SipTesting

Version 36 (Adrian Georgescu, 04/21/2009 01:41 pm)

1 8 Adrian Georgescu
= Using SIP = 
2 1 Adrian Georgescu
3 2 Adrian Georgescu
[[TOC(WikiStart, Sip*, depth=3)]]
4 2 Adrian Georgescu
5 16 Adrian Georgescu
SIP2SIP service is NAT traversal proof for SIP signaling, RTP and MSRP media. 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.
6 9 Adrian Georgescu
7 22 Adrian Georgescu
You need un-restricted access to the following ports used by SIP2SIP infrastructure:
8 10 Adrian Georgescu
9 12 Adrian Georgescu
|| '''Port range''' || '''Protocol''' || '''Description''' || '''Application''' ||
10 15 Adrian Georgescu
|| 5060 || UDP || SIP signaling || OpenSIPS - SIP Proxy/Registrar/Presence Agent ||
11 15 Adrian Georgescu
|| 50000:60000 || UDP || RTP media || !MediaProxy  - RTP media relay ||
12 20 Adrian Georgescu
|| 2855 || TLS || MSRP media || MSRP relay - MSRP media relay ||  
13 20 Adrian Georgescu
|| 443 || TLS || XCAP storage || OpenXCAP - Presence policy management  ||
14 10 Adrian Georgescu
15 25 Adrian Georgescu
== Your SIP address ==
16 23 Adrian Georgescu
17 28 Adrian Georgescu
When you register a SIP account on SIP2SIP, a SIP address under @sip2sip.info domain is automatically allocated to you.
18 26 Adrian Georgescu
19 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.
20 26 Adrian Georgescu
21 26 Adrian Georgescu
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.
22 1 Adrian Georgescu
23 35 Adrian Georgescu
First [wiki:SipDeviceConfiguration configure your SIP device].
24 24 Adrian Georgescu
25 35 Adrian Georgescu
== Internet sessions ==
26 2 Adrian Georgescu
27 1 Adrian Georgescu
 * To test audio sessions, call 3333, you should hear some music playing 
28 1 Adrian Georgescu
 * To test microphone call 4444, you should hear your echo back 
29 1 Adrian Georgescu
 * You may call to any other SIP account user@domain 
30 2 Adrian Georgescu
31 29 Adrian Georgescu
=== Voicemail === 
32 29 Adrian Georgescu
33 1 Adrian Georgescu
 * To access your voicemail or mailbox settings dial *70 
34 1 Adrian Georgescu
 * Your voice messages are delivered to ag@ag-projects.com 
35 2 Adrian Georgescu
36 31 Adrian Georgescu
=== PSTN outbound  ===
37 31 Adrian Georgescu
38 31 Adrian Georgescu
To make calls to PSTN destinations add credit to your SIP account at http://x.sip2sip.info?tab=prepaid
39 31 Adrian Georgescu
40 31 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. To use the PSTN gateway you must have a positive credit. To add credit for your account login to the SIP settings page and click on Credit tab.
41 31 Adrian Georgescu
42 31 Adrian Georgescu
Price list for dialing to PSTN destinations is available [https://secure.dns-hosting.info/sip2sip_rates.html here]. The call costs are logged in the Credit section of your [http://x.sip2sip.info SIP settings page].
43 31 Adrian Georgescu
44 31 Adrian Georgescu
=== PSTN inbound  ===
45 31 Adrian Georgescu
46 31 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 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.arap 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.
47 31 Adrian Georgescu
48 31 Adrian Georgescu
[[Image(msp-enum-lookup.png)]]
49 31 Adrian Georgescu
50 29 Adrian Georgescu
=== Session details ===
51 2 Adrian Georgescu
52 1 Adrian Georgescu
 * To review your SIP sessions go to https://secure.dns-hosting.info/CDRTool 
53 1 Adrian Georgescu
54 34 Adrian Georgescu
[[Image(sip2sip-sessions-search.png,link=http://cdrtool.ag-projects.com)]]
55 29 Adrian Georgescu
56 17 Adrian Georgescu
== Presence ==
57 2 Adrian Georgescu
58 36 Adrian Georgescu
SIP2SIP provides you with a SIP SIMPLE presence agent.
59 2 Adrian Georgescu
60 2 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 XML format
61 2 Adrian Georgescu
 1. To subscribe to a SIP address, send a SUBSCRIBE message for '''Event: presence'''
62 2 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'''
63 2 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
64 2 Adrian Georgescu
 1. To allow others to subscribe to your published information you must XCAP protocol for manipulating '''pres-rules''' policy document 
65 1 Adrian Georgescu
 1. To store your buddy list on the server you must XCAP protocol for manipulating '''resource-lists''' and '''rls-services''' documents
66 2 Adrian Georgescu
67 34 Adrian Georgescu
[[Image(http://openxcap.org/raw-attachment/wiki/WikiStart/SIMPLE-Server.png,link=http://openxcap.org)]]
68 1 Adrian Georgescu
69 33 Adrian Georgescu
== IM and file transfer ==
70 1 Adrian Georgescu
71 32 Adrian Georgescu
 * For instant messaging your client must support MSRP protocol and its MSRP relay extension
72 1 Adrian Georgescu
 * 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. 
73 1 Adrian Georgescu
 * 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].
74 33 Adrian Georgescu
75 34 Adrian Georgescu
[[Image(http://msrprelay.org/chrome/site/files/MSRP-Relay-Scenarios.png,link=http://msrprelay.org)]]