Tuesday 23 September 2008

Asterisk for OpenWRT

Asterisk for OpenWRT

last updated: september 19, 2007

This page contains binaries and sources for running the Asterisk PBX software on OpenWRT devices.

This page has become obsolete since the majority of these patches were merged and improved in the official OpenWRT repositories since version 1.4.11. All requests and bug reports should now be propagated through tickets in the OpenWRT development process. People are strongly advised to use the official repositories for further Asterisk development an installations since there will be no new releases or developement through this site.

The original repositories for version 1.4.9 are still maintained because of some extra features that have not been merged yet, but will be in the near future.
(obsolete) Asterisk 1.4.9
Asterisk 1.4 packages can be obtained through a package repository by adding a new line to your /etc/ipkg.conf:

For OpenWRT WhiteRussian 0.9 on mipsel/brcm (Broadcom and compatibles):
src asterisk14 http://members.home.nl/hans.zandbelt/openwrt/whiterussian/packages/asterisk-1.4

For OpenWRT Kamikaze 7.06 on mipsel/brcm (Broadcom and compatibles):
src asterisk14 http://members.home.nl/hans.zandbelt/openwrt/kamikaze/packages/asterisk-1.4

For OpenWRT Kamikaze 7.06 on mips/atheros (Fonera and compatibles):
src asterisk14 http://members.home.nl/hans.zandbelt/openwrt/kamikaze/packages/asterisk-1.4/atheros

For OpenWRT Kamikaze 7.06 on x86:
src asterisk14 http://members.home.nl/hans.zandbelt/openwrt/kamikaze/packages/asterisk-1.4/x86

NB1: The 1.4.x packages are named "asterisk14-*" to allow them to co-exist with the 1.2.x packages, *BUT* the 1.4 packages _do_ use the same installation directories so installing them next to 1.2 must be done with a different "-d" flag to "ipkg install".
NB2: For the Kamikaze releases based on kernel 2.6 the zaptel/ztdummy kernel module is not yet available thus app_meetme and IAX2 trunking are not yet usable on that platform.

Update: Asterisk core updated to version 1.4.9; added asterisk-addons 1.4.2 zaptel-libtonezone updated to version 1.4.4.
asterisk14-addons includes backported-from-trunk chan_mobile (svn 384)
asterisk14-app-fax package is included, including app_rxfax and app_txfax.
asterisk14-chan-gtalk is included, with stability fixes, providing GTalk support for Asterisk.
(obsolete) Building from source
You can build Asterisk 1.4.9 and addons for OpenWRT yourself, together with all supported modules, using the buildroot environment provided by the OpenWRT team, called the SDK. The script for doing so (download) :

#!/bin/sh

ZAPTEL="1.4.4"
IKSEMEL="1.2"
ASTERISK="1.4.9"
ADDONS="1.4.2"

URL="http://zandbelt.dyndns.org/asterisk"
PREFIX="openwrt-packages"

DIR="openwrt-devel"
SDK="OpenWrt-SDK-Linux-i686-1"

do_patch() {
PATCH=${PREFIX}-$1-$2.patch
wget ${URL}/${PATCH}
patch -p0 < ${PATCH}
}

mkdir ${DIR} && cd ${DIR}
wget http://downloads.openwrt.org/whiterussian/newest/${SDK}.tar.bz2
tar jxvf ${SDK}.tar.bz2
svn co https://svn.openwrt.org/openwrt/packages
do_patch zaptel ${ZAPTEL}
do_patch iksemel ${IKSEMEL}
do_patch asterisk ${ASTERISK}
do_patch asterisk-addons ${ADDONS}
cd ${SDK}/package && ln -s ../../packages/*/* . && cd ..
make package/asterisk14-compile V=99
make package/asterisk14-addons-compile V=99

After finishing successfully, packages can be found in:
openwrt-devel/OpenWrt-SDK-Linux-i686-1/bin/packages
(obsolete) Asterisk 1.2.16
NB: these packages are obsoleted by the 1.4.x ones; people are encouraged to switch to 1.4.x as OpenWRT support for 1.2.x will probably end in the near future. These packages can be obtained through a package repository by adding a new line to your /etc/ipkg.conf:

src asterisk http://members.home.nl/hans.zandbelt/openwrt/whiterussian/packages
Installing dummy Zaptel timer support (ztdummy)
This enables MeetMe audio tele-conferencing bridge support and IAX2 trunking.
It is only useful on routers with UHCI USB features such as the Asus WL-500GdL.

1. Install kmod-zaptel
2. Install kmod-usb-uhci-iv
3. Install asterisk-app-meetme
4. Remove the alternate USB kernel module (if loaded): rmmod uhci
5. Insert the required USB module required for Zaptel timer support: insmod ./usb-uhci.o
6. Insert the Zaptel module: insmod ./zaptel.o
7. Insert the dummy USB Zaptel timer module: insmod ./ztdumy.o
8. Install the Asterisk core package (or *-mini ipk for iax2-only support) from the list above.
9. For tele-conferencing: configure meetme.conf and extensions.conf and make sure the app_meetme.so is loaded.
10. For trunking: configure iax.conf.

Have fun with tele-conferencing and/or IAX2 trunking...
Sounds
Here's a package with Dutch (male) voices for Asterisk 1.x, created by Jeroen Naeff from the samples provided by Born Digital.
# asterisk-dutchmale-sounds_0.1_mipsel.ipk
Contact
Preferably send your comments to the OpenWRT forum in one of the topics on Asterisk in the section for Community Releases.
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Sunday 7 September 2008

What is VoIP?

VoIP (Voice over Internet Protocol) is simply the transmission of voice traffic over IP-based networks.

The Internet Protocol (IP) was originally designed for data networking. The success of IP in becoming a world standard for data networking has led to its adaption to voice networking.

The Economics of VoIP

VoIP has become popular largely because of the cost advantages to consumers over traditional telepone networks. Most Americans pay a flat monthly fee for local telephone calls and a per-minute charge for long-distance calls.

VoIP calls can be placed across the Internet. Most Internet connections are charged using a flat monthly fee structure.

Using the Internet connection for both data traffic and voice calls can allow consumers to get rid of one monthly payment. In addition, VoIP plans do not charge a per-minute fee for long distance.

For International calling, the monetary savings to the consumer from switching to VoIP technology can be enormous.

VoIP Telephones

There are three methods of connecting to a VoIP network:

  • Using a VoIP telephone
  • Using a "normal" telephone with a VoIP adapter
  • Using a computer with speakers and a microphone

Types of VoIP Calls

VoIP telephone calls can be placed either to other VoIP devices, or to normal telephones on the PSTN (Public Switched Telephone Network).

Calls from a VoIP device to a PSTN device are commonly called "PC-to-Phone" calls, even though the VoIP device may not be a PC.

Calls from a VoIP device to another VoIP device are commonly called "PC-to-PC" calls, even though neither device may be a PC.

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Why VoIP?

The number one reason to switch to VoIP technology for telephone service is cost reduction. From that base, VoIP is able to provide some compelling features which makes switching even more attractive.

Eliminating Phone Lines

With VoIP service, you can cancel your traditional phone service through your local telephone company and place all of your telephone calls over your broadband Interner connection.

For a residential customer, this will save around $40 a month. For business customers, the savings can be thousands of dollars a month.

Eliminating Long Distance Charges

VoIP technology can also save money on long-distance charges. Most residential and business telephone customers pay per-minute fees for long-disatance telephone calls. VoIP can reduce or eliminate those long-distance fees.

This saving is especially valuable with International calls, where per-minute charges for traditional telephone calls can be very expensive.

Number Portability

With VoIP service, you can take your phone number anywhere you go, easily. If you have a Chicago number and you move to New York, you can keep your Chicago number. This is very convenient for friends and family to keep in contact with you wherever you go.

Computer Telephony Integration (CTI)

VoIP service providers are designing and implementing new features which implement Computer Telephony Integration (CTI).

For example, VoIP customers may be able to receive their voice messages in e-mail as .WAV file attachments. This can make managing voice mail messages much easier and more powerful, because it enables recipients to archive voicemails or forward them to anyone with an email address.

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How can I get free VoIP?

The first thing to know about free VoIP calls is that none of them are actually completely free. Even in the best hypothetical case in which the VoIP provider doesn't charge you at all, remember that you still have to pay for your broadband Internet connection. You must understand that the goal is not to achieve completely free calls to all destinations, but to use the VoIP operator that suits your needs best. Keeping that in mind, you will learn that most VoIP companies will let you talk for free in their own network but also they will charge you for making calls outside their proprietary network.

The main way for free VoIP calls companies is to offer free calls inside their own network and also towards other specially selected destinations. Using this tactic users are drawn to make calls to free destinations and afterwards purchase credits to make calls towards paid destinations.

There are several ways used by VoIP companies to lure customers and make a profit in the same time. The great thing about VoIP calls is that they're very cheap, but not completely free, here are some systems used today in the VoIP calls market:

If you take Skype for example, one of the most popular VoIP services on the market, you will see that you can initiate conversations with other PC users of Skype free of charge. Of course, this is an advantage for long distance calls, as there is no fee for calls inside the Skype network. But if you want to make calls to regular landlines, you'll have to pay. The subscription fee for calls in North America is $30 per year. It's not a great deal of money but it still isn't free. You can make free phone calls on a PC to PC basis using the Skype software, and the number of users on the Skype network is continuously growing. On the other hand if you want to reach someone that doesn't have a PC or an Internet connection, you'll have to pay the required fees.

Another approach to this, could be the way Raketu is seeing things. Raketu is offering free phone calls to landlines in 42 countries and besides that, it also offers live video television. The downside to Raketu's service is that they ask you to pay $9.95 up front in order to use their free services. They say it's used as credit if you happen to call destinations that are not on the free call list, but either way you look at it; it's money that you have to pay to use the service.


If you are in the pursuit of a real cost free VoIP service you can use something like voipCheap that allows you to make free calls to PC's and regular phone lines. It also includes many destinations outside USA and Canada that can be called without paying a cent. The downside is that you have a limited number of 300 minutes that you can use each week, per IP Address. If you talk more than the included 300, you are required to pay for further calls, you are also required to pay for calls outside of the destinations listed on the free call list.

To better understand how to make free VoIP calls, let's take a look at how VoIP telephony works. Basically all you need is to setup a VoIP gateway, that's most commonly done by using a PBX. A PBX (Private Branch Exchange) is a device that allows the VoIP provider to purchase as many telephone lines, as the maximum number of simultaneous callers. In general, around 10% of the users will make calls at the same time. This means that the VoIP can purchase fewer telephone lines, instead of buying one for each user. This brings us to the point, configuring a PBX is almost cost free, the actual costs come from the prices collected by the telephone companies for connecting to the PSTN (Public Switched Telephone Network). What VoIP providers do is use PSTN connections for accessing the public land and mobile telephone lines, but at the same time uses the SIP (Session Initiation Protocol) to stream media content such as voice over the Internet.

With so many VoIP providers appearing every day the problem of interconnecting, preferably free of charge, with other VoIP providers arises. There are several ways to do this, as some of the existing VoIP providers that use SIP technology have already made peering arrangements to allow users interconnect for free. All you have to do, is use the special prefix code put at your disposal. Although this is not standard procedure, just yet, it's still widely used. The only problem with this is that some devices, such as VoIP phones cannot input the format of the prefix (the prefix usually looks similar to this: number@some_provider.com). As a solution to this, groups like SIP Broker or Voxalot, assign numeric values for your SIP URL, so it can be used on a wider scale of devices.

Although there is no such thing as completely free VoIP telephony, and with some VoIP providers there are some serious hidden costs and restrictions you have to look for, VoIP calls are still very profitable. Knowing that you can make free calls towards destinations in the same VoIP network as you are, and in VoIP networks you have peering with and also towards special destinations selected by your service provider, you can see which combination better suits your needs. If you manage to use services that offer good deals for what you require, you can save up to 98% from your phone bills. This means that you will need to put some time and research into it, but in the end you can achieve almost free calls via the VoIP technology using several and payment methods.

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How do I compare VoIP providers?

VoIP (Voice over Internet Protocol) is changing the way people communicate. VoIP utilizes a broadband internet connection for routing telephone calls, as opposed to conventional switching methods, providing efficient use of existing Internet connections as well as lowering overall costs. Interestingly, there is no need for any large scale infrastructures; just combine a conventional phone with a broadband Internet connection to utilize a single service with minimal software and hardware support.

VoIP service providers are touting unlimited local and long distance calling for as little as $199 per year. This provides customers with substantial annual savings. There are several VoIP providers offering VoIP service for both residential customers as well as business. However, from a customer's standpoint it is an ideal option to compare several VoIP providers in selecting the best deal.

VoIP Product Features

There are several VoIP providers who claim outstanding services and comprehensive features. Don't be fooled - not all VoIP services are created equal. The VoIP package includes many features that may not be available on traditional phones. The most common VoIP features include 3-way calling and call waiting. As the competition between VoIP providers escalates, some providers are offering additional features to establish branding of their business while attracting additional customers. That's why it's always a good option to compare several VoIP providers to discover the VoIP product features you will get when taking a connection from the provider.

Monthly Rates

One of the main advantages of VoIP is reduced long distance cost and inexpensive local phone service with several enhanced features conventional telephone services are ill equipped to provide. Compare various VoIP providers to know the monthly rates they charge for their service. Selecting an ideal VoIP provider will help you to save up to 75% on expected annual charges.

Using VoIP for International Calling

If you make a lot of international calls, do a bit of research to find a VoIP provider who offers outstanding international services at the best rates. International rates differ from one VoIP provider to another. There are also some carriers which offer unlimited overseas calling. Though this offer is limited to certain countries, check whether the country to which you call falls in this category.

911 Service

Today, majority of the VoIP providers offer E911 service. While selecting a VoIP provider, make sure the provider offers 911 service.

Keeping Your Number

There are many VoIP providers who allow the customers to transfer (port) their current phone number to the VoIP service. Not all VoIP providers offer this service. If you need to change your phone number in this way, then you need to do research on the various VoIP providers to discover whether they offer such services. However, before asking your VoIP provider to switch your current number to the VoIP service, it is advisable to try out the provider's service and make sure that you are satisfied with the end result.

Money Back Guarantee

As VoIP is a relatively new product, most of the VoIP providers will offer a free money back guarantee. As a customer you will be in a risk-free position if your VoIP provider is offers a money back guarantees for up to 30 days.

Comparing various VoIP providers will help you to select the one VoIP service provider whose terms and conditions meet your specific needs and calling pattern, especially if you make regular long distance or international calls.

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What are VoIP phones?

VoIP phones are telephones which connect to VoIP networks instead of to the PSTN.

  • VoIP phones with Ethernet connections
  • VoIP phones with Wi-Fi / 802.11 connections
  • VoIP phones with dialup modem connections
  • Software VoIP phones

VoIP phones with Ethernet connections

A VoIP phone with an Ethernet connection is the easiest type of VoIP telephone to use. Instead of a standard telephone RJ-11 connector to plug into the PSTN, these phones have RJ-45 connectors to plug into Ethernet networks.

The Ethernet connection is used to connect these VoIP phones to the VoIP server or VoIP gateway.

VoIP phones with Wi-Fi / 802.11 connections

Wi-Fi (802.11) VoIP phones provide the same service as Ethernet VoIP phones, but they do it wirelessly.

A Wi-Fi enabled VoIP phone connects to a VoIP server or VoIP gateway through your existing Wi-Fi network.

VoIP phones with dialup modem connections

VoIP phones with dialup modem connections are very similar to VoIP phones with Ethernet connections.

Instead of connecting to an Ethernet network, these VoIP phones dialup over the PSTN to VoIP service providers.

Using a VoIP phone with a dialup modem connection requires a regular analog POTS telephone line, but enables long-distance and international calls to be made over VoIP networks, usually at a significant savings.

Software VoIP phones

Software VoIP phones turn your PC into a VoIP telephone.

Software VoIP telephones are less expensive than the choices listed above, if you already own a personal computer.

Hardware for Software VoIP phones

Software VoIP phones use the PC's sound card, speakers or earphones, and microphone. This hardware works to emulate a telephone, even though this is not what the PC was designed for.

For better ease-of-use, many companies manufacture USB VoIP phones. These devices give your PC a normal-looking telephone handset or headset.

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How do I choose a VoIP phone?

The first choice is determining if you want a hardware VoIP phone or a software VoIP phone.

Hardware phones are generally easier to use and do not require a PC. Software phones are usually less expensive and may offer better options for CTI (Computer Telephony Integration).

Choosing a VoIP Phone

With either a hardware or software VoIP phones, the major considerations remain the same:

  • What VoIP call control protocols does the phone support?
    • H.323
    • SIP
    • MGCP
    • IAX2
  • What VoIP codecs does the phone support?
    • G.711
    • G.722
    • G.723
    • G.726
    • G.727
    • G.728
    • G.729
    • ILBC
    • Speex
    • GSM - Full Rate
    • GSM - Enhanced Full Rate
    • GSM - Half Rate
    • DoS FS-1015
  • Does the phone support 3-way calling
  • Does the phone support Do-Not-Disturb (DND)
  • Does the phone support custom ringtones?
  • Does the phone provide a method to work behind routers and NAT?
  • Does the phone support STUN?
  • Does the phone support Symmetric RTP?
  • Does the phone support a SIP outbound proxy?
  • Does the phone support QoS
  • Does the phone support encryption?
    • Secure RTP
    • AES

Choosing a Hardware VoIP Phone

When selecting a hardware VoIP phone, you should consider these items:

  • What connections does the VoIP phone support?
    • Ethernet
      • Does the phone support Power Over Ethernet?
    • Wi-Fi
    • Dialup
    • ISDN
  • Does the phone support IPv6?
  • Does the phone support videoconferencing?
  • Is the phone handset corded or cordless?
  • Does the phone have a handset or a headset?
  • Does the phone have a speakerphone?
  • Does the phone have an LCD display?
    • Is the LCD display backlit?
  • Does the phone have good ergonomics?
  • Do you like the style of the phones?

Choosing a Software VoIP Phone

If you choose a software VoIP phone, you should consider these items:

  • Does the phone software support my Operating System?
  • Is the phone software easy to use?
  • Does the software support customizable skins?
  • Does the software support videoconferencing?
  • Does the software support shared whiteboarding?

And, of course, the final purchasing decision should always include price as a criteria.

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How do I Become A VoIP Reseller?

If you are serious about reselling Voice over Internet Protocol (VoIP) services, there are some questions you will need to ask yourself first. Here are some simple guidelines to help you determine if and how you should pursue your goal of becoming a VoIP reseller.

Know the Service

If you really want to be a reseller for VoIP services, you need at least a basic working knowledge of how VoIP works and what type of applications are currently commonly used. Among the things you will need to understand are gateways and how they interact with voice switches. You will also need to understand the process for creating an integrated voice package that allows easy switching to and from conventional digital switches. Educate yourself on the basics before you attempt to move on to the next step--reselling.

Determine the Applications You Want to Sell

You may want to market VoIP to audio teleconferencing companies as a cost efficient means of participation during conference calls from any location. You may want to focus on providing Fortune 500 companies with a VoIP telephone service that virtually eliminates long distance charges. By selecting the types of applications you want to resell, you set the stage for moving on to your next step, which is becoming an agent or reseller.

Decide Whose Services You Want to Resell

Once you know what applications you want to resell, it is easy to begin investigating the companies that offer those types of services. Look into such qualities as reliability, customer support, private labelling options (if you want to sell under your own company name), and the rates offered. You may also want to see if billing and receiving payments are something you will have to do, or whether your supplier handle those functions for you. Don't be afraid to ask questions if you can't find documentation to specifically address a concern of yours. Companies that rely on resellers to generate revenue typically are very happy to work with persons who are serious and can think for themselves.

Being a VoIP reseller is an excellent way to make a living; and also a career choice that should be secure for a number of years to come. Investigate this possibility in more detail. You may find that this opportunity is right for you.

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What is a VoIP Gateway?

A VoIP Gateway, or Voice over IP Gateway, is a network device which helps to convert voice and fax calls, in real time, between an IP network and Public Switched Telephone Network (PSTN). It is a high performance gateway designed for Voice over IP applications. Typically, a VoIP gateway comes with the ability to support at least two T1/E1 digital channels. Most VoIP gateways feature at least one Ethernet and telephone port. Controlling a gateway can be done with the help of the various protocols like MGCP, SIP or LTP.

Benefits of VoIP Gateways

The main advantage of VoIP gateway is that it can provide connection with your existing telephone and fax machines through the traditional telephone networks, PBXs, and key systems. This makes the process of making calls over the IP network familiar to VoIP customers.

VoIP gateways can end a call from the telephone and can provide user admission control using IVR (Interactive Voice Response) system and provide accounting records for the call. Gateways also help direct outbound calls to a specific destination, or can end the call from another gateway and send the call to the PSTN.

VoIP gateways plays a major role in enhancing carrier services and also supports the simplicity of the telephone calls for less cost and easy access. Flexible call integration has been developed at less cost which enables programmable call progress tones and distinctive ring tones.

Functions of VoIP Gateways

The main functions of VoIP gateways include voice and fax compression or decompression, control signaling, call routing, and packetization. VoIP gateways are also power packed with additional features such as interfaces to external controllers like Gatekeepers or Softswitches, network management systems, and billing systems.

Future of VoIP Gateway Technology

Over the years, VoIP gateway has become an efficient and flexible solution and is used for office data and voice connectivity. Besides the connectivity performance, VoIP also offers better reliability under a variety of circumstances.

The future of VoIP gateway is very clear and precise; high-density, scaleable, open platforms need to be designed and implemented to allow the millions of installed telephones and fast-growing number of H.323 computer clients (such as Netscape's Communicator and Microsoft's NetMeeting) to communicate over IP. Many vendors are in the process of designing interoperable VoIP gateways according to the latest architectures to meet the changing demands of service providers, corporate network clients, and individual carriers.

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What is an IP PBX?

A PBX (Private Branch Exchange) is a small telephone switch owned by a company or organization. An IP PBX is simply a PBX which supports VoIP (Voice over IP). An IP PBX can also be referred to as a VoIP PBX.

An IP PBX may support VoIP both internally and externally. Internal VoIP support means that the IP PBX uses VoIP to communicate with each of its connect PBX phones. External VoIP supports means that the IP PBX uses VoIP to route calls to the outside world.

Most IP PBX's also support older analog or digital PBX phones and also support external connections on the public switched telephone network (PSTN).


Books on IP PBX's

PBX Systems for IP Telephony
PBX Systems for IP Telephony

The most efficient and economical ways to bring enterprise communication systems into the digital age are in this guide. PBX Systems for IP Telephony evaluates technologies, markets, and best practices for enterprise voice systems, messaging, and customer contact centers.

The heart and brains of an enterprise communications network, the PBX (Private Branch Exchange) can be the vital link that interfaces businesses and their customers. This guide, from the recognized expert in telephony systems, provides answers. Whether you need to IP-enable a PBX system for a small business, make complex choices for the advanced call center, or gain the expertise to integrate a variety of communication systems into a state-of-the-art foundation for your e-business vision, PBX Systems for IP Telephony should be your first choice.

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What is VoIP Security?

Any technology that involves transfer of data or information is prone to compromised security. It happens with telephones, cell phones, email and Internet transactions. Because VoIP (Voice Over Internet Protocol) has the internet as its mode of transference it's possible to have your Internet-based called intercepted. To make matters worse, there are techno-troublemakers who are armed with the hacking skills needed to eavesdrop on virtually any call over the Internet they want to. It is impossible to ensure total security on information flow over the web including Internet based phone calls. As new technologies emerge with more highly developed security protocols, there will be those who consider it a unique challenge to crack these online defenses rendering security advances antiquated. The Internet has been notorious for alternating security breaches and accompanying fixes.

As VoIP becomes more popular, VoIP security continues to be stressed as a key to advancement of this technology, especially since it will thrive in the realm of the World Wide Web. There are, however, advances in VoIP security that have been utilized by VoIP providers in order to ensure protection of customer's personal information.

VoIP Security is IP Security

VoIP is vulnerable to all security issues that generally affect the traditional IP data networks. This includes viruses, worms and denial of service (DoS), spoofing, port scanning, unauthorized access from a third party. and toll fraud. In short, the same issues you deal with in compromised Internet function can be linked to the use of VoIP technology.

VoIP's Defensive Linemen

The two primary methods of security for VoIP users are tunneling and encryption. These security measures assist in providing a mechanism of trust in the safe use of the VoIP user's personal data. Most VoIP providers use Layer 2 tunneling and an encryption method called Secure Sockets Layer or SSL to keep hackers at bay. Large corporate enterprises are using similar security mechanisms based on encryption for all internal traffic flowing over the VoIP system as well. It is advisable to route all inbound VoIP traffic that flows via a firewall through a proxy server, thus eliminating any direct connection with the internet.

On a larger level, organizations that are using VoIP as a popular mode of communication rely on a multiple level defense that addresses most VoIP security issues. In this scenario, the VoIP network is divided into secure zones protected by layers of firewall, intrusion prevention, and various additional security mechanisms. The advantage with this strategy is that it allows an organization to logically split and secure separate voice and data networks in front of individual voice and data components and between interactive points within the network. A system (like the one just described) should be complete with authentication, controls access (passwords and firewalls), encryption, an audit trail of calls, and facilities. Recording these issues can prevent security issue to a large degree because they are traceable.

Securing Your VoIP Network

While VoIP being internet-based is a key vulnerability, it also has its beneficial side. The years of experience in fending off or foiling internet attacks is experience that can be used in blocking VoIP assaults; the lessons learned in the data networking field are just as applicable to VoIP networking.

One approach that should be given serious consideration is setting up a separate network for VoIP applications, running in parallel but separate from the data network. This may be considered a serious expense item that is incompatible with the perceived savings from VoIP. On the other hand, one has to consider the potential costs involved if both networks become corrupted or disrupted from an attack on one which also disrupts the other.

Here are some other methods for securing a VoIP network:

  • – Enable as many of the manufacturer's security protocols as possible, adapting or 'tweaking' these to your own specifications rather than simply following manufacturer's defaults. Keep in mind that hackers and other attackers would probably know these defaults as well.
  • – Apply a strong authentication and encryption for both data and voice networks. As noted above, use the lessons learned in dealing with data network security problems to establish a preemptive stance in dealing with potential VoIP security concerns.
  • – Work out access controls and authentication protocols to ensure that only legitimate users can gain access to the VoIP network.
  • – Use gateway and host-based anti-virus as well as anti-spyware programs to protect crucial VoIP servers. At the same time, consider establishing perimeter security protocols to protect both networks.

A key point to remember is that VoIP is built on already established equipment and applications. The experiences and lessons gained from dealing with security threats to the data network can and should be used in developing security for the voice network.

Conclusion

Because VoIP is a newer technology there is a lot of discussion about its security and reliability. But it may be interesting to note that VoIP is actually more secure than normal email or even bill paying online. You may not need to be too worried about the security issues related to VoIP technology. Many newer technologies are emerging and, given the current trend, it won't take long before VoIP will be as secure as any other communication technology available today. Until then, if you are not sending highly sensitive information over the internet, VoIP is a relatively safe, reliable, and cost effective means of communication.

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What are books on VoIP?

Voice over IP Fundamentals
Voice over IP Fundamentals		 The authors of Voice over IP Fundamentals--three packet-voice specialists at Cisco Systems--initiate their exploration of next-generation technologies for supporting conversations across large distances: the switched telephone network as implemented on large (intercontinental) and small (building and enterprise) scales. They then point out problems with the old way of doing things and illuminate the standards and regulatory conditions that have made Internet telephony attractive. Signaling System 7 (SS7) gets particularly insightful coverage, with ample graphical support for the clear, fact-rich, example-laden prose.

The authors do a great service for readers by breaking packet telephony into its component technologies and explaining each one carefully. Coverage of the various protocols that enable voice over IP, particularly H.323 and Session Initiation Protocol (SIP), is simultaneously clear and deep. The same goes for media gateway protocols and various schemes for translating sounds into digital signals and back again, while retaining maximum clarity. There's even some practical material; concluding chapters diagram Cisco router configurations for voice traffic and flesh out solutions with case studies.

You'll like this book if you need to implement a voice over IP system and know more about IP than you do about traditional voice telecommunications. The patient and detailed explanations of traditional telephony concepts and voice over IP protocols will mesh nicely with your existing data communications knowledge, enabling you to make wise design and product decisions.
Switching to VoIP
Switching to VoIP		More and more businesses today have their receive phone service through Internet instead of local phone company lines. Many businesses are also using their internal local and wide-area network infrastructure to replace legacy enterprise telephone networks. This migration to a single network carrying voice and data is called convergence, and it's revolutionizing the world of telecommunications by slashing costs and empowering users. The technology of families driving this convergence is called VoIP, or Voice over IP.

VoIP has advanced Internet-based telephony to a viable solution, piquing the interest of companies small and large. The primary reason for migrating to VoIP is cost, as it equalizes the costs of long distance calls, local calls, and e-mails to fractions of a penny per use. But the real enterprise turn-on is how VoIP empowers businesses to mold and customize telecom and datacom solutions using a single, cohesive networking platform. These business drivers are so compelling that legacy telephony is going the way of the dinosaur, yielding to Voice over IP as the dominant enterprise communications paradigm.

Developed from real-world experience by a senior developer, O'Reilly's Switching to VoIP provides solutions for the most common VoIP migration challenges. So if you're a network professional who is migrating from a traditional telephony system to a modern, feature-rich network, this book is a must-have. You'll discover the strengths and weaknesses of circuit-switched and packet-switched networks, how VoIP systems impact network infrastructure, as well as solutions for common challenges involved with IP voice migrations.
VoIP For Dummies
VoIP For Dummies		Put your phone system on your computer network and see the savings See how to get started with VoIP, how it works, and why it saves you money VoIP is techspeak for "voice over Internet protocol," but it could spell "saving big bucks" for your business! Here's where to get the scoop in plain English. Find out how VoIP can save you money, how voice communication travels online, and how to choose the best way to integrate your phone system with your network at home or at the office. Discover how to Use VoIP for your business or home phone service Choose the best network type Set up VoIP on a wireless network Understand transports and services Demonstrate VoIP's advantages to management.
VoIP Crash Course
VoIP Crash Course		Recent advances in VoIP (Voice over IP) technology have made it the solution of choice for voice service because of its low cost and increased reliability. Voice Over IP Crash Course offers practical technology coverage, while discussing the business, strategic and competitive implications of VoIP deployment in corporations. The book also covers the challenges faced by service providers as they evolve to an IP infrastructure while continuing to operate the PSTN.
IP Telephony Unveiled
IP Telephony Unveiled		 This book explains four key points to help you successfully implement your IP telephony strategy:
  • IP telephony works today. This is not new, unproven technology. Thousands of customers have implemented IP telephony successfully. So can you.
  • Expect to save money. IP telephony may well cost your organization money-initially. But the business impact and post-installation process improvements give you a significant and rapid return on your investment.
  • It's more than voice over IP. You'll understand the difference between voice over IP (VoIP) and IP telephony and what that means for your business. This is critical. They are not the same.
  • It's more than a dial tone. There are potential business-impacting applications within your own organization. IP Telephony Unveiled helps you recognize these applications.
The emerging IP telephony market is fraught with misunderstandings and misinformation. IP telephony can impact a company's business model in tremendous ways. It can open new revenue streams, enhance profitability, drive new levels of customer and employee satisfaction, and be a key enabler in a company's strategy to differentiate itself competitively-but only if you're aware of these benefits.

IP Telephony Unveiled is written for all those responsible for corporate strategies for revenue generation, cost containment, and customer satisfaction. IP Telephony Unveiled uncovers the value behind this technology, which helps you see past what might appear to be only a new telephone system, to understand the strategic enabler laying dormant in many companies' networks. Through this book, you will understand the real benefits of an IP telephony strategy and get assistance in developing this strategy inside your organization.
Taking Charge of Your VoIP Project
Taking Charge of Your VoIP Project		The step-by-step approach to VoIP deployment and management enables you to plan early and properly for successful VoIP integration with your existing systems, networks, and applications.
  • The detailed introduction offers a common grounding for members of both the telephony and data networking communities.
  • IT managers and project leaders are armed with details on building a business case for VoIP, including details of return-on-investment (ROI) analysis and justification.
  • A VoIP deployment is presented as a major IT project, enabling you to understand the steps involved and the required resources.
  • The comprehensive look at quality of service and tuning describes when and where to use them in a VoIP deployment. These are often the most complex topics in VoIP; you'll get smart recommendations on which techniques to use in various circumstances.
  • You learn how to plan for VoIP security, including prevention, detection, and reaction.
Voice over IP (VoIP) is the telephone system of the future. Problem is, VoIP is not yet widely deployed, so there are few skilled practitioners today. As you make your move to VoIP, how will you know how to make VoIP work and keep it working well? What changes will you need to make without disrupting your business? How can you show your return on this investment?

Many books contain technical details about VoIP, but few explain in plain language how to make it run successfully in an enterprise. Taking Charge of Your VoIP Project provides the detailed plans you need to be successful in your organization's deployment of VoIP. Through their years of work in the field, authors John Q. Walker and Jeffrey T. Hicks bring a project-oriented approach to VoIP, with much-needed clarity on getting VoIP to work well.

Taking Charge of Your VoIP Project starts with simple concepts, each chapter building on the knowledge from the last. Although not a technical manual, you learn about the standards, such as H.323, G.711, and Real-Time Transport Protocol (RTP), and the implications they have on your VoIP system. Most importantly, you'll gain expert advice and a systematic guide on how to make VoIP work for your organization.
VoIP Security
VoIP Security		VoIP Security has been designed to help the reader fully understand, prepare for and mediate current security and QoS risks in todays complex and ever changing converged network environment and it will help you secure your VoIP network whether you are at the planning, implementation, or post-implementation phase of your VoIP infrastructure.

This book will teach you how to plan for and implement VoIP security solutions in converged network infrastructures. Whether you have picked up this book out of curiosity or professional interest . . . it is not too late to read this book and gain a deep understanding of what needs to be done in a VoIP implementation.

In the rush to be first to market or to implement the latest and greatest technology, many current implementations of VoIP infrastructures, both large and small, have been implemented with minimal thought to QoS and almost no thought to security and interoperability.
IP Telephony: Deploying Voice-over-IP Protocols
IP Telephony: Deploying Voice-over-IP Protocols		IP Telephony, enabled by softswitches, is going to usher in a new era in telecommunications. By putting voice and data over one IP network, operators can enjoy lower costs and create new, revenue-generating "multimedia" services. This valuable reference offers a comprehensive overview of the technology behind IP telephony and offers essential information to network engineers, designers and managers who need to understand the protocols and explore the issues involved in migrating the existing telephony infrastructure to an IP-based real time communication service. Drawing on extensive research and practical development experience in VoIP from its earliest stages, the authors give access to all the relevant standards and cutting-edge techniques in a single resource. IP Telephony: Deploying Voice-over-IP Protocols: Assumes a working knowledge of IP and networking and addresses the technical aspects of real-time communication over IP. Presents a high level overview of packet media transport technologies, covering all the major VoIP protocols - SIP, H323 and MGCP Details specific strategies to design services for public networks where endpoints cannot be trusted and can be behind firewalls. Explores the problems that may arise from incomplete protocol implementations, or architectures optimized for private networks which fail in a public environment. This amply illustrated, state-of-the art reference tool will be an invaluable resource for all those involved in the practical deployment of VoIP technology.
Beyond VoIP Protocols: Understanding Voice Technology and Networking Techniques for IP Telephony
Beyond VoIP Protocols: Understanding Voice Technology and Networking Techniques for IP Telephony		This book offers a comprehensive overview of the issues to solve in order to deploy global revenue-generating effective "multimedia" services. Drawing on extensive research and practical deployment experience in VoIP, the authors provide essential advice for those seeking to design and implement a post-bubble VoIP network. Beyond VoIP Protocols: Understanding Voice Technology and Networking Techniques for IP Telephony Introduces the basics of speech coding and voice quality Demonstrates how quality of service may be built into the network and deals with dimensioning aspects, e.g. multipoint communications and how to model call seizures. Explores the potential of multicast to turn an IP backbone into an optimized broadcast medium Includes amply illustrated, state-of-the-art practical advice for formulating a complete deployment strategy A companion volume to "IP Telephony: Deploying VoIP Protocols", this book takes the reader a stage deeper into how to prepare the network and exploit VoIP technology to its full potential.
VoIP Hacks: Tips and Tools for Internet Telephony
VoIP Hacks: Tips and Tools for Internet Telephony		Voice over Internet Protocol (VoIP) is gaining a lot of attention these days, as more companies and individuals switch from standard telephone service to phone service via the Internet. The reason is simple: A single network to carry voice and data is easier to scale, maintain, and administer. As an added bonus, it's also cheaper, because VoIP is free of the endless government regulations and tariffs imposed upon phone companies.

VoIP is simply overflowing with hack potential, and VoIP Hacks is the practical guide from O'Reilly that presents these possibilities to you. It provides dozens of hands-on projects for building a VoIP network, showing you how to tweak and customize a multitude of exciting things to get the job done. Along the way, you'll also learn which standards and practices work best for your particular environment.
IP Telephony - The Integration of Robust VoIP Services
IP Telephony - The Integration of Robust VoIP Services		 Now that virtually every leading telecommunications service provider has committed to delivering IP-based telephony services, communications professionals face the enormous challenge of implementation. This hands-on guide brings together today's best-known answers and solutions for delivering Voice Over IP (VoIP) services with the quality customers demand. No other book covers the combined issues of protocol signaling, media transport methodology, reference topological considerations, and voice quality testing in service offerings. Bill Douskalis presents systematic coverage of every aspect of IP-based telephony:

Coverage includes:
  • A realistic reference topology for implementing and benchmarking voice quality in IP telephony
  • Detailed explanations of call setup using each major competing technology
  • Signaling, bearer transport, and other key network elements
  • In-depth network and service performance analysis in both "normal" and impaired scenarios
  • State-of-the-art traces and performance measurements taken from actual IP networks
No matter what your role in delivering VoIP services, IP Telephony delivers the specifics you need to speed deployment, improve reliability, ensure quality, and simplify troubleshooting. Precise, thorough, and based firmly in the real world, it is simply indispensable.

The accompanying CD-ROM contains Hewlett-Packard Internet Advisor software that runs off-line—view live VoIP traffic examples! It also includes sample capture files of the H.323, MGCP, and SIP protocols; the latest IETF Working Group documents for VoIP; and an assortment of white papers and application notes that provide a real-world view of IP telephony.
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What is SPIT?

VoIP spam or Spam over Internet Telephony (SPIT) is one of the foreseen future forms of spamming that Internet authorities are preparing for today. With the increasing use and dependence on the Internet for communications and data transfer, malicious software programmers have taken advantage by creating VoIP bots with the ability to harvest data and advertise massively at a very small cost. These advertising methods include email spams, SPIMS or spams over instant messaging applications, malicious bots that generate pop up ads, initiate redirects, etc.

With the inevitable popularity of VoIP over the traditional telephone, authorities are convinced that this is where the next form of spam will come from. In this case, the unsolicited emails will be replaced by video or audio recordings advertising dubious products and services. Prank callers will also take advantage of this new frontier as the new technology becomes more available. This is even more profitable for such users as they can send automated or pre-recorded advertising messages to thousands of users with just one click, making it a very cheap operation to run.

SPIT will also have more impact on users than unsolicited instant messaging and email spam as it has the potential of clogging up the network. Given enough SPIT volume, users may not have any other options that to hang the VoIP phone 'off the hook'.

Other threats include spammers who might take temporary control of a user's systems to launch VoIP attacks on other networks, hackers that will inject profane words in conversations, fake voice mails and viruses that have the ability to use critical bandwidth.

Furthermore, These VoIP bots have the capability of launching automated DDoS or distributed denial of service attacks against rival corporations or users using VoIP with SIP protocols and vulnerabilities. Botnets armed with VoIP-directed software will play a big role in launching these kinds of attacks.


VoIP Built-In Security

VoIP, however, will have the usual array of spam defenses that other forms of Internet communication applications like emails and instant messengers have to combat unsolicited video/voice communication. This will include the stealth mode of instant messenger applications, privacy options as well as spam reporting options.

Other security measures may also include separating the voice and data streams so that in the event that the voice lines do get clogged with traffic, the website traffic will not be affected and will remain operational. Anti-spyware and anti-virus systems coupled with SIP encryption systems designed for VoIP will also help a lot in screening incoming calls and data and detect any instructions in the system. This will prevent DDoS attacks from being launched against your company. Moreover, increased collaboration between ISPs and Internet authorities will also be effective in determining the locations of these spammers as well as blocking calls and data from dubious IP addresses.

Improvements and updates on the security systems for VoIP systems in its initial stages will also play a crucial role in making this communication option a cost effective and reliable alternative than telephones.

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What is H.323?

H.323 is an ITU standard multimedia conferencing protocol, which includes voice, video, and data conferencing, for use over packet-switched networks.

H.323 was the first standard for VoIP, but is being supplanted by SIP.

H.323 defines five components of a multimedia network:

  • Terminals
  • Multipoint Control Units (MCUs)
  • Gateways
  • Gatekeeper
  • Border Elements

Terminals are telephone and PC equipment which connect end-users to the H.323 network.

MCUs are responsible for managing conferences. MCU's consist of a Multipoint Controller (MC) and an optional Multipoint Processor (MP). The MC manages signaling and the MP manages media mixing and switching.

Gateways nterface the H.323 network with other networks, including PSTN (Public Switched Telephone Network) and other H.323 networks. Gateways consist of a Media Gateway Controller (MGC) and a Media Gateway (MG). The MGC is is responsible for call signaling functions and the MG is responsible for media-related functions.

Gatekeepers are responsible for admission control and address resolution. Gatekeepers are able to provide advanced services such as normally found in PBX's.

Border Elements are positioned between two H.323 networks and assist in call routing and call authorization.

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What is IAX?

IAX is a call control protocol for VoIP.

IAX was designed to replace the earlier call control protocols, H.323 and SIP.

IAX is much more bandwidth efficient than the competing VoIP call control protocols, enabling it to support more concurrent VoIP calls over the same amount of bandwidth.

IAX traffic uses UDP port 4569. The use of a single well-known port enables IAX to be compatible with NAT (Network Address Translation), which can be a serious difficulty for earlier VoIP call control protocols.

IAX supports authentication using RSA public keys with the SHA-1 message digest algorithm for digital signatures.

IAX was developed for the Asterisk PBX and originally stood for Inter-Asterisk eXchange. IAX is now supported by many other VoIP platforms.

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What is RTP?

RTP (Real-Time Transport Protocol) is used to encapsulate VoIP data packets inside UDP packets.

RTP is defined in RFC 3550 - RTP: A Transport Protocol for Real-Time Applications.

RTP provides end-to-end network transport functions suitable for applications transmitting real-time data, such as audio, video or simulation data, over multicast or unicast network services. RTP does not address resource reservation and does not guarantee quality-of-service for real-time services. The data transport is augmented by a control protocol (RTCP) to allow monitoring of the data delivery in a manner scalable to large multicast networks, and to provide minimal control and identification functionality. RTP and RTCP are designed to be independent of the underlying transport and network layers. The protocol supports the use of RTP-level translators and mixers.

Other RFCs which document RTP include:

Books on RTP

RTP: Audio and Video for the Internet
RTP: Audio and Video for the Internet		 RTP (Real-time Transport Protocol) provides a framework for the delivery of audio and video across IP networks with unprecedented quality and reliability. In RTP: Audio and Video for the Internet, Colin Perkins provides readers with detailed technical guidance for designing, implementing, and managing any RTP-based system.

By bringing together crucial information that was previously scattered or difficult to find, Perkins has created an resource that enables network professionals to leverage RTP's benefits in a wide range of Voice-over IP (VoIP) and streaming media applications. Colin demonstrates how RTP supports audio/video transmission in IP networks, and shares strategies for maximizing performance, robustness, security, and privacy.

Comprehensive, exceptionally clear, and filled with examples, RTP: Audio and Video for the Internet is the definitive reference on RTP for every audio/video application designer, developer, researcher, and administrator.

Key coverage in the book includes:
  • RTP's goals, design philosophy, and relationships with other protocols
  • The psychology of human perception in the design of media delivery systems
  • RTP data transfer and control protocols, including framing, loss detection, reception quality feedback, and membership control
  • Media playout, timing, and synchronization, including lip synchronization
  • Mitigating network problems: error concealment, error correction, and congestion control
  • Optimizing performance over low-speed links: header compression, multiplexing, and tunneling
  • Integrating leading media codecs and standards into RTP systems
  • Securing RTP sessions: encryption, authentication, and the new secure RTP profile for wireless networks
  • Extensive references and practical examples
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What is RSVP?

RSVP (Resource ReSerVation Protocol) is a protocol used in VoIP to manage QoS (Quality of Service).


RSVP works by requesting that required bandwidth and latency be "reserved" for the VoIP telephone call by every network device between the two endpoints.

RSVP is defined in RFC 2205: Resource ReSerVation Protocol (RSVP).

RSVP is a unicast and multicast signaling protocol, designed to install and maintain reservation state information at each router along the path of a stream of data.

The RSVP protocol is used by a host to request specific qualities of service from the network for particular application data streams or flows. RSVP is also used by routers to deliver quality-of-service (QoS) requests to all nodes along the path(s) of the flows and to establish and maintain state to provide the requested service. RSVP requests will generally result in resources being reserved in each node along the data path.

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What is MGCP?

MGCP (Media Gateway Control Protocol) is a protocol used within a Voice over IP (VoIP) system. This internal protocol was primarily developed to address the demands of carrier-based IP telephone networks. MGCP is a complementary protocol for both H.323 and SIP, which was designed as an internal protocol between the Media Gateway Controller and the Media Gateway. In MGCP, an MGC primarily handles all the call processing by linking with the IP network through constant communications with an IP signaling device, for example an SIP Server or an H.323 gatekeeper.

MGCP is comprised of a Call Agent, one MG (media gateway) which performs the conversion of media signals between circuits and packets, and one SG (signaling gateway) when connected to the PSTN (Public Switched Telephone Network). MGCP is widely used between elements of a decomposed multimedia gateway. The gateway has a Call Agent, which is comprised of the call control "intelligence" and a media gateway boasting the media functions, for example conversion from TDM voice to Voice over IP.

Media Gateways feature endpoints for the Call Agent to create and manage media sessions with other multimedia endpoints. Endpoints are sources and/or sinks of data that can be physical or virtual. For creating physical endpoints, hardware installation is needed while virtual endpoint can be created using available software.

Call Agents come with the capability of creating new connections, or modify an existing connection. Generally, a media gateway is a network element which provides conversion between the data packets carried over the Internet or other packet networks and the voice signals carried by telephone lines. The Call Agent provides instructions to the endpoints to check for any events and - if there is any - create signals. The endpoints are designed in such a way as to automatically communicate changes in service state to the Call Agent. The Call Agent can audit endpoints and the connections on endpoints.

MGCP Connections

MGCP connections can be point to point or multipoint. Point to point connection can be a connection between two endpoints for transmitting data between these endpoints. Once the connection is setup between two endpoints, data transfer takes place between the endpoints. In a multipoint connection, the connection is set up between an endpoint and a multipoint session. In a multipoint connection, connections can be created over various types of bearer networks.

MGCP Architecture

MGCP came to be a much sought after application of VoIP technology because it is not involved in the frustrating work of encoding, decoding, and transferring voice signals. Though, the MGCP Call Agent works as a software switch for a VoIP network, it really does nothing more than simply direct the media gateways and signaling gateways which perform all the work. One of the main tasks in building a Call Agent is implementing the numerous possibilities in the protocol. There are various informational RFCs which may explain the expected behavior under a wide range of conditions.

In MGCP architecture, each and every command features a transaction ID, gets an acknowledgement and receives a response. This can be better understood as subscription architecture, as the Call Agent informs the media gateways and the signaling gateways as to what events it takes care of and what events it leaves unattended.

MGCP packets are generally found wrapped in UDP port 2427. Similar to what you might find in TCP protocols, MGCP datagrams are formatted with whitespace. An MGCP packer can either be a command or a response. Commands start with a four-letter verb while "responses" start with a three number response code.

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What is SIP?

SIP (Session Initiation Protocol) is an IETF standard multimedia conferencing protocol, which includes voice, video, and data conferencing, for use over packet-switched networks.

SIP is an open standard replacement for the ITU's H.323.

SIP is described in RFC 3621 - SIP: Session Initiation Protocol.

SIP is an application-layer control (signaling) protocol for creating, modifying, and terminating sessions with one or more participants. These sessions include Internet telephone calls, multimedia distribution, and multimedia conferences.

SIP invitations used to create sessions carry session descriptions that allow participants to agree on a set of compatible media types. SIP makes use of elements called proxy servers to help route requests to the user's current location, authenticate and authorize users for services, implement provider call-routing policies, and provide features to users. SIP also provides a registration function that allows users to upload their current locations for use by proxy servers. SIP runs on top of several different transport protocols.

Other RFC's which document SIP include:


Books on SIP

SIP Demystified
SIP Demystified		 State-of-the-art SIP primer SIP (Session Initiation Protocol) is the open standard that will make IP telephony an irresistible force in communications, doing for converged services what http does for the Web. SIP Demystified - authored by Gonzalo Camarillo, one of the contributors to SIP development in the IETF-;gives you the tools to keep your company and career competitive. This guide tells you why the standard is needed, what architectures it supports, and how it interacts with other protocols. As a bonus, you even get a context-setting background in data networking. Perfect if you're moving from switched voice into a data networking environment, here's everything you need to understand:
  • Where, why, and how SIP is used
  • What SIP can do and deliver
  • SIP's fit with other standards and systems
  • How to plan implementations of SIP-enabled services
  • How to size up and choose from available SIP products.
SIP: Understanding the Session Initiation Protocol
SIP: Understanding the Session Initiation Protocol		 This newly revised edition of the ground-breaking Artech House bestseller, SIP: Understanding the Session Initiation Protocol offers a thorough and up-to-date understanding of this revolutionary technology for IP Telephony. Essential reading for anyone involved in the development and operation of voice or data networks, the second edition includes brand new discussions on the use of SIP as a wireless communications protocol and mobility technology. Professionals find details on the latest application areas such as instant messaging.

The book explains how SIP is a highly-scalable and cost-effective way to offer new and exciting telecommunication feature sets. From an examination of SIP as a key component in the Internet multimedia conferencing architecture to a look at the future direction of SIP, practitioners get the knowledge they need to design "next generation" networks and develop new applications and software stacks.
Internet Communications Using SIP
Internet Communications Using SIP		Session Initiation Protocol (SIP) has gained tremendous market acceptance since it became an official IETF Internet communications standard in 1999. SIP is the technology that makes it possible for multimedia communications sessions on the Web--ones that allow voice, video, chat, interactive games, and others to run all at the same time. Now that the deployment of real SIP networks is about to take off, two leaders of the commercial rollout deliver complete guidance on this exciting new technology. Geared to IT and networking professionals and decision-makers at Internet service providers (ISPs), as well as networking (NSPs) and application (ASPs) service providers, this book helps readers sort through the available vendor offerings and services to discover how to integrate and maximize SIP's power across their networks.
SIP Beyond VoIP: The Next Step in the IP Communications Revolution
SIP Beyond VoIP: The Next Step in the IP Communications Revolution		VON Publishing's latest effort is SIP Beyond VoIP, an extraordinary 333-page effort that picks up where previous books have left off about SIP (Session Initiation Protocol), the protocol that has revolutionized the world of VoIP. The book's three distinguished authors relate in great detail how this versatile and extensible protocol has truly "moved beyond VoIP" and is now starting to have an impact on the whole telecommunication industry, including wireless and enterprise communications. Anyone who thinks that SIP has any real competitors will come away from this book in astonishment. "SIP Events" are the glue that even now integrates communications and applications. And "SIP Presence" may well be the "dial tone" of the 21st century. The book's advanced discussion of SIP interleaves with such associated topics as DNS (the Domain Name Service), ENUM (electronic numbering), NAT (Network Address Translation) and firewall traversal, security, Peer-to-Peer SIP (P2P SIP) networks, SIP-based conferencing/collaboration and even accessibility to communications for disabled people. This heavily illustrated, footnoted and fully-indexed book also has a foreword by Vinton Cerf, who writes: "It is my honest opinion that we have barely scratched the surface of the various applications to which SIP may be adapted. If we have seen 1% of the applications of SIP so far, then there are still 99% waiting to be invented, developed or deployed. The generality of SIP will make it a major workhorse the Internet of this century." If you think you know SIP, think again. Get this book its authors will set you straight about SIP, once and for all!

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What is common VoIP hardware?

VoIP hardware falls into several categories:
  • VoIP Interface Cards for PCs
  • PC Telephones
  • VoIP Telephones
  • VoIP Switches
  • VoIP Gateways
  • VoIP Routers
  • VoIP PBX's
  • VoIP Telephones

VoIP Interface Cards for PCs

VoIP Interface cards for PCs turn your PC into a very capable VoIP telephone.

Leading manufacturers of VoIP interface cards for the PC include:

  • Digium
  • VoiceTronix
  • Quicknet

PC Telephones

PC Telephones are telephones which attach to your PC, usually via the USB port, and allow you to make telephone calls through your PC.

VoIP Telephones

VoIP telephones are telephones which attach directly to Ethernet network ports.

VoIP Switches

VoIP switches are devices which allow you to connect multiple phone lines to one Ethernet port. This allows every telephone which is connected to the switch to place VoIP calls.

VoIP Gateways

VoIP Gateways connect VoIP networks to the PSTN (Public Switched Telephone Network).

VoIP Routers

VoIP Routers route VoIP traffic in much the same way that regular routers route IP (Internet Protocol) traffic.

VoIP PBX's

VoIP PBX's are high-tech low-cost equivalents of traditional telephone PBX's. In addition to traditional PBX functionality, VoIP PBX's configure and manage VoIP network capabilities.

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Saturday 2 August 2008

Image
Original Website - http://www.asterisk.org/

Asterisk is a complete PBX in software. It runs on Linux, BSD, Windows (emulated) and OS X and provides all of the features you would expect from a PBX and more. Asterisk does voice over IP in four protocols, and can interoperate with almost all standards-based telephony equipment using relatively inexpensive hardware.

Asterisk provides Voicemail services with Directory, Call Conferencing, Interactive Voice Response, Call Queuing. It has support for three-way calling, caller ID services, ADSI, IAX, SIP, H.323 (as both client and gateway), MGCP (call manager only) and SCCP/Skinny. Check the Features section for a more complete list.

Asterisk needs no additional hardware for Voice-over-IP, although it does expect a non-standard driver that implements dummy hardware as a non-portable timing mechanism (for certain applications such as conferencing). A single (or multiple) VOIP provider(s) can be used for outgoing and/or incoming calls (outgoing and incoming calls can be handled through entirely different VOIP and/or telco providers)

For interconnection with digital and analog telephony equipment, Asterisk supports a number of hardware devices, most notably all of the hardware manufactured by Asterisk's sponsor, Digium. Digium has single and quad span T1 and E1 interfaces for interconnection to PRI lines and channel banks. In addition, single to quad port analog FXO and FXS cards are available and are popular for small installations. Other vendors' cards can be used for BRI (ISDN2) or quad- and octo- port BRI based upon CAPI compatible cards or HFC chipset cards.

For interconnection with the cellular network (GSM or CDMA), Asterisk can use the Celliax channel driver or chan_mobile that is in the trunk now and there is also a unofficial backported version.

Lastly, standalone devices are available to do a wide range of tasks including providing fxo and fxs ports that simply plug into the LAN and register to Asterisk as an available device.

The current release versions of Asterisk are 1.2.27, 1.4.21.1 and 1.6.0-beta9.


This Wiki covers both the stable and the development branch of Asterisk. When adding new commands, applications and options, please also add a note on *when* this was added so that users may compare with their version date.

News





after the 4&8 ports anolog cards,and we will release our high performance 16 ports anlog cards to add our product line,and we will have 1/4/8/16 ports anolog cards and 1/4 ports PRI cards..

Starting Out

Books


Introduction


Hardware


Administration and system layout


Configuration


Management


Troubleshooting


General Reference



Country-Specific Information



Commercial support


SIP Service Providers


User Groups


Weekly SIP Asterisk Users Conference

  • x2z.eu THis conference is open to users at all levels of asterisk expertise

Howtos and Tutorials

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