sip2sip.info

sip2sip.info has a free SIP service. An account is easily created by giving name, acount name, password and email. After that it is possible to log in to access the setttings for the account. The SIP address that is created has the form user@sip2sip.info

The settings to be used for the service are listed here. More information is available here and here.

It is possible to use UDP, TCP eller TLS, but sip2sip.info recommend using TLS, since that will avoid problems that might be related to the used network.
TLS encrypts the call meta data, so this is another upside of using TLS. The only reason to use anything else that TLS is is the SIP phone that is used does not support TLS. But hardware phones of high quiality and software phones usually have support for TLS.

The settings are as follows:

  • Domain: sip2sip.info
  • Protocol: TLS
  • Outgoing proxy: proxy.sipthor.net port 443

Normally there is no need for any extra settings to publish your own public IP address. There are a number of different methods to handle the public IP address from behind a NAT, for example STUN, TURN, ICE och SIP ALG. But sip2sip.info's servers can normally handle this automatically without any extra configuration. When using a router that does not handle SIP correctly, there might be problems when using UDP or TCP. But when TLS is used, the router can not modify the encrypted traffic, and no problems will will be introduced in the local router.

Extra services

sip2sip.info has an answer machine at the sip address: 1233@sip2sip.info
There are also settings for the answer machine in the web portal. It is for example possible to configure if messages should be stored on the server, and if the messages should be sent to an email address.

There is an echo address that echoes incoming counds at the address echo@conference.sip2sip.info alternatively 4444@sip2sip.info. This can be useful when testing a sip address.

Pause music is at the address 3333@sip2sip.info. This may also be useful when testing a sip configuration.

There is a possibility to use phone conferences by using the address room@conference.sip2sip.info, where "room" can be changed to an arbitrary text string. All persons in the conference should call the same room, which willl make them hear each other.

Here is an example of how the configuration for sip2sip.info might look like:




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