VoIP Cookbook: Test your SIP Softphone

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Now that you have adjusted both softphones (or just one of them), the next important thing you have to do is to test whether it could run properly. Note that the quality of the voice produced by the softphone during the test may have been just fine, but when your softphone is connected to a VoIP provider, the voice quality could be poor, depending on many other things such as bandwidth availability and the type of codec run by the softphone. For this test purpose, VoIP providers usually provide the telephone number to which you can dial.

If your computer is connected to an internet behind a firewall, the firewall might block your connectivity. In order to make your VoIP connectivity working behind the firewall, you have to open Port 5060-6060 to enable Session Initiation Protocol (SIP) and Port 8000-20000 for voice data delivery using Real Time Protocol (RTP). But if you're not sure what to do, you can simply ask your network administrator to do what is told here.

Just like other VoIP Providers, VoIP Rakyat provides its users with some numbers with which the users can use for testing their VoIP quality
Just like other VoIP Providers, VoIP Rakyat provides its users with some numbers with which the users can use for testing their VoIP quality

Go to VoIP Rakyat's Service Number page, http://voiprakyat.or.id/services/. This page provides you with some numbers that can be used to test your VoIP connection and their functions. Some of them are:

901 - which indicates the time Jakarta's time and nearby countries.
902 - noise
903 - echo test
904 - record caller voice and playback to listen to the VoIP call quality 
Through VoIP Rakyat's Phonebook, you can see who's online
Through VoIP Rakyat's Phonebook, you can see who's online

In testing this connectivity, what users will often do is to call anyone found online in http://www.voiprakyat.or.id/?inc=online_phones. So don't be surprised if someone dials your number. Depending on where the users are, the call comes from a variety of countries, including the U.S.

There are of course other VoIP phone numbers which you can use to test your VoIP connection. These are provided in a long list available in http://www.voip-info.org/wiki/view/Phone+Numbers. If you want to call using SIP address format (sip@domain.com), the following is a table of some numbers you may use:


Function SIP Provider SIP Enum
Autoattendant BC Wireless (http://www.bcwireless.net/moin.cgi/NetworkServices/VoiceServices/PublicConferenceRoom). 1000@mutual.bcwireless.net 1 604 484 5289 x8600 through E164.org
Enum2go (http://enum2go.com/) 878107472000010@sip2go.com
Echo Test N3 Network Lab. (http://www.n3network.ch/) Echo test sip: echo@n3network.ch and sip: 905100@n3network.ch (no G.729)
Mouselike.org (UK) (http://www.mouselike.org/) 904@mouselike.org +441483604781
VoipTalk UK (http://www.voiptalk.org/) 904@voiptalk.org
Reread Called ID 95861111@mutual.bcwireless.net
Welcome Line FWD 55555@fwd.pulver.com
Ewing IT 611300766674@sip.like2fone.com
Xmission (http://xmission.com/transmission) xmission@pbx.xmission.com (tidak ada G.729)
UCLA (http://internet2.edu/sip.edu) 13108254321@ucla.edu (tidak ada G.729)
TELL 18005558355@proxy01.sipphone.com
U. Philippines 0116329818500@proxy01.sipphone.com
Personal Telco (http://wiki.personaltelco.net/moin.cgi/SipPhoneDirectory) 274185@fwd.pulver.com
Patton Electronics (http://www.patton.com/support) support@patton.com (tidak ada G.729)
Party Line 17475552663@proxy01.sipphone.com (VoIP conference setiap sabtu jam 20:00 GMT)
Ingate (http://www.ingate.com/trysip.php) music@trysip.ingate.com
MIT (http://sipphone.com/numbers) 16172531000@proxy01.sipphone.com


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