SIP2SIP

= Using SIP =

NAT Traversal

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

Practically you do not need to set anything special in the client, NAT traversal is solved automatically by the SIP2SIP server infrastructure. We recommend actually that you check to have disabled all client features related to NAT traversal:

1. Disable any keep-alive technique like PING, NOTIFY, OPTION, CR/LF or STUN
 1. Disable any SIP ALG support in the border router, most of the so called 'SIP enabled' routers on the market today are [http://www.voip-info.org/wiki/view/Routers+SIP+ALG simply broken]
 1. Disable voice activation detection (VAD) in your device

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

SIP Address

When you register a SIP account on SIP2SIP, a SIP address under @sip2sip.info domain is automatically allocated to you. You must provide this SIP address to those that want to reach 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.

First [wiki:SipDeviceConfiguration configure your SIP device].

Internet Sessions * 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 your e-mail address

=== PSTN Outbound ===

Price list for dialing to PSTN destinations is available [https://secure.dns-hosting.info/sip_rates.html here]. The call costs are logged in the Credit section of your [http://x.sip2sip.info SIP settings page].

To make calls to PSTN destinations you must have a positive credit. To 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 set your caller id please send an e-mail to support address.

'''Important notes'''

• Service numbers for premium services may not be reachable
• Emergency access number (e.g. 911, 112) are not available

=== 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.arpa 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)

=== IM and File Transfer ===

• For instant messaging your client must support MSRP protocol and its 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 XYZ@chatserver.ag-projects.com 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].

[[Image(http://msrprelay.org/chrome/site/files/MSRP-Relay-Scenarios.png,link=http://msrprelay.org)]]

=== Session Details ===

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

[[Image(sip2sip-sessions-search.png,link=http://cdrtool.ag-projects.com)]]

Presence

SIP2SIP provides a SIP presence agent that handles SUBSCRIBE and PUBLISH methods for presence events.

1. To publish your presence send PUBLISH for '''Event: presence''' to your own SIP address, containing the body describing your presence information in PIDF 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 use XCAP protocol for manipulating '''pres-rules''' policy document 
 1. To store your buddy list on the server you must use XCAP protocol for manipulating '''resource-lists''' and '''rls-services''' documents

[[Image(http://openxcap.org/raw-attachment/wiki/WikiStart/SIMPLE-Server.png,link=http://openxcap.org)]]