our Great-Grandparents knew and loved and opt for these wireless replacements, yet the reasons and facts are clearly visible.

VoIP offers many benefits, features, and cost savings that are not offered with standard telephone service. For instance, Voice Mail, Call Forwarding, Call Waiting, Three Way Calling, and Call Transfer, are often included in your basic VoIP package at no additional cost. Users also appreciate the flexibility and freedom offered with using your Internet connection to not only handle your telephone needs, but your E-mail needs, Video and Web cam needs and Web application needs as well. VoIP can offer this integration and it has proved extremely beneficial to small business owners. All of these benefits have contributed to the rise and popularity of VoIP services.

However, this technology is not without its flaws. For instance, there is a big problem with the integration of VoIP technology and the Emergency 911 system. The Emergency system has been developing for many years, based upon the PSTN networks. Everyone knows that in case of an emergency, they can call 911 and help will be on the way. Yet, not was the case with many of VoIP’S early customers. Unfortunately, due to the fact that VoIP is wireless, 911 had no way of tracing VoIP calls. The US government’s response to the issue seems short of shutting down VoIP in favor of landlines. However, the FCC ruled that customers will be required to sign a statement that they acknowledge that their 911 services may be impaired due to VoIP. VoIP providers have responded by incorporating Enhanced 911 services and for allowing users to call and register their personal information with 911. Yet the fact remains, that a 911 operator cannot trace the VoIP call back to the location of the user.

Time will tell where the Emergency debate will end up. But for now, it is recommended that VoIP users maintain a landline phone or cell phone solely for the ability to have direct access to 911 if needed.

Now, 911 isn’t the only problem that Uncle Sam has with VoIP. He is also upset that he can’t “tap in” and listen in on potentially dangerous calls. How do you wiretap the wireless? Well, VoIP is the same technology, Packet Switching, which is used by E-mails, gamers who are communicating, Instant Messaging services and Video Email programs. To tap the phone line seems a little contradictory, since there isn’t a phone line, but we all understand that Uncle Sam must do what he must do.

Where these rulings will take the nature of VoIP is anyone’s guess and only the future will tell for sure. Yet we can rest assured that while VoIP continues to gather strength and become a formidable force in the telecommunications arena, that products will change, services will be enhanced, and many of the issues will be addressed and resolved.

Until then, remember to work with your service provider, register with 911, enjoy your VoIP service, and don’t take advantage of your VoIP provider by conducting illicit activities because you think they can not be traced.