SipTesting » History » Version 54
Adrian Georgescu, 08/11/2010 05:38 PM
1 | 45 | Adrian Georgescu | = Using SIP = |
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4 | 48 | Adrian Georgescu | == NAT Traversal == |
5 | 45 | Adrian Georgescu | |
6 | 2 | Adrian Georgescu | [[TOC(WikiStart, Sip*, depth=3)]] |
7 | 1 | Adrian Georgescu | |
8 | 47 | 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: |
9 | 1 | Adrian Georgescu | |
10 | 47 | Adrian Georgescu | 1. Disable any keep-alive technique like PING, NOTIFY, OPTION, CR/LF or STUN |
11 | 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] |
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12 | 1. Disable voice activation detection (VAD) in your device |
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14 | 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: |
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15 | 10 | Adrian Georgescu | |
16 | 12 | Adrian Georgescu | || '''Port range''' || '''Protocol''' || '''Description''' || '''Application''' || |
17 | 15 | Adrian Georgescu | || 5060 || UDP || SIP signaling || OpenSIPS - SIP Proxy/Registrar/Presence Agent || |
18 | || 50000:60000 || UDP || RTP media || !MediaProxy - RTP media relay || |
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19 | 20 | Adrian Georgescu | || 2855 || TLS || MSRP media || MSRP relay - MSRP media relay || |
20 | || 443 || TLS || XCAP storage || OpenXCAP - Presence policy management || |
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21 | 10 | Adrian Georgescu | |
22 | 48 | Adrian Georgescu | == SIP Address == |
23 | 23 | Adrian Georgescu | |
24 | 48 | Adrian Georgescu | 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. |
25 | 26 | Adrian Georgescu | |
26 | 28 | Adrian Georgescu | 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. |
27 | 26 | Adrian Georgescu | |
28 | 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. |
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29 | 1 | Adrian Georgescu | |
30 | 35 | Adrian Georgescu | First [wiki:SipDeviceConfiguration configure your SIP device]. |
31 | 24 | Adrian Georgescu | |
32 | 48 | Adrian Georgescu | == Internet Sessions == |
33 | 2 | Adrian Georgescu | |
34 | 1 | Adrian Georgescu | * To test audio sessions, call 3333, you should hear some music playing |
35 | * To test microphone call 4444, you should hear your echo back |
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36 | * You may call to any other SIP account user@domain |
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37 | 2 | Adrian Georgescu | |
38 | 29 | Adrian Georgescu | === Voicemail === |
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40 | 52 | Adrian Georgescu | * To access your voicemail or mailbox settings dial 1233 |
41 | 39 | Luci Stanescu | * Your voice messages are delivered to your e-mail address |
42 | 2 | Adrian Georgescu | |
43 | 48 | Adrian Georgescu | === PSTN Outbound === |
44 | 31 | Adrian Georgescu | |
45 | 54 | Adrian Georgescu | Price list for dialing to PSTN destinations is available [https://mdns.sipthor.net/sip_rates.html here]. The call costs are logged in the Credit section of your [http://x.sip2sip.info SIP settings page]. |
46 | 31 | Adrian Georgescu | |
47 | 51 | Adrian Georgescu | 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=credit |
48 | 31 | Adrian Georgescu | |
49 | 41 | Adrian Georgescu | 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. |
50 | 31 | Adrian Georgescu | |
51 | 40 | Adrian Georgescu | To set your caller id please send an e-mail to support address. |
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53 | 42 | Adrian Georgescu | '''Important notes''' |
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55 | * Service numbers for premium services may not be reachable |
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56 | 1 | Adrian Georgescu | * Emergency access number (e.g. 911, 112) are not available |
57 | 53 | Adrian Georgescu | * Not all international PSTN prefixes may be available depending on the capabilities of our outbound carriers |
58 | 42 | Adrian Georgescu | |
59 | 48 | Adrian Georgescu | === PSTN Inbound === |
60 | 31 | Adrian Georgescu | |
61 | 39 | Luci Stanescu | 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. |
62 | 31 | Adrian Georgescu | |
63 | [[Image(msp-enum-lookup.png)]] |
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65 | 49 | Adrian Georgescu | === IM and File Transfer === |
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67 | * For instant messaging your client must support MSRP protocol and its MSRP relay extension |
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68 | * 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. |
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69 | * 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]. |
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71 | [[Image(http://msrprelay.org/chrome/site/files/MSRP-Relay-Scenarios.png,link=http://msrprelay.org)]] |
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73 | 48 | Adrian Georgescu | === Session Details === |
74 | 2 | Adrian Georgescu | |
75 | 54 | Adrian Georgescu | * To review your SIP sessions go to https://mdns.sipthor.net/CDRTool |
76 | 1 | Adrian Georgescu | |
77 | 34 | Adrian Georgescu | [[Image(sip2sip-sessions-search.png,link=http://cdrtool.ag-projects.com)]] |
78 | 29 | Adrian Georgescu | |
79 | 17 | Adrian Georgescu | == Presence == |
80 | 2 | Adrian Georgescu | |
81 | 46 | Adrian Georgescu | SIP2SIP provides a SIP presence agent that handles SUBSCRIBE and PUBLISH methods for presence events. |
82 | 2 | Adrian Georgescu | |
83 | 1 | Adrian Georgescu | 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 |
84 | 48 | Adrian Georgescu | 1. To subscribe to a SIP address, send a SUBSCRIBE message for '''Event: presence''' |
85 | 1 | Adrian Georgescu | 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''' |
86 | 32 | Adrian Georgescu | 1. To monitor who has subscribed to your presence information you must send SUBSCRIBe for '''Event: presence.winfo''' to your own SIP address |
87 | 1 | Adrian Georgescu | 1. To allow others to subscribe to your published information you must use XCAP protocol for manipulating '''pres-rules''' policy document |
88 | 1. 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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89 | 33 | Adrian Georgescu | |
90 | 34 | Adrian Georgescu | [[Image(http://openxcap.org/raw-attachment/wiki/WikiStart/SIMPLE-Server.png,link=http://openxcap.org)]] |