SIP2SIP

h1. Using your SIP account

First [[SipDeviceConfiguration|configure your SIP device]]

h2. Incoming calls

When you enroll a SIP account on SIP2SIP infrastructure, 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. Then you must have your SIP device registered with SIP2SIP infrastructure.

h3. Calls from Web browsers

People can call your account using an WEBRTC enabled browser at https://webrtc.sipthor.net/#!/call/user@sip2sip.info

h3. Calls from PSTN network 

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 SIP2SIP) and if the SIP gateway provider that handles that number can translate that number into your SIP address. Technically, if you number is in ENUM e164.arpa or e164.org trees you can simply map your ENUM number to your SIP address yourself. Any ENUM ready gateway will be able to automatically find the SIP address you configured for your ENUM number.

h2. Calling Out

Using your SIP device you can call any other SIP address reachable over the Internet in the form of user@domain. If you use a SIP enabled phone featured only with a classic 12 keys keypad you will experience a crippled service as is either impossible or hard to dial other SIP addresses with it.

h3. Calls to PSTN network 

Every numeric destination stating with a zero under local domain is considered a PSTN destination and will be routed to the PSTN gateway.

Calls to PSTN are possible when using SIP accounts under @sip2sip.info domain. If you have used your own Internet domain, it is not possible to call out to PSTN. 

You can call to the classic telephone network (a.k.a. PSTN) after you have purchased credit. Price list for dialing to PSTN destinations is available "here":https://mdns.sipthor.net/sip_rates.html. The call costs are logged in the Credit section of your "SIP settings page":http://x.sip2sip.info. To add credit to your SIP account at http://x.sip2sip.info?tab=credit 

To dial a PSTN destination dial + or 00 in front of the actual number including country code. 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 open a ticket in the support interface. Caller id presentation works depending on the support for this feature of all intermediate gateways to the destination, it is not possible to guarantee its working.

To limit fraud in case of lost account credentials, a maximum of 2 simultaneous calls are permitted. If you need more, please ask our support.

*Important notes*

* Service numbers for premium services may not be reachable

* Emergency access number (e.g. 911, 112) are not available

* Not all international PSTN prefixes may be available depending on the capabilities of our outbound carriers

h3. Test Numbers 

* To test outgoing audio sessions, call 3333, you should hear some music playing 

* To test your microphone, call 4444, you should hear your echo back

* echo@conference.sip2sip.info can be used for echoing back both RTP audio and MSRP chat

h2. Conferencing

SIP2SIP supports ad-hoc multi-party conferencing for audio, chat and file transfers.

 * Using any SIP client connect to <room>@conference.sip2sip.info Replace the <room> with the desired room name. 

 * Using "Blink":http://icanblink.com SIP client go to menu Call -> Join Conference. Choose a room and connect.

 * XMPP clients can connect to room@conference.sip2sip.info. Replace room with the desired room name.

h3. Supported media

 * Audio codecs: Opus 48kHz, Speex 32kHz, G.722 16kHz, G.711 8kHz

 * Text Chat is supported for SIP clients using MSRP protocol

 * File Transfers are supported for SIP clients using MSRP protocol

 * SIP Clients that implement "Conference Event Package RFC 4575":http://tools.ietf.org/html/rfc4579 can retrieve the participants information

 * Multiparty Text Chat is supported between XMPP and SIP MSRP clients

h3. Room Access

| *Support Media*| *Address* | *Protocol* | *Recommended Clients*|

| Narrow band audio| +1-330-4090385 | PSTN US Fidelity Voice|  Any Phone|

| Narrow band audio| +1-425-9982650 | PSTN US IPKall |  Any Phone|

| Narrow band audio| +31-20-8005161 | PSTN EU|  Any Phone|

| Wideband Audio | ROOM@conference.sip2sip.info | SIP| Blink, Bria, Jitsi|

| MSRP Chat, MSRP File Transfer| ROOM@conference.sip2sip.info | SIP| Blink for OSX|

| Conference Information | ROOM@conference.sip2sip.info | SIP| Blink for OSX|

| Group Chat| ROOM@conference.sip2sip.info| XMPP Muc| Jitsi, Adium, iChat, Google|

| Ultra-wideband Audio| ROOM@conference.sip2sip.info| XMPP Jingle |Jitsi|

h2. XMPP interoperability

It is possible to exchange audio (using Jingle), presence and chat messages with external XMPP domains. The following domains are configured for XMPP gatewaying:

 * gmail.com

 * jit.si

 * xmpp

 * jabb

 * im.%

 * %.im

h2. NAT Traversal

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:

# Disable STUN as is unreliable  and leading to unexpected results

# Disable any SIP ALG support in the border router, most of the so called 'SIP enabled' routers on the market today are "simply broken":http://www.voip-info.org/wiki/view/Routers+SIP+ALG

Beware that corporate firewalls that have an explicit policy against SIP or poorly implemented "SIP ALGs":http://www.voip-info.org/wiki/view/Routers+SIP+ALG may still block your SIP signaling and/or media traffic. You need un-restricted access to the following ports used by SIP2SIP infrastructure:

| *Ports* | *Protocol* | *Description* | *Application* |

| 5060 | UDP | SIP signaling | OpenSIPS - SIP Proxy/Registrar/Presence Agent |

| 5060 | TCP  | SIP signaling | OpenSIPS - SIP Proxy/Registrar/Presence Agent |

| 443 | TLS | SIP signaling | OpenSIPS - SIP Proxy/Registrar/Presence Agent |

| 5269 | TCP  | XMPP signaling | SylkServer SIP/XMPP gateway |

| 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  |

h2. Voicemail

* To access your voicemail or mailbox settings dial 1233

* Your voice messages are delivered to your e-mail address

h2. IM and File Transfer

* For instant messaging your client must support MSRP protocol and its MSRP relay extension. 

h2. Presence

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

# To publish your presence send PUBLISH for *Event: presence* to your own SIP address, containing the body describing your presence information in PIDF format

# To subscribe to a SIP address, send a SUBSCRIBE message for *Event: presence*

# 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*

# To monitor who has subscribed to your presence information you must send SUBSCRIBe for *Event: presence.winfo* to your own SIP address

# To allow others to subscribe to your published information you must use XCAP protocol for manipulating *pres-rules* policy document 

# To store your buddy list on the server you must use XCAP protocol for manipulating *resource-lists* and *rls-services* documents

More information is available at http://wiki.sip2sip.info/news/23