THE VOIP ACCOUNT FOR LIFETIME

IPTEL.ORG

SERVICES

iptel.org provides free VoIP services for lifetime. Our users have been connecting with each other since 2002 using SIP and newly also WebRTC technology. A user can obtain an individual iptel.org account or an account for a whole domain.

hand with phone

LIFETIME VOIP ACCOUNT

You receive a lifetime email-like SIP account like john.doe@iptel.org. Use the account to make audio/video calls with users of iptel.org and other domains.You can even have your own domain hosted and use address like john@doe.com.

identity

3RD PARTY IDENTITY

You can set up a third-party SIP identity in your profile and use it to complete your iptel.org calls through other networks. This is often used for termination of your calls in PSTN.

web call

WEB TELEPHONY

The VoIP telephony services can also be used through web-browsers. You don't need any special equipment, just a web-browser and your credentials. Use of browsers also gets you security by encryption -- wiretappers will not be able to hear your call.

REQUIREMENTS

To use the iptel.org service you need some SIP-compliant (RFC3261) equipment: "hardphone", "softphone" or simply a smartphone app. Alternatively you can use a WebRTC-compliant browser.

Access to iptel.org's service is being provided on an 'AS IS' and 'AS AVAILABLE' basis. iptel.org makes no representation or warranties with respect to user's access of the service, and that the service will be available at any give time, free from errors, defects, omissions, failures or delays in delivery of data.

ACCOUNT

NEW SUBCRIBERS

Sign up here for a free SIP account. Choose an iptel.org name if it is not already taken.. Sign Up

EXISTING SUBSCRIBERS

Sign in here if you already have an account. If you have forgotten your password, proceed HERE. Sign In

DOMAIN HOSTING

Proceed here if you would like to have your domain served by a hosted SIP service. You must have administrative access to your DNS names. Have my domain

FAQ FOR SERVICE USERS

HOW TO CONFIGURE A SIP PHONE?

The minimum information which must be put in every SIP phone is your SIP address (like sip:john.doe@iptel.org) and the password you have chosen during subscription. Some phones require also outbound proxy address and registrar address: use sip.iptel.org then. If port number is asked, use the default 5060. Unfortunately many SIP phones have way too many other configuration parameters whose description is beyond this brief FAQ.


HOW TO TERMINATE MY CALLS IN PSTN?

iptel.org is not offering PSTN termination services. However, you may use a third-party SIP account with PSTN termination and use it for terminating calls to numerical destinations prefixed with the plus sign. Set it up in your profile under "My Account - Other".


WHAT IF SOME SIP EQUIPMENT ONLY HAS NUMERICAL KEYBOARD?

Use then a numerical alias to your SIP address. You will find it in the "My Account - General" webpage.


HOW CAN I TEST THE SERVICE IS WORKING FOR ME?

Call sip:music@iptel.org for an audio announcement, or sip:echo@iptel.org to hear yourself.


HOW CAN I MAKE PHONE CALLS USING A BROWSER?

Login using your iptel.org credentials on the page https://tryit.iptel.org.


I HAVE NOT RECEIVED AN ACCOUNT SETUP CONFIRMATION EMAIL

Most likely you included a wrong email address or your email server requires a confirmation from a new email address. Unfortunately our service does not handle such requests.

ABOUT US

IPTEL.ORG SERVICE

The iptel.org VoIP service was started back in 2000 as an early public VoIP service by Fraunhofer Fokus. Since then it has become a de-facto reference implementation. It is based on the SIP protocol and open-source implementations..

KAMAILIO / SER / SIP-ROUTER

The iptel.org site has been hosting the development activities around the open-source SIP server known as SIP Express Router or Kamailio. The iptel.org service is still using this great software. The development webpages have moved to https://www.kamailio.org/

SESSION INITIATION PROTOCOL

The iptel.org service is using the Session Initiation Protocol family for comunication between the VoIP devices.

JSSIP FOR WEB TELEPHONY

iptel.org is using JSSIP for web telephony. JSSIP is the most famous and wide-spread browser client.

JIRI KUTHAN

I run the iptel.org site since about 2000 when I started working on VoIP and SIP Express Router (now known as kamalio) at Fraunhofer Fokus in Berlin. My professional interests are Internet telephony, network security, Firewall and NAT technology, and software architecture. I live with my family in Prague and Berlin, current location almost subject to the Heisenberg uncertainty principle. Berlin is one of the most exciting multicultural European cities to live in -- it is green and blue, with great cultural life and long historical, cultural and enterpreneurial legacy. If my professional life occasionally leaves some free time for me I read books and go windsurfing or sailing. I fell in love with the "flying saiboat", RS700. I love returning to Antoine Exupery (The Wisdom of the Sands is remarkable), Milan Kundera, Stefan Zweig, Karel Capek. I also find Hamming's speech "You and your research" very inspiring..

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