SIP2SIP

h1. SIP Device Configuration

There are thousands of SIP devices on the market, for how to configure them we advise you to consult the support forum of the device manufacturer.  Please do not open a ticket related to how a particular device must be configured. Setup your SIP device as follows:

h2.  SIP Account Credentials

Account credentials are used for authentication and authorization of SIP requests performed by the SIP device. Your SIP address is XXX@sip2sip.info replace XXX with the username chosen during the account enrollment.

| Username| XXX|

| Password| YYY|

| Domain/Realm|sip2sip.info|

Register must be turned On in order to receive incoming calls. If your SIP devices is smart enough, there is no need to set manually anything else than the above settings. If you need to manually fine tune the configuration read below.

[[SipDevices|Specific SIP devices configuration]]

h2. Server Location

There are multiple SIP servers distributed in multiple geographic locations. To locate them, the SIP device must always perform DNS lookups as defined in SIP standard "RFC3263":http://www.ietf.org/rfc/rfc3263.txt (NAPTR + SRV + A DNS lookups). You must never set manually a host address or transport belonging to SIP2SIP server infrastructure into your SIP device as it may and will change over time. Your device must use DNS lookups instead of hardwiring any such settings into your SIP device. For informational purposes, the servers are reachable at the following addresses, but again you must query the DNS to discover them as they may and will change in the future.

| Host | Port | Protocol |

| proxy.sipthor.net| 5060| UDP |

| proxy.sipthor.net| 5060| TCP |

| proxy.sipthor.net| 5061| TLS |

h2. Presence

h3. XCAP Root

| XCAP Root | https://xcap.sipthor.net/xcap-root/|

h3. Required functionality 

To use SIMPLE presence the SIP client must support the following standards related to SIP and XCAP protocols:

 * SIP Presence Agent mode (PUBLISH method) RFC3903

 * Event packages: *presence* and *presence.winfo* 

 * RLS SUBSCRIBE RFC4662 and RLMI NOTIFY RFC4662

 * XCAP documents *rls-services* and *resource-lists* for contacts storage RFC4826

 * XCAP document *org.openmobilealliance.pres-rules* for authorization rules OMA XML 1.1

h3. Optional

Additional the SIP end-point may support the following extensions:

 * Event packages: xcap-diff for synchronizing XCAP documents between clients  

 * XCAP org.openmobilealliance.pres-content document for serving user icon

 * XCAP pidf-manipulation for offline status

 * XCAP xcap-caps and org.openmobilealliance.xcap-directory for listing all XCAP documents on the server

Example: Blink SIP client implementation http://projects.ag-projects.com/news/15

h2. MSRP Relay

If you use SIMPLE instant messaging based on "MSRP":http://tools.ietf.org/html/rfc4975, a "relay":http://tools.ietf.org/html/rfc4976 is available for helping traverse the NAT. You need to use the relay if you are the receiving party and you are behind a NAT-ed router. The MSRP relays can be found in the DNS by using SRV lookup for _msrps._tcp.sip2sip.info.

h2. STUN Servers

You may use STUN for ICE NAT traversal. The STUN servers can be found in the DNS by using SRV lookup for _stun._udp.sip2sip.info.

h2. NAT Traversal

SIP2SIP infrastructure is smart enough to handle the NAT traversal for both SIP signaling, RTP and MSRP media sessions. Also it supports ICE negotiation in the clients and provides automatically a TURN relay candidate. 

Practically, you should not set any NAT traversal features in the client as the chance of fixing things is much smaller than breaking them.

 * Do not use STUN for Register purposes

 * Do not set your client to discover a global IP address