Project

General

Profile

SipTesting » History » Version 10

Adrian Georgescu, 04/19/2009 02:14 PM

1 8 Adrian Georgescu
= Using SIP = 
2 1 Adrian Georgescu
3 2 Adrian Georgescu
[[TOC(WikiStart, Sip*, depth=3)]]
4
5 9 Adrian Georgescu
The service is NAT traversal proof for SIP signaling, RTP and MSRP media. Beware that corporate firewalls that have an explicit policy against SIP or [http://www.voip-info.org/wiki/view/Routers+SIP+ALG SIP ALGs] may still block your SIP traffic.
6
7 10 Adrian Georgescu
You need outgoing access to the following ports used by SIP2SIP infrastructure:
8
9
|| '''Port range''' || '''Protocol''' || '''Description''' ||
10
|| 5060 || UDP || SIP signaling ||
11
|| 443 || TLS || XCAP storage ||
12
|| 2855 || TLS || MSRP relay ||
13
|| 50000:60000 || UDP || RTP media ||
14
15
16 2 Adrian Georgescu
== Internet calls ==
17
18 1 Adrian Georgescu
 * To test audio sessions, call 3333, you should hear some music playing 
19
 * To test microphone call 4444, you should hear your echo back 
20
 * You may call to any other SIP account user@domain 
21 2 Adrian Georgescu
22
== Voicemail ==
23
 
24 1 Adrian Georgescu
 * To access your voicemail or mailbox settings dial *70 
25
 * Your voice messages are delivered to ag@ag-projects.com 
26 2 Adrian Georgescu
27
== Call detail records ==
28
29 1 Adrian Georgescu
 * To review your call go to https://secure.dns-hosting.info/CDRTool 
30 2 Adrian Georgescu
31 6 Adrian Georgescu
[[Image(sip2sip-sessions-search.png)]]
32 4 Adrian Georgescu
33
34 2 Adrian Georgescu
== Multi-party IM ==
35
36 1 Adrian Georgescu
 * To test multi-party instant messaging using MSRP setup a MSRP session to XYZ@chatserver.ag-projects.com 
37 2 Adrian Georgescu
38
== PSTN outbound  ==
39
40
To make calls to PSTN destinations add credit to your SIP account at http://x.sip2sip.info?tab=prepaid
41
42
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.
43
44
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].
45
46
== PSTN inbound  ==
47
48
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.
49 3 Adrian Georgescu
50
[[Image(msp-enum-lookup.png)]]