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.
You receive a lifetime email-like SIP account like firstname.lastname@example.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 email@example.com.
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.
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.
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.
Sign up here for a free SIP account. Choose an iptel.org name if it is not already taken.. Sign Up
Sign in here if you already have an account. If you have forgotten your password, proceed HERE. Sign In
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
The minimum information which must be put in every SIP phone is your SIP address (like sip:firstname.lastname@example.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.
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".
Use then a numerical alias to your SIP address. You will find it in the "My Account - General" webpage.
Call sip:email@example.com for an audio announcement, or sip:firstname.lastname@example.org to hear yourself.
Login using your iptel.org credentials on the page https://tryit.iptel.org.
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.
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..
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/
The iptel.org service is using the Session Initiation Protocol family for comunication between the VoIP devices.
iptel.org is using JSSIP for web telephony. JSSIP is the most famous and wide-spread browser client.
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