SipTesting » History » Revision 78
Revision 77 (Adrian Georgescu, 10/11/2012 11:45 PM) → Revision 78/120 (Adrian Georgescu, 10/13/2012 12:14 PM)
h1. Using your SIP account First [[SipDeviceConfiguration|configure your SIP device]]. 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. > Calls to SIP addresses containing only numbers and starting with a zero are always routed to the PSTN gateways. This is an arbitrary convention configured in the platform for detecting calls meant to be routed to PSTN. h2. Test Numbers * To test audio sessions, call 3333, you should hear some music playing * To test microphone call 4444, you should hear your echo back h2. Session Details * To review your SIP sessions go to https://mdns.sipthor.net/CDRTool 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: | *Port range* | *Protocol* | *Description* | *Application* | | 5060 | UDP | SIP signaling | OpenSIPS - SIP Proxy/Registrar/Presence Agent | | 5060 | TCP | 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. 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://mdns.sipthor.net. 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. 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. Conferencing * Using "Blink":http://icanblink.com SIP client go to menu Call -> Join Conference. Choose a room and connect. * Using any SIP client connect to <room>@conference.sip2sip.info Replace the <room> with the desired room name. * XMPP clients can connect to room@conference.sip2sip.info. Replace room with the desired room name. h3. Supported media * Audio is supported for SIP clients using G722, G711 or Speex codecs * 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. Tested Clients for Multiparty Text * XMPP clients: "PSI":http://psi-im.org/download/, "Adium":http://adium.im, Apple iChat * SIP clients: "Blink":http://icanblink.com 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 h2. Calls to PSTN network Calls to PSTN are possible to 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. *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 h2. 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.