Version 11 (Adrian Georgescu, 04/19/2009 10:52 am) → Version 12/161 (Adrian Georgescu, 04/19/2009 10:54 am)

= Welcome to SIP2SIP =

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Make acquaintance with your new SIP address. SIP stands for Session
Initiation Protocol and is becoming the universal way to communicate over
the Internet using both software and hardware devices.

You may use this service to register a SIP account that allows you to communicate using audio and video over the Internet using the SIP protocol. The account is Presence/IM enabled with support for relevant SIP SIMPLE standards for MSRP relay extension, Presence Agent, XCAP and RLS services.

== Register an account ==

Go to:

== Account features ==

* Publicly reachable SIP address
* Works behind NAT
* Parallel forking to multiple devices
* Support for SIP SIMPLE presence (PUBLISH method, XCAP storage, RLS services)
* Support for SIP SIMPLE instant messaging (IM based on MSRP and MSRP relay extension)
* Voice and video
* T.38 fax support
* ENUM routing based on lookups under
* Call forwarding
* Time of day routing
* White list and black list for incoming callers

== Configure your account ==

Setup your SIP software or hardware device as follows:

* Username: XXX
* Password: YYY
* Domain:
* Outboud proxy:

== How to use your SIP address ==

* To test audio, call 3333, you should hear some music playing
* To test microphone call 4444, you should hear your echo back
* You may call to any other SIP account user@domain
* To access your voicemail or mailbox settings dial *70
* Your voice messages are delivered to
* To review your call go to

== How to change your settings ==

Login to the settings page with your SIP credentials:
* SIP address:
* Password: YYY

== Paid services ==

If you wish to support the further development of this free service and the
open-source software behind it, we currently provide you with the following

* To create SIP addresses under your own domains, register or transfer your Internet domains at
* To make calls to PSTN destinations add credit to your SIP account at

== PSTN dialing ==

To dial a PSTN destination dial +NUMBER. The NUMBER must be a fully qualified E.164 number (country code + network number + subscriber number). First an ENUM lookup is attempted. If a SIP destination is found, the call will be routed to it, if ENUM lookup does not return a valid SIP address, the call is directed to a PSTN gateway. To use the PSTN gateway you must have a positive credit. To add credit for your account login to the SIP settings page and click on Credit tab.

Price list for dialing to PSTN destinations is available [ here]. The call costs are logged in the Credit section of your [ SIP settings page].

== Software used ==

If you plan to offer a similar SIP service, you can setup your own SIP infrastructure using the following elements:

|| OpenSIPS || SIP server ||||
|| OpenXCAP|| Policy server ||||
|| MediaProxy || RTP media relay ||||
|| MSRP Relay|| MSRP media relay||||

Or purchase a commercial solution like [ Multimedia Service Platform]