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Adrian Georgescu, 04/21/2009 11:17 am

= Using SIP =

<acronym title="WikiStart, Sip*, depth=3">TOC</acronym>

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 SIP address

When you register a SIP account on SIP2SIP, a SIP address under @sip2sip.info domain is automatically allocated to you.

SIP protocol uses the same identifier format as an email address in the form of user@domain. Actually, you can use the same address for both email messaging and SIP applications providing that you have control upon your own domain, its DNS records and access to a service like SIP2SIP.

To create SIP addresses under your own domains, you can register or transfer for a fee 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 sessions

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

=== 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.


=== Session details ===



SIP2SIP provides you with a SIP SIMPLE presence agent. To use presence you must:

1. To publish your presence send PUBLISH for '''Event: presence''' to your own SIP address, containing the body describing your presence information in XML format
1. To subscribe to a SIP address, send a SUBSCRIBE message for '''Event: presence'''
1. To subscribe to a list of SIP addresses (a.k.a. rls-services), send a SUBSCRIBE message for '''Event: presence''' containing an extra header: '''Require: eventlist'''
1. To monitor who has subscribed to your presence information you must send SUBSCRIBe for '''Event: presence.winfo''' to your own SIP address
1. To allow others to subscribe to your published information you must XCAP protocol for manipulating '''pres-rules''' policy document
1. To store your buddy list on the server you must XCAP protocol for manipulating '''resource-lists''' and '''rls-services''' documents


Instant messaging and file transfer * For instant messaging your client must support MSRP protocol, MSRP relay extension * 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 Replace XYZ with any username you wish, a new chat room will be created for you. * For file transfer your client must support [http://tools.ietf.org/html/draft-ietf-mmusic-file-transfer-mech-11 draft-ietf-mmusic-file-transfer-mech-11].

msp-enum-lookup.png (60.3 kB) Adrian Georgescu, 04/19/2009 12:23 pm

sip2sip-sessions-details.png (59.3 kB) Adrian Georgescu, 04/19/2009 12:29 pm

sip2sip-sessions-search.png (41.6 kB) Adrian Georgescu, 04/19/2009 12:29 pm