Difference between revisions of "VoIP Cookbook: Test your SIP Softphone"
Onnowpurbo (talk | contribs) |
|||
| (One intermediate revision by one other user not shown) | |||
| Line 1: | Line 1: | ||
| − | Now that you have adjusted both | + | Now that you have adjusted both [[softphone]]s (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. | + | 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. |
[[Image:Voiprakyat-service1.jpeg|right|200px|thumb|Just like other VoIP Providers, VoIP Rakyat provides its users with some numbers with which the users can use for testing their VoIP quality]] | [[Image:Voiprakyat-service1.jpeg|right|200px|thumb|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: | + | 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 | + | 901 - which indicates the time Jakarta's time and nearby countries. |
902 - noise | 902 - noise | ||
903 - echo test | 903 - echo test | ||
| + | 904 - record caller voice and playback to listen to the VoIP call quality | ||
[[Image:Voiprakyat-service2.jpeg|left|200px|thumb|Through VoIP Rakyat's Phonebook, you can see who's online]] | [[Image:Voiprakyat-service2.jpeg|left|200px|thumb|Through VoIP Rakyat's Phonebook, you can see who's online]] | ||
| Line 15: | Line 16: | ||
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. | 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: | + | 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: |
Latest revision as of 00:55, 21 August 2010
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.
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
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 |