SipTesting

Version 61 (Adrian Georgescu, 09/06/2011 09:57 am)

1 45 Adrian Georgescu
= Using SIP =
2 45 Adrian Georgescu
3 61 Adrian Georgescu
[[TOC(WikiStart, Sip*, depth=3)]]
4 45 Adrian Georgescu
5 61 Adrian Georgescu
First [wiki:SipDeviceConfiguration configure your SIP device].
6 1 Adrian Georgescu
7 61 Adrian Georgescu
== Calling Out ==
8 1 Adrian Georgescu
9 61 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.
10 55 Adrian Georgescu
11 61 Adrian Georgescu
== Test Numbers  ==
12 1 Adrian Georgescu
13 61 Adrian Georgescu
 * To test audio sessions, call 3333, you should hear some music playing 
14 61 Adrian Georgescu
 * To test microphone call 4444, you should hear your echo back 
15 55 Adrian Georgescu
16 61 Adrian Georgescu
== Session Details ==
17 55 Adrian Georgescu
18 61 Adrian Georgescu
 * To review your SIP sessions go to https://mdns.sipthor.net/CDRTool 
19 1 Adrian Georgescu
20 1 Adrian Georgescu
== NAT Traversal == 
21 1 Adrian Georgescu
22 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:
23 1 Adrian Georgescu
24 47 Adrian Georgescu
 1. Disable STUN as is unreliable  and leading to unexpected results
25 1 Adrian Georgescu
 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]
26 60 Adrian Georgescu
27 47 Adrian Georgescu
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:
28 47 Adrian Georgescu
29 12 Adrian Georgescu
|| '''Port range''' || '''Protocol''' || '''Description''' || '''Application''' ||
30 15 Adrian Georgescu
|| 5060 || UDP || SIP signaling || OpenSIPS - SIP Proxy/Registrar/Presence Agent ||
31 15 Adrian Georgescu
|| 50000:60000 || UDP || RTP media || !MediaProxy  - RTP media relay ||
32 41 Adrian Georgescu
|| 2855 || TLS || MSRP media || MSRP relay - MSRP media relay ||  
33 60 Adrian Georgescu
|| 443 || TLS || XCAP storage || OpenXCAP - Presence policy management  ||
34 40 Adrian Georgescu
35 42 Adrian Georgescu
== SIP Address ==
36 42 Adrian Georgescu
37 61 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. 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.
38 1 Adrian Georgescu
39 1 Adrian Georgescu
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.
40 1 Adrian Georgescu
41 61 Adrian Georgescu
== Voicemail ==
42 1 Adrian Georgescu
43 1 Adrian Georgescu
 * To access your voicemail or mailbox settings dial 1233
44 1 Adrian Georgescu
 * Your voice messages are delivered to your e-mail address 
45 1 Adrian Georgescu
46 61 Adrian Georgescu
== IM and File Transfer ==
47 1 Adrian Georgescu
48 1 Adrian Georgescu
 * For instant messaging your client must support MSRP protocol and its MSRP relay extension
49 1 Adrian Georgescu
50 61 Adrian Georgescu
== Conferencing ==
51 42 Adrian Georgescu
52 48 Adrian Georgescu
To test the conference server send an INVITE with any of the supported media to sip:test@conference.sip2sip.info You may replace test with any username you wish, a conference room will be created on the fly. To obtain the conference information send a SUBSCRIBE request for Event conference RFC 4575. The NOTIFY contains detailed information with the list of participants, their connected endpoints, media type and stream status. To add or remove participants, the server support INVITE and REFER methods as defined in RFC4579.
53 31 Adrian Georgescu
54 31 Adrian Georgescu
See http://sylkserver.com/applications.phtml for more information about conferencing capabilities.
55 31 Adrian Georgescu
56 31 Adrian Georgescu
== Presence ==
57 49 Adrian Georgescu
58 49 Adrian Georgescu
SIP2SIP provides a SIP presence agent that handles SUBSCRIBE and PUBLISH methods for presence events.
59 49 Adrian Georgescu
60 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
61 57 Adrian Georgescu
 1. To subscribe to a SIP address, send a SUBSCRIBE message for '''Event: presence'''
62 57 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 59 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 1 Adrian Georgescu
 1. To allow others to subscribe to your published information you must use XCAP protocol for manipulating '''pres-rules''' policy document 
65 58 Adrian Georgescu
 1. To store your buddy list on the server you must use XCAP protocol for manipulating '''resource-lists''' and '''rls-services''' documents
66 58 Adrian Georgescu
67 61 Adrian Georgescu
== Calls to PSTN network  ==
68 61 Adrian Georgescu
69 61 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 [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]. To add credit to your SIP account at http://x.sip2sip.info?tab=credit 
70 61 Adrian Georgescu
71 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. 
72 61 Adrian Georgescu
73 61 Adrian Georgescu
To set your caller id please open a ticket in the support interface.
74 61 Adrian Georgescu
75 61 Adrian Georgescu
'''Important notes'''
76 61 Adrian Georgescu
77 61 Adrian Georgescu
 * Service numbers for premium services may not be reachable
78 61 Adrian Georgescu
 * Emergency access number (e.g. 911, 112) are not available
79 61 Adrian Georgescu
 * Not all international PSTN prefixes may be available depending on the capabilities of our outbound carriers
80 61 Adrian Georgescu
 
81 61 Adrian Georgescu
== Calls from PSTN network  ==
82 61 Adrian Georgescu
83 61 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.