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This eliminates the need to have two separate applications open for basic communication tasks. An optional package providing support for the iLBC codec is there as well.Īlong with meeting your VoIP needs, Linphone also has a simple instant messenger application built in.

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Linphone requires the libosip2 package, which is included in Linphone’s download directory.

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Linphone is distributed only as source code no binary packages are available from the project. Linphone is more than four years old, making it the grandfather of the bunch. I opted for the tarball, and installation went without a hitch. Kiax has packages available for Debian, Red Hat, and SUSE, as well as a plain tarball with the Kiax binary. Kiax can talk to SIP clients only through an Asterisk server. If there is an Asterisk server between Kiax (or another IAX client) and a SIP client, everything will work out.

The major drawback for Kiax is its lack of SIP support. There is not much more to Kiax’s interface simplicity is a good thing sometimes. A nice touch is the ability to sort calls in the register based on whether they were incoming, outgoing, or missed. Kiax sports a call register that lists all calls made and received. What makes Kiax stand out is its simple interface. Let’s take a look at what does set these softphones apart. Under these conditions, they all delivered roughly the same voice quality. Because we did our testing on a switched 100Mbps LAN - in other words, a very fast network - we used the G.711u codec with all the products, and found voice quality was not a distinguishing issue. G.711u provides high voice quality but requires high bandwidth. The codec that gives the highest voice quality is the G.711u, which is the standard codec that traditional plain old telephone service (POTS) providers use. The choice of codec depends on the bandwidth available.

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In addition to a signalling protocol, VoIP endpoints need to specify codec software that turns analog voice communication into digital packets for transmission over the network and back again at the receiver’s end. IAX softphones work behind firewalls without the need for external proxy servers or the need to change firewall settings. The main advantage of IAX over SIP is its transparency to firewalls. It uses Digium’s Inter Asterisk Exchange (IAX) protocol. With its IETF backing, SIP is quickly becoming the standard protocol for VoIP. Three out of the four support Session Initiation Protocol (SIP), a signalling protocol under development by the Internet Engineering Task Force (IETF) to establish VoIP connections.

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X-Lite is not, but is available as a free download. Kiax, Linphone, and Twinkle are open source. I tested the four Linux-based programs using an Asterisk server and multiple Linux workstations on an internal LAN. If equipment costs are stopping you from experimenting with VoIP, softphones can provide an inexpensive way for businesses to get up and running with VoIP, as I recently discovered by putting Kiax, Linphone, Twinkle, and CounterPath’s X-Lite to the test.Ī softphone runs on your computer and provides all the features and functionality of a regular phone. VoIP systems require IP phones or analog telephone adapters to allow your existing phones to work. Nmap done: 1 IP address (1 host up) scanned in 0.Many businesses are turning to Voice over IP (VoIP) to save money on infrastructure and communications costs, but just ripping out your existing phone system and replacing it with VoIP will not work. Interesting ports on localhost (127.0.0.1): Warning: Hostname localhost resolves to 2 IPs. I ran an nmap on localhost and found the following blaine :: /etc/asterisk » nmap localhost System details: blaine :: /etc/asterisk » uname -a I'm just trying to get Asterisk to work with a basic setup so I can experiment with it.

When I add an account it says Could not register I've also tried setting the registrar in Ekiga to 127.0.0.1. My Asterisk server is running on my desktop, as well as my softphone (Ekiga). I'm up to the point where I've configured sip.conf and nf according to the book.Įxten => s,1,Verbose(1|Unrouted call handler)Įxten => 500,1,Verbose(1|Echo test application) I'm familiarizing myself with Asterisk, and I'm reading the book 'Asterisk: The Future of Telephony.'