Quick, Easy, and FREE Way to Start Using VoIP with Open Standards

VoIP ( Voice over Internet Protocol ) refers to communicating by voice over IP ( Internet Protocol ) networks. That includes the internet itself, and there are several software applications and systems for doing it. Most are chat programs that connect users of a closed network. Some allow you to place calls to and receive calls from “real” telephones i.e., those on the PSTN ( Public Switched Telephone Network ). Skype is probably the best known of these, and is available for Windows, Mac, Linux, and Android. With a free membership, you can call other Skype phones, and if you upgrade your account by depositing funds, you can call “real” phones. However, Skype is essentially just another proprietary and closed chat program.

In contrast, there are several software applications and phone services that allow you to call over the internet and even private networks that use open standards. Using open standards has some benefits. For one, you know, or at least can know, what you're getting. Being open means the source is public knowledge, so everybody who wants to know, can. There are no proprietary or other secrets lurking in the code. Also, being based on standards means that the various software applications can interoperate – even across computing platforms and operating systems. You don't have to use the same software as your mom, to call her. Many of the software applications are open source, so they are free, though many rely on user donations for support.

There are several physical forms that a telephone connection to an IP network can take. There are adapters that allow you to connect analog telephones. There are USB and bluetooth telephones and headsets. There are IP telephones that connect directly to an ethernet LAN ( Local Area Network ). Smart phones can connect to the internet via WiFi or 3G/4G cellular connections and run a VoIP application. Since your computer is already connected to the internet, the easiest way to start using VoIP is to install a soft phone and connect a headset to your computer.

When you called your mom the old fashioned way; i.e., using your bakelite wired phone, the telephone company's switches connected your phone with your mom's. A VoIP phone call (aka “session”) works by exchanging packets carrying digitized voice data through a virtual connection between phones (aka “endpoints”). You can make a VoIP call simply by entering the IP address of the other phone into your soft phone, but IP addresses change frequently. SIP (Session Initiation Protocol) was created to allow IP phones to be addressed like e-mail. A SIP server is used to set up calls between them.

IPTel operates a free SIP server service. Create an account at http://serweb.iptel.org/user/reg/index.php. Fill in your basic information, and create a user name and password. If you agree with the user terms and conditions, click accept. If you entered your e-mail address correctly, you will soon receive an e-mail from registrar@iptel.org thanking you for your registration, and asking you to follow a link to confirm your registration. After that, you will probably never hear from them, again.



Fill Out the Short Sign Up Form


There are several SIP soft phones (aka "clients") available for the major OSs. Some are particularly easy to set up, as they come pre-configured for the most popular SIP services. One of these is called “Yate” – an acronym for “Yet Another Telephony Engine” It is available for Windows, MacOS, and linux.

Visit their web site at yate.ro and you will see there is a lot going on with this project: including a PBX ( Private Branch Exchange ) and a cellular base station. You can download it directly from here, or if you are using Ubuntu, you can install it from the Software Center.



If you are using Ubuntu, you can install Yate from the Software Center.


Launch Yate, by clicking its icon.


Open The Yate SIP Client


To configure Yate to use your new IPTel account, click settings → accounts.


Select Accounts


The Accounts window will open. Click the +New button.


Click the +New Button


Select iptel.org from the drop down menu that appears.


Select iptel.org from the drop down menu


Enter your IPTel username and password into the boxes. Check the Save password box to avoid having to enter your password every time you start the program. Make sure sip is showing in the Protocol box.


Fill in Your Username and Password


You should now be ready to start making and taking calls through the iptel.org SIP sevice. At the bottom of the Yate window, you should see a message informing you of the status of your connection to the IPTel SIP server. If it does not tell you that you are registered, make sure that the account box on the main Yate Client window shows your IPTel account. If that isn't the problem, open the accounts window and verify you have entered your login credentials correctly.


Registered

Registration is a short and simple conversation between your softphone and the SIP server. To register, your Yate will send a “REGISTER” message to the server, and the server responds with an “OK” message. If registration fails, and your username and password are correct, it is most likely due to something blocking the message from you, to you, or both. This can be a local firewall running on your computer, a firewall running on your router, or a firewall running on your ISP's ( Internet Service Provider ) network. It is usually easiest to disable any firewall that you have under your control to eliminate it as a possible culprit. Do this only as a temporary step while troubleshooting, and be sure to re-enable it once you are finished. If your packets are suddenly able to get through, you'll know it is your firewall that is blocking the registration. If you can open port 5060 for UDP protocol through the firewall, you should be able to get everything working.

If you have a router with a feature called SIP ALG (Session Initiation Application Layer Gateway ), it is usually better to disable this. SIP ALG attempts to "help" your SIP connection. However, in most cases it prevents it from working. Unfortunately, many router firewalls do not work properly. Some indicate their firewall is disabled when, in fact, it is not. This is especially true for routers provided by somer ISPs, as many of these run a customized version of firmware to disable some features. The most common feature disabled is the ability to disable the firewall. If you are still having trouble, conduct an internet search of your router's make and model to see if others have experienced problems configuring your type of router for use with SIP.

Some ISPs have been known to block or corrupt SIP messages to prevent the use of VoIP over their networks. Usually, this is done by providers offering voice phone service, and do not want you to have an alternative their offering. If you discover your ISP is blocking any service, consider switching to another provider.

Once you have successfully registered Yate to IPTel, test your connection by trying a few calls. First, to check that you are able to receive audio, enter music in the call box, and click the Call button. La você saudade de fado singer Anamar should then begin pouring from your speakers. The music will serve to test the ability of your set-up to receive RTP (Real Time Protocol) audio. If you hear it playing, you're half way there.

You can test the operation of your microphone and the ability of your computer to transmit your sound out to the network by performing an echo test. As a convenience, IPTel provides an extension for doing this. Call IPTel's echo test function by entering “echo” in the destination box. When the server answers, you should hear your own voice echoed back to you, as you speak.

IPTel will not terminate calls to standard numbers on the PSTN. However, they will terminate calls to US toll free numbers, via their interconnections with a carrier called CallWithUs. To call a toll free number, precede the toll free number with the code “2227.” For example, to call information services for the 800 area code, you would "dial" or key in 222718005551212. This should route to AT&T 800 directory services. Unfortunately, and possibly because they have figured out that such calls are not from paying AT&T customers, no one will answer.

In the old days, telephone technicians working out in the field needed to test the lines on which they were working. This required automated, or as they used to be called “robotic,” services to handle these tests. MCI still operates an ANI verification responder at 1-800-437-7950. Enter 222718004377950 in the destination number box, and click on the call button. Each time you call this, MCI will speak back a different number. Presumably the PSTN side number for the appearance of the bridge between IPTel and the PSTN. While the reported ANI doesn't hold much meaning, it serves to verify your ability to make free calls to toll free numbers in the PSTN.

Your IPTel username forms the first part of your SIP address, which is in the form of username@iptel.org. If you have a keyboard, you can type in the full SIP address of the end point you want to call. Most telephones, however, only have a numeric keypad. When you created your IPTel account, you were also given a numeric telephone number that can be "dialed" from telephones not having a full keyboard. You can also set up a voicemail2email account so callers can leave a message for you when you are not online. You can even set it to send you messages to you via e-mail.

You should now be able to call and receive VoIP calls from any other IPTel subscriber using the open standards of SIP and RTP. You can use Yate to explore other services based on these and the other standards, including those that offer low cost interconnection with the PSTN.