View Source: VoiceOverIP - Waikato Linux Users Group

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[AKA] [VoIP]

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.

[Cisco] also deploy [MGCP] and [Skinny] on their AVVID solution.
Digium also deploys [IAX2] (Inter-Asterisk eXchange] for their [Asterisk] software.

See http://voip.fast.co.nz , a NZ voip PABX you can connect to and make free calls.

The New Zealand Dialplan is available on the Number Administration Deed website http://www.nad.org.nz (Note that there are conflicts between the 2 documents published on their website)

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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
*[Qutecom|http://www.qutecom.org] formerly OpenWengo include Jabber/MSN/Yahoo chat
*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
* Skype - [http://skype.com]. Closed source, See https://wiki.ubuntu.com/SkypeHowto
!Windows
* Xlite - http://www.xten.com/index.php?menu=products&smenu=xlite
* FireFly - http://www.virbiage.com/firefly/
* Skype - http://skype.com
* 3CX VOIP Phone - http://www.3cx.com/VOIP/voip-phone.html
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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]%%%

__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]

__[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/

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.
* Supports both TCP and UDP SIP

!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))

!Microsoft Office Live Communications Server
http://www.microsoft.com/office/livecomm/prodinfo/default.mspx

SIP (TCP Only), H.323, etc. Nice and integrated.

!3CX Phone System
http://www.3cx.com/phone-system/

SIP based IP PBX - currently Windows, Linux version planned for 2007


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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.

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.

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 - Authors of [Asterisk] and producers of hardware VoIP cards, such as PCI PSTN gateway cards. See [Asterisk] for information about using then in NZ.
* http://www.asterisk.org - Digium's OpenSource PBX software 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)
* http://www.voip-info.org - Great resource for phone devices, configurations and [Asterisk] configuration options

From the looks of things, the Quicknet LineJACK cards also work in New Zealand - they are currently being deployed by a crowd in christchurch, Telepermitting of these cards is unknown at this stage (waiting for a reply from the telepermit guys)
5 pages link to VoiceOverIP:
Last edited on Monday, August 31, 2009 8:47:39 pm by "ChrisOh"
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