Differences between current version and revision by previous author of VoIP.
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| Newer page: | version 55 | Last edited on Thursday, August 11, 2005 10:47:05 am | by AdrianKitto | |
| Older page: | version 53 | Last edited on Sunday, June 12, 2005 3:42:38 pm | by GregSnover | Revert |
@@ -1,101 +1,4 @@
-[Acronym] for __V__oice __o__ver __IP__
+[Acronym] for __V__oice __o__ver __IP__ see [VoiceOverIP]
Known to be pronounced __''voy-oop''__ (What will the marketing/sales people come up with next ?)%%%
''I strongly recommend using one those paper bags found in front of you when you are seated in a passager aircraft when you practice pronouncing this acronym.''
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-The two major protocols used for VoIP are [SIP] and [H.323]. (They are completely different standards, not related to each other). SIP isn't so much a complete VoIP protocol - other protocols are required for the actual voice data transmission. See the [SIP] page for more details.
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-[Cisco] also deploy [MGCP] and [Skinny] on their AVVID solution.
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-See http://voip.fast.co.nz , a NZ voip PABX you can connect to and make free calls.
-
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-!!Free Linux/Win32 VoIP Client/Server
-This client/server works for both Win32/Linux. It's a great PC to PC communication program.
-*http://www.goteamspeak.com/
-
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-!!Free SIP clients
-!Linux
-*http://www.zultys.co.nz/LIPZ4.htm http://www.zultys.com/products/lipz4/softphone-1.3.11-0.i386.rpm
-* [Kphone|http://www.wirlab.net/kphone]
-* [Linphone|http://www.linphone.org/?lang=us] - [Free] ([GPL]d) and [GTK]2. (as of Sep 2004 this is now in Debian Sarge and Sid (unstable))
-* [SJPhone|http://www.sjlabs.com] SJ Labs VOIP software
-!Windows
-*Xlite - http://www.xten.com/index.php?menu=products&smenu=xlite
-*FireFly - http://www.virbiage.com/firefly/
-
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-!!Hardware Based VoIP Products
-!Cisco
-*[VoIP Gateways|http://www.cisco.com/en/US/products/hw/gatecont/index.html]%%%
-*[VoIP Phones|http://www.cisco.com/en/US/products/hw/phones/ps379/index.html]%%%
-*[VoIP Software|http://www.cisco.com/en/US/products/sw/voicesw/index.html]%%%
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-__Approved FWD Cisco SIP Clients__
-*[Cisco ATA-186 2 Port Analog Telephone Adaptor|http://www.cisco.com/en/US/products/hw/gatecont/ps514/products_data_sheet09186a008007cd72.html]
-**[FWD Configuration Guide|http://www.freeworldialup.com/support/configuration_guide/configure_your_fwd_certified_phone/cisco_ata_186]
-*[Cisco 7960 IP Phone|http://www.cisco.com/en/US/products/hw/phones/ps379/ps1855/index.html]
-**[FWD Configuration Guide|http://www.freeworldialup.com/support/configuration_guide/configure_your_fwd_certified_phone/cisco_7960]
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-__[Example of a SIP gateway using a Cisco ATA-186 connected through a small PBX|http://www.lindsay.wired.net.nz/projects/sip/]__
-
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-!!Server Software
-
-[Asterisk]
-
-
-!GNUGK
-http://www.openh323.org/
-GnuGK and friends make up the openh323 project. GK is an h.323 gatekeeper (central server thing).
-
-!SIP Express Router
-http://www.iptel.org/ser/
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-A more 'hardcore' SIP server. Less features, more commercial. SIP supports textual messaging, so SER has a [Jabber] gateway.
-
-* PCH runs this, (think INOC-DBA).
-* sipphone.com run this.
-* IPTel run this.
-
-!OnDO SIP Server and OnDO IPPBX by Brekeke
-http://www.brekeke.com
-
-Very easy to use, free for personal use and actually works quite well (And Windows versions also) Alas, no presence (i.e. no IM (sigh))
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-!Microsoft Office Live Communications Server
-http://www.microsoft.com/office/livecomm/prodinfo/default.mspx
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-SIP, H.323, etc. Nice and intergrated.
-
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-!!Free ENUM DNS services
-
-E164 ([E164.org]) http://www.e164.org/
-
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-!!NOTES
-Clients can be connected to more than one service at a time, think of it as having 2 separate phone lines, both connected to different providers but both phones plugging into one unit.. so feel free to sign up for accounts at more than one service.. you can select which service you want when dialling away from the default service by starting to dial your numbers with #2 for the second account you have setup etc.
-
-
-SIP is the new protocol used to initiate communication between various units, and it is a standard met by clients on Linux, Windows, or on stand-alone VoIP phones. This means you can call any user on the network not caring what kind of hardware they are using, the SIP should help start a common conversation so you can talk to each other.
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-To try it out,
-get yourself a client from the links listed at the top of this page, if you find any other clients please add them to this list, or if you want to point out problems or good points add them to this list also.
-Then register with a service listed on the SipServices page.
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-If you have a public IP address with the right ports unfirewalled, people can also connect directly using asip://username@ip.add.re.ss
-[URI].
-
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-!! Also of possible interest
-
-* http://www.digium.com - This crowd produce a PCI PSTN gateway card, no idea at this point as to wether it works in New Zealand. Of more interest however is their PBX gateway software below
-* http://www.asterisk.org - This is an Opensource PBX gateway that supports VoIP <-> PSTN mappings as well as SIP, H.323 and a rather large number of codecs.
-* http://www.pingtel.com - A VoIP handset phone
-* http://www.snom.com - Another VoIP handset, this site is great tho, lots of VoIP resources. AND their phones use linux as the base OS !
-* http://www.sipcenter.com/ - Everything you didn't want to know about SIP (Session Initiation Protocol)
-
-From the looks of things, the Quicknet LineJACK cards also work in New Zealand - they are currently being deployed by a crowd in christchurch, Telepermiting of these cards is unknown at this stage (waiting for a reply from the telepermit guys)
