SIP2SIP

= Using SIP =

[[TOC(WikiStart, Sip*, depth=3)]]

SIP2SIP service is NAT traversal proof for SIP signaling, RTP and MSRP media. Beware that corporate firewalls that have an explicit policy against SIP or poorly implemented [http://www.voip-info.org/wiki/view/Routers+SIP+ALG SIP ALGs] may still block your SIP signaling and/or media traffic.

You need un-restricted access to the following ports used by SIP2SIP infrastructure:

|| '''Port range''' || '''Protocol''' || '''Description''' || '''Application''' ||
|| 5060 || UDP || SIP signaling || OpenSIPS - SIP Proxy/Registrar/Presence Agent ||
|| 50000:60000 || UDP || RTP media || !MediaProxy - RTP media relay ||
|| 2855 || TLS || MSRP media || MSRP relay - MSRP media relay ||
|| 443 || TLS || XCAP storage || OpenXCAP - Presence policy management ||

== Your own SIP address ==

SIP protocol uses the same identifier format as email in the form of user@domain. Actually you can use the same address for fro both email and SIP providing you have full control upon the DNS records of your Internet domain.

To create SIP addresses under your own domains, register or transfer your Internet domains at https://secure.dns-hosting.info. On the same platform you can provision your DNS zones and records required for SIP services and create your own SIP accounts. The number of SIP addresses you may create is limited by a fair use policy dependent on the general use of the platform.

== Internet calls ==

First [wiki:SipDeviceConfiguration configure your device].

* To test audio sessions, 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

== Voicemail ==

* To access your voicemail or mailbox settings dial *70
* Your voice messages are delivered to ag@ag-projects.com

== Session details ==

* To review your SIP sessions go to https://secure.dns-hosting.info/CDRTool

[[Image(sip2sip-sessions-search.png)]]

== Multi-party IM ==

* To test multi-party instant messaging using MSRP, setup a [http://sipsimpleclient.com/wiki/sip_trace_msrp_rtp#sip_trace_msrp_rtp MSRP SIP session] to XYZ@chatserver.ag-projects.com Replace XYZ with any username you wish, a new chat room will be created for you.

== PSTN outbound ==

To make calls to PSTN destinations add credit to your SIP account at http://x.sip2sip.info?tab=prepaid

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 [https://secure.dns-hosting.info/sip2sip_rates.html here]. The call costs are logged in the Credit section of your [http://x.sip2sip.info SIP settings page].

== PSTN inbound ==

As you own a publicly reachable SIP address, you may receive calls from any SIP device that knows your address including a PSTN gateway. You can receive calls from PSTN if you own a telephone number (not provided by this service) and if the SIP gateway provider that handles that number can translate that number into your SIP address. Technically if you number is in public ENUM e164.arap tree you can simply map your ENUM number to your SIP address. Any ENUM ready gateway will be able to automatically find the SIP address you configured for your ENUM number.

[[Image(msp-enum-lookup.png)]]