SipTesting
Version 93 (Adrian Georgescu, 07/25/2013 01:27 pm)
1 | 74 | Adrian Georgescu | h1. Using your SIP account |
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2 | 1 | Adrian Georgescu | |
3 | 66 | Adrian Georgescu | First [[SipDeviceConfiguration|configure your SIP device]]. |
4 | 66 | Adrian Georgescu | |
5 | 79 | Adrian Georgescu | h2. Incoming calls |
6 | 66 | Adrian Georgescu | |
7 | 79 | Adrian Georgescu | 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. |
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9 | 66 | Adrian Georgescu | h2. Calling Out |
10 | 66 | Adrian Georgescu | |
11 | 1 | Adrian Georgescu | 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. |
12 | 61 | Adrian Georgescu | |
13 | 66 | Adrian Georgescu | > 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. |
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15 | 79 | Adrian Georgescu | h3. Test Numbers |
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17 | 66 | Adrian Georgescu | * To test audio sessions, call 3333, you should hear some music playing |
18 | 1 | Adrian Georgescu | * To test microphone call 4444, you should hear your echo back |
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20 | 79 | Adrian Georgescu | h3. Session Details |
21 | 72 | Adrian Georgescu | |
22 | 1 | Adrian Georgescu | * To review your SIP sessions go to https://mdns.sipthor.net/CDRTool |
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24 | 83 | Adrian Georgescu | h2. Conferencing |
25 | 83 | Adrian Georgescu | |
26 | 83 | Adrian Georgescu | * Using any SIP client connect to <room>@conference.sip2sip.info Replace the <room> with the desired room name. |
27 | 83 | Adrian Georgescu | * Using "Blink":http://icanblink.com SIP client go to menu Call -> Join Conference. Choose a room and connect. |
28 | 83 | Adrian Georgescu | * XMPP clients can connect to room@conference.sip2sip.info. Replace room with the desired room name. |
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30 | 93 | Adrian Georgescu | h3. Supported media |
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32 | 93 | Adrian Georgescu | * Audio codecs: Opus, Speex, G.722, G.711 |
33 | 93 | Adrian Georgescu | * Text Chat is supported for SIP clients using MSRP protocol |
34 | 93 | Adrian Georgescu | * File Transfers are supported for SIP clients using MSRP protocol |
35 | 93 | Adrian Georgescu | * SIP Clients that implement "Conference Event Package RFC 4575":http://tools.ietf.org/html/rfc4579 can retrieve the participants information |
36 | 93 | Adrian Georgescu | * Multiparty Text Chat is supported between XMPP and SIP MSRP clients |
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39 | 93 | Adrian Georgescu | h3. Conference Room Access |
40 | 93 | Adrian Georgescu | |
41 | 93 | Adrian Georgescu | Supported Audio codecs: |
42 | 93 | Adrian Georgescu | |
43 | 91 | Adrian Georgescu | | *Support Media*| *Address* | *Protocol* | *Recommended Clients*| |
44 | 88 | Adrian Georgescu | | Narrow band audio| +31208005161 | PSTN| Any Phone| |
45 | 88 | Adrian Georgescu | | RTP Audio | ROOM@conference.sip2sip.info | SIP| Blink, Bria| |
46 | 88 | Adrian Georgescu | | MSRP Chat, MSRP File Transfer| ROOM@conference.sip2sip.info | SIP| Blink for OSX| |
47 | 92 | Adrian Georgescu | | Conference Information | ROOM@conference.sip2sip.info | SIP| Blink for OSX| |
48 | 1 | Adrian Georgescu | | Group Chat| ROOM@conference.sip2sip.info| XMPP Muc| Jitsi, Adium, iChat, Google| |
49 | 1 | Adrian Georgescu | | Ultra-wideband Audio| ROOM@conference.sip2sip.info| XMPP Jingle |Jitsi| |
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52 | 93 | Adrian Georgescu | |
53 | 66 | Adrian Georgescu | h2. NAT Traversal |
54 | 1 | Adrian Georgescu | |
55 | 1 | Adrian Georgescu | 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: |
56 | 66 | Adrian Georgescu | |
57 | 66 | Adrian Georgescu | # Disable STUN as is unreliable and leading to unexpected results |
58 | 1 | Adrian Georgescu | # 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 |
59 | 66 | Adrian Georgescu | |
60 | 1 | Adrian Georgescu | 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: |
61 | 1 | Adrian Georgescu | |
62 | 1 | Adrian Georgescu | | *Port range* | *Protocol* | *Description* | *Application* | |
63 | 72 | Adrian Georgescu | | 5060 | UDP | SIP signaling | OpenSIPS - SIP Proxy/Registrar/Presence Agent | |
64 | 66 | Adrian Georgescu | | 5060 | TCP | SIP signaling | OpenSIPS - SIP Proxy/Registrar/Presence Agent | |
65 | 66 | Adrian Georgescu | | 5269 | TCP | XMPP signaling | SylkServer SIP/XMPP gateway | |
66 | 1 | Adrian Georgescu | | 50000:60000 | UDP | RTP media | !MediaProxy - RTP media relay | |
67 | 1 | Adrian Georgescu | | 2855 | TLS | MSRP media | MSRP relay - MSRP media relay | |
68 | 66 | Adrian Georgescu | | 443 | TLS | XCAP storage | OpenXCAP - Presence policy management | |
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72 | 81 | Adrian Georgescu | h2. Voicemail |
73 | 81 | Adrian Georgescu | |
74 | 69 | Adrian Georgescu | |
75 | 69 | Adrian Georgescu | * To access your voicemail or mailbox settings dial 1233 |
76 | 69 | Adrian Georgescu | * Your voice messages are delivered to your e-mail address |
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78 | 70 | Adrian Georgescu | |
79 | 70 | Adrian Georgescu | h2. IM and File Transfer |
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82 | 31 | Adrian Georgescu | * For instant messaging your client must support MSRP protocol and its MSRP relay extension |
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84 | 66 | Adrian Georgescu | |
85 | 66 | Adrian Georgescu | h2. Presence |
86 | 57 | Adrian Georgescu | |
87 | 59 | Adrian Georgescu | SIP2SIP provides a SIP presence agent that handles SUBSCRIBE and PUBLISH methods for presence events. |
88 | 66 | Adrian Georgescu | |
89 | 66 | Adrian Georgescu | # To publish your presence send PUBLISH for *Event: presence* to your own SIP address, containing the body describing your presence information in PIDF format |
90 | 66 | Adrian Georgescu | # To subscribe to a SIP address, send a SUBSCRIBE message for *Event: presence* |
91 | 66 | Adrian Georgescu | # 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* |
92 | 66 | Adrian Georgescu | # To monitor who has subscribed to your presence information you must send SUBSCRIBe for *Event: presence.winfo* to your own SIP address |
93 | 66 | Adrian Georgescu | # To allow others to subscribe to your published information you must use XCAP protocol for manipulating *pres-rules* policy document |
94 | 61 | Adrian Georgescu | # To store your buddy list on the server you must use XCAP protocol for manipulating *resource-lists* and *rls-services* documents |
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96 | 80 | Adrian Georgescu | More information is available at http://wiki.sip2sip.info/news/23 |
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98 | 66 | Adrian Georgescu | |
99 | 65 | Adrian Georgescu | h2. Calls to PSTN network |
100 | 76 | Adrian Georgescu | |
101 | 66 | Adrian Georgescu | 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. |
102 | 66 | Adrian Georgescu | |
103 | 66 | Adrian Georgescu | 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 |
104 | 61 | Adrian Georgescu | |
105 | 61 | Adrian Georgescu | 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. |
106 | 61 | Adrian Georgescu | |
107 | 61 | Adrian Georgescu | 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. |
108 | 61 | Adrian Georgescu | |
109 | 61 | Adrian Georgescu | To limit fraud in case of lost account credentials, a maximum of 2 simultaneous calls are permitted. |
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111 | 61 | Adrian Georgescu | *Important notes* |
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113 | 66 | Adrian Georgescu | * Service numbers for premium services may not be reachable |
114 | 1 | Adrian Georgescu | * Emergency access number (e.g. 911, 112) are not available |
115 | 66 | Adrian Georgescu | * Not all international PSTN prefixes may be available depending on the capabilities of our outbound carriers |
116 | 66 | Adrian Georgescu | |
117 | 66 | Adrian Georgescu | h2. Calls from PSTN network |
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119 | 1 | Adrian Georgescu | |
120 | 1 | Adrian Georgescu | 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. |