Softphones
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1. The user selects an account from the drop down menu Iaxcomm can connect to multiple servers at the same time and handle up to 100 connec tions The GUI displays a 2 Outgoing call 159853 PMA COVER STORY Softphones Unfortunately iaxComm is limited to the IAX2 protocol which won t help it to become widely accepted However if a company is already using Asterisk as its telephone system it does get a simple to use client free of charge KPhone The KPhone SIP telephone is similar GnomeMeeting being built on the Qt library It also provides video conferenc ing client facilities assuming that a pre installed VIC package 3 If you intend to use a distribution version of KPhone make sure that you have some experience of SIP telephony You will search in vain for the documentation and the KPhone Web site is a gaping hole This is a pity actually as the pro gram has everything you need to blast off into the world of VoIP During our test two versions came into play release 3 13 from the Suse Pro fessional 9 1 distribution and the current source code version 4 02 which is available for download from the web site To compile KPhone 4 02 you need the Qt3 libraries which should already be on your system assuming that KDE 3 is your standard desktop The compila tion of the sources ran as expected without problems although that didn t stop version 4 02 displaying half a screen full of configuration help2. Once started KPhone ran extremely quickly offering an easy to use graphical interface see Figure 5 When the program is first launched a configuration window opens to prompt the list of active lines For secu rity IAX2 implements the Status Remote gl Audio We user for critical data like the user and host por Unattached MDS __ challenge response method MDS for authentica DTMF Hold Transfer tion purposes Users can tion of the SIP URL In addition users are prompted to specify Hide require security for internal connections and assign e whether they have a proxy connection and passwords through the pro 4 GHI 2 ABC 5 JKL whether KPhone should gram settings Automatic gain control 7 PQRS TUV automatically register After completing these echo cancellation and noise reduction can all be enabled via checkboxes According to the help file the GC and echo cancellation should not be run together as this will degrade the output quality significantly www linux magazine com am Figure 5 KPhone makes life easy for newbies You can pro duce DTMF tones at a push of a button and use them to query an answering machine steps you can fine tune the details in the Prefer ences menu The KPhone user is typically busy with two windows the main window shows the con nection stat
3. makes KPhone stand out against most other softphones SIPset SIPset is a simple but barely usable client The softphone supports just one codec PCMU and is thus very restricted However if the user agent is used on both sides in peer to peer mode it is entirely sufficient It is too bad that SIPset does not have a bigger choice of codecs as most other softphones fail to provide a similar range of functions and options The VoIP client can be launched either in graphic mode see Figure 6 or in text mode The pro gram supports both OSS and ALSA audio plug ins Even support for IPv6 is inte erated In order to place calls in a VoIP provider s network you only need to register your proxy server s IP address along with the appropriate password The provider proxy registration can be individually re configured if it expires SIPset is also suitable for video tele phony To support this the program needs the MPEG4IP package which is available at 5 Among other things the Open Source kit contains MPEG 4 and H 261 codecs for video applications as well as AAC and MP3 codecs for compression decompression of audio streams Table 1 Softphones Feature List Version 0 8 9d 0 3 0 Web www iptel org products www cornfed com products index html bonephone License GPL Cornfed Prices free free Functions VoIP Protocol SIP SIP Audio Codecs PCMU L16 G 711 uLaw STUN no SOCKS no ILS Directory no ALSA O
4. own con tact database Cornfed is de signed for users who do not want to waste time on the configu ration but do have experience with the command line tools Very positive is the support both for Personal Data General Settings Directory Settings Call Options NAT Settings Sound Events v H 323 Settings Advanced Settings Call Forwarding Gatekeeper Settings Gateway Proxy Settings v Codecs Audio Codecs Video Codecs v Devices Audio Devices Video Devices Audio Codecs Available Audio Codecs Audio Codecs Settings Automatically adjust jitter buffer between 20 A Name info Bandwidth X MS GSM Good 13 Kbps X SpeexNarrow L5k Excellent 15 Kbps T up X ILBC 15k2 Excellent 15 2 Kbps iX GSM 06 10 Good 16 5 Kbps Down X SpeexNarrow 8k Good 8 Kbps Information iX G 726 32k Excellent 32 Kbps X G 711l uLaw 64k Excellent 64 Kbps X G 711 ALaw 64k Excellent 64 Kbps Cj G 723 1 A POr and 500 ms O Enable silence detection ALSA and for OSS compatible sound cards On the down side there is only www linux magazine com X Close Figure 3 Spoilt for choice GnomeMeeting has audio codecs to suit any kind of Internet connections no matter what bandwidth you have August 2004 shows the used bandwidth see Figure 3 GnomeMeeting uses plug ins to for audio stream management and can han dle both AL
5. 34 119 Talk Figure 6 SIPset s GUI version does not overload the user enter an address click on the Talk but ton call deployed behind a NAT gateway without having to jump through hoops With multiple network interfaces the user selects the appropriate one from a drop down menu The registration with a proxy service can occur automatically if desired The choice of codecs is not extensive However it still provides a suitable encoder decoder for each environment Thus codecs can be selected as needed in the range between 8 KBit s G 729 and 64 KBit s PCMU Included in the group is the Speex codec 6 which encodes speech with variable bit rate as well as iLBC TkPhone is easy to use and leaves nothing to be desired This said the question arises as to why one should look to a commercial client on Linux as there are enough free clients Perhaps an account with VoIP gateway provider iConnecthere will convince some users as they give a special rate to the TkPhone manufacturer Otherwise one can completely wipe the program from the disk with the help of the provided uninstallation script So what should I use None of the tested programs proved to be completely up to day to day use The connection was always successful crashes occurred only sporadically with SIPset GnomeMeeting is recommended for readers who only need to make peer to peer calls The program is very mature and offers extensive documen
6. Asterisk users iaxComm is a mature program that does not leave any thing to be desired Thanks to the wxWidgets toolkit it runs on any stan dard platform However because it is restricted to the IAX2 protocol it is cur rently not usable with commercial providers The really satisfying thing about this test lies in the realization that VoIP softphones are available for Linux for all protocols requirements and tastes free of charge and with Open Source code If VoIP does become a success it might have a lot to do with the Commu nity s willingness to become involved in development work E 1 GnomeMeeting Workshop Kilian Krause Christian Strauf Telepnonitis Linux Magazine Issue 40 March 2004 Page 54 2 wxWidgets http www wxwidgets org 3 VIC http www nrg ee bl gov vic 4 STUN TURN http www newport networks com whitepapers fwnatwpes3 html 5 MPEG4IP http www mpeg4ip net 6 Speex http www speex org 7 Microtelco http www linuxjack com August 2004
7. SA and OSS compatible sound cards Calling from a local network that uses NAT is just as easy as with the Cornfed client GnomeMeeting also has an exter nal service that determines the public IP addresses of the routers and registers them automatically in the configuration In addition the client must be registered in the ILS Internet Locator Service database of seconix gt jiaxComm laxComm The custom made iaxComm softphone is well suited to the Asterisk PBX server s proprietary IAX2 protocol The program lacks a choice of protocols but it com pensates thanks to good functionality and extensive platform independence IaxComm works equally well on Linux MacOS X or Win32 as guaranteed by the wxWidgets Framework 2 The installation on smaller machines can take a while as the wxWidgets sources need to be recompiled as static libraries and with XRC XML based resource sys tem support On our lab machine the install took about an hour Later the software run without any com and the IP neers Address Translation MIC option must be ee enabled You need Siis ie to configure appro priate port for Extension Account warding rules at the firewall to support this Check the FAQ on the GnomeMeet ing Web site for more details If you use a gate keeper service you can also use Gnome Meeting to call organized There are two rows of normal phones buttons for speed dialing bottom However thi
8. SS Video Plugins no Audio Plugins IPv6 Support yes Bandwidth Profile 33 6 Kbit s bis 1 Mbit s DTMF Tones yes User Functions GUI CLI Addressbook Dialpad Multiple simultaneous calls Automatic call Call holding Video conferencing August 2004 avl Amm i aX omm NI HVilt 1 0 2 www gnomemeeting org GPL free H 323 iLBC GSM 06 10 MS GSM G 711 Alaw G 711 uLaw G 726 G 723 1 no no yes ALSA Quicknet Video4Linux Webcam AVC Firewire Cameras yes no yes www linux magazine com 20040228 http iaxclient sourceforge net iaxcomm GPL free IAX2 G 732 1 G 726 GSM G 711uLaw G 7nAlaw iLBC LPC 10 Speex U a no no no 4 02 www wirlab net kphone index html GPL free SIP G 711uLaw GSM iLBC yes no yes OSS ALSA VIC 1 5 0 http vovida org applications downloads sipset Vovida free SIP no no no OSS ALSA MPEGaIP 1 0 5 www thekompany com products tkphone 10US SIP PCMU GSM G 729 Speex yes yes yes OSS ALSA no We had no trouble building the appli cation from the source in our labs A HTML formatted manpage provided con figuration support although the page is only available online on the SIPset web site The only known bug to appear was an error in the call to URL function A workaround is available and is likewise to be found on the Sipset website How ever the program crashe
9. Softphones for SIP H 323 and IAX2 Give me a virtual world of Internet telephony you don t need to start out with extensive hardware A soft I order to enter into the phone is enough A small program allows Voice over IP VoIP conversations and is happy with a sound card micro phone and external speakers The only thing to worry about is the supported protocols The two most important ones are H 323 and SIP which are also used by VoIP providers who route their customers to conven tional or mobile telephone networks Up to now no softphone exists that supports H 323 as well as SIP However there are programs that don not know what to do with either although they are designed for VoIP tele phony For example the PBX software Asterisk uses two of its own protocols IAX and IAX2 Linux magazine took a look at seven different VoIP user agents and put each to the test Soft ware for H 323 SIP and IAX2 was considered An overview of all soft phones including download links is provided in Table 1 We used two slightly ancient PCs for our lab environment Celeron 433 Pen tium II 800 As is the case in many companies machines of this type are still faithfully fulfilling their duty as desktop workstations The machines had a SoundBlaster 32 PnP and AC97 com patible on board chip as sound devices We used a cheap microphone as an input device and ran the output through a standard PC headset This configu
10. d now and then while accepting input into the options window There were no problems with an existing VoIP connection however Because of the concentration on the PCMU codec we would hesitate to rec ommend Sipset although the program behaved almost perfectly in our tests Since it already supports video calls and can handle IPv6 speedy development of this softphone would be welcome TkPhone The only commercial representative in the test was the TkPhone softphone by TheKompany At 10 the program isn t exactly expensive and it does create a good impression A free demo version is available for download although it restricts calls to two minutes Before running the install sh script to start the install the user must decide whether or not to use system font anti aliasing Depending on your system you first need to install one of the three sup port libraries noxft xftl or xft2 If the TkPhone installation script does not find any of these libraries it stops the setup The Readme is available as a decision making aid as to which of the three packages should be installed Visually TkPhone can easily hold sway with GnomeMeeting or KPhone see Figure 7 and can be set up quickly using the configuration menus The soft phone contains support for STUN and the SOCKS service which was unique in the test group Thus the program can be _ gt SIPSet by VOVIDA ORG File Settings View Tools joe sip marco 212 211
11. d to modify etc hosts This is im portant since Java ap ased0joYyd MMM 0 0Ud jeuro a TT re Ptererareerrteretentesetene joe lmag softphones Cornfed cornfedsipua gt sip Jun 10 08 41 25 Cornfed SIP User Agent Jun 10 08 41 25 Version 0 2 6 Jun 10 08 41 25 Copyright C 2004 Cornfed Systems Jun 10 08 41 25 Written by Frank W Miller Jun 10 08 41 25 full duplex soundcard operation Jun 10 08 41 25 local URI sip 100 192 168 1 103 Jun 10 08 41 25 userinfo 100 Jun 10 08 41 25 user 100 Jun 10 08 41 25 hostport 192 168 1 103 Jun 10 08 41 25 host 192 168 1 103 Jun 10 08 41 25 port 5060 Jun 10 08 41 25 remote URI sip 613 fwd pulver com Jun 10 08 41 25 userinfo 613 Jun 10 08 41 25 user 613 Jun 10 08 41 25 hostport fwd pulver com Jun 10 08 41 45 port 5060 Jun 10 08 41 45 registrar URI sip 276140 fwd pulver com Jun 10 08 41 45 userinfo 276140 Jun 10 08 41 45 user 276140 Jun 10 08 41 45 hostport fwd pulver com Jun 10 08 41 45 domain fwd pulver com Jun 10 08 41 45 host 192 246 69 223 Jun 10 08 41 45 port 5060 Jun 10 08 41 45 ethO IP address 192 168 10 6 Jun 10 08 41 45 ethO subnet mask 255 255 255 0 Jun 10 08 41 45 www myipaddress com 12 148 220 166 Jun 10 08 41 46 visible IP address 217 232 160 137 sip gt Figure 2 By means of the detour to myipaddress com next to last line Cornfed determines the official IP address of the the NAT gateway the visible add
12. es call download needs to be put into the correct place in the user s home directory There are two README files to help with the installation and configuration These are available in English but stick to the basics and lead the user step by step through the necessary tasks Four files need to be edited for the configuration all of which are in the user s home directory At no time are root privileges required The user first needs to set the paths to the JDK in the file JAVAHOME Then the main settings can be defined in bonephone bonerc Here is where you configure the loca tion of the rat 4 2 20 media engine which takes care of Bonephone audio processing In addition bonerc includes per sonal data like the SIP identity user name in SIP URLs and the IP address of the local computer All the other settings can remain untouched for the time being Next we continue in the main direc tory bonephone with the file configfile In this file the Fully Qualified Domain Name FQDN is initially set The FQDN is composed of the hostname and domain name for example pcl domain local The local IP address likewise needs to be set All other set tings should remain untouched For most users the configuration is now complete and the IP telephone can be started from the main directory with the run command If you have two network cards with a public and a private IP address or use IPv6 you will nee
13. ly address with which Cornfed connects itself unasked The SIP Phone contains a Tro jan horse which independently sends information to the Cornfed developers The developers nevertheless refer to the existence of the data collector in the User Guide and the FAQs on their web site According to these the user agent trans mits only warning and error messages to a Cornfed web server However this function cannot be deactivated The first time we launched the GUI on our lab system it failed as Cornfed was missing the libpopt so 0 library The newer version libpopt so 1 was already in usr lib and thus a softlink fixed the problem with ln s The GUI showed up but now the user was missing a few things Since a GUI is only available with the current version 0 3 0 it does not yet contain all the options that the com mand line client has However the first test with the trimmed down GUI ran pos itively In general the softphone is stingy on configuration options You can disable NAT or select the desired network inter face but not much more Currently the only audio codec Cornfed supports is G 711 muLaw On the upside the user can produce DTMF tones which can be used for example to query an answer ing machine The documentation which mainly deals with the details of the command line interface is very good The GUI is easy to use as it is extremely concise lacking only and Softphones COVER STORY one audio codec Con
14. ration is typical of many office and home PCs Bonephone Bonephone the IP telephone is still at the alpha stage although this is not immediately apparent when running the application The SIP client s GUI front August 2004 Softphones are the inexpensive way to make a phone call using the Internet The hardware is already sitting in each PC and besides that virtual telephones are simple to configure BY JORG REITTER end is Java based It is kept simple and lets the user make only a few configura tion changes One outstanding thing is the ability is to specify a certain band width for up or downstream in order to optimally configure the VoIP communi cation for your existing Internet connection For this Bonephone sup ports data rates of 33 6 KBit s for modem connections up to a comfortable 1 MBit s see Figure 1 The Bonephone install is quite fast However you need to be aware that the application assumes version 1 4 or later of the Java develop ment kit JDK In addition the configura tion package which is available as a separate www linux magazine com BonePhone 0 8 9d IPv4 ing ER ETE O lt 33 6 Kbps rr Se T al ri be O lt 256 Kbps O lt 768 Kbps O lt 1 Mbps O gt 1 Mbps i user pref Siu 36 90 dTa Device OSS CTL1745 Device No Audio Device Figure 1 Bonephone allows users to individually configure the Internet connection bandwidth COVER STORY Softphon
15. ress might otherwise use the wrong IP address for VoIP communication In addition Suse users should change the localhost entry in the hosts file according to the instructions in the README config file Otherwise the SDP transmits the local host address as the official IP address which makes communication impossible Bonephone is a good place to get started as it is easily set up just by fol lowing the the instructions The individual bandwidth specification fea ture additionally opens up SIP telephony to modem owners Contacts are stored in a simple telephone book using URL nota tion Bonephone logs all calls in a text file which by default is in the working directory of the SIP user agent Bone phone is provided for download under the GPL as binary or source packages Cornfed Cornfed comes as both a GUI version and a command line client which only supports SIP A user manual in pdf for mat shows the configuration as well as how to call using the console Cornfed is available only as a binary package which in addition to the docu mentation and the license contains only the executable This can be launched directly from the directory in which it is unpacked One outstanding function of the Cornfed user agent is the automatic determination of the official IP address in case the user is behind a NAT gate way For this the client contacts the website myipaddress com see Figure 2 However this is not the on
16. s as _ right sumes special hard ware like a Quicknet or a Creative Voice Blaster card The Gatekeeper serves administrators as a security point on the border between LAN and Internet where users are required to supply their name and password to authenticate It is easy to call other VoIP telephone users A local address book allows you to sort contacts to be sorted by group and provides speed dialing Users need to register with the ILS database to find other GnomeMeeting or MS NetMeeting accounts They can also be accessed through the address book GnomeMeeting is recommended as a mature application that does not leave anything to be desired in terms of H 323 protocol based VoIP telephony GnomeMeeting lends itself to intuitive use and the option to implement a videophone makes the program extremely attractive When work on implementing SIP is completed GnomeMeeting might well move up to take the top notch in VoIP softphones August 2004 LJ C P A s Tom U heh U7 LE WLS J Marco COCU M ee Figure 4 The laxComm GUI is well further ado provided we used the binary version IaxComm offers all of the facilities one expects from a graphical VoIP phone see Figure 4 The GUI is full featured with buttons for dialing hold ing calls and speed dialing You can use the dial pad to produce DTMG tones allowing you to con nect to a voice mailbox The address book is accessed from the options menu how ever
17. sidering the fact that Cornfed can only be had as a binary which happens to contain a Trojan we would hesitate to recommend the VoIP client GnomeMeeting GnomeMeeting comes with a reputation of being an Internet telephony classic that has dared to make the jump to version 1 0 in the meantime It was the only VoIP Client in the test field suit able for communication using H 323 SIP support is planned for release 2 0 GnomeMeeting has several outstanding characteristics that make it equally inter esting for beginners as well as professionals In addition to voice communication via VoIP the program allows you to hold video conferences and text chats A com prehensive workshop can be found in Linux User 1 Those who want to jump right in without tinkering around can launch the built in configuration wizard Installation is very simple since each of the major distributions includes GnomeMeeting If you would like to look at the current version 1 0 2 you can download source code or binary RPM or DEB from the website Conveniently gnomemeeting org also hosts the current source packages for the required libraries H 323 and Pwlib which need to be installed first This communication genius supports a substantial set of audio codecs that allow users to get the best quality out of their Internet connections The appropriate configuration window immediately v General address book Thus users will need to create their
18. tation as well as support through a mailing list and the FAQs Calling normal phone numbers is a little difficult with GnomeMeeting however since it cur rently supports only H 323 And only a few providers such as Microtelco 7 support H 323 connections Things look a little different with SIP which most VoIP providers support A SIP client will support both peer to peer connections and normal PST connec tions through a provider s VoIP gateway KPhone is a good SIP softphone to get started with Not because it represents the KDE counterpart to GnomeMeeting but simply because it is available in every distribution and provides the func tions one needs If you can t warm to KPhone take a look at Cornfed or SIPset Both come with a GUI and command www linux magazine com Softphones COVER STORY sip joe 192 168 10 6 Call Log Mixer DTMF B E Eee A E Eee Figure 7 TkPhone starts up a tidy interface and was the only softphone in the test which also support SOCKS Servers line interface which has a charm of its own The Bonephone Java telephone pro vides an interesting alternative to the conventional programs The softphone is slim fast and offers only the most neces sary features Let s not totally dismiss the commercial product TkPhone either Although you really should avoid spending the 10 unless you intend get yourself an account with iConnecthere the TheKompany partner For
19. us and lets COVER STORY Softphones the user configure whether he or she is on line or does not want to be disturbed at the moment among other things You can use the right mouse button to access actions such as Call or Send Message The call window has a dial pad available for placing calls or transferring calls to another URI There are numerous configuration options KPhone was the only GPL pro gram in our test that supported a STUN Simple Traversal of UDP through NAT server This allows you to run a SIP Client behind a NAT gateway On request the STUN server assigns the client ports on the public IP address for incoming and outgoing communication packets However this function is not recommended in connection with sym metrical NAT since this special kind of port mapping opens up a huge security hole More information on STUN and the safer successor TURN among other things can be found here 4 In the audio settings the user can choose between three codecs G 711u GSM iLBC or change the ring tone The SIP configuration defines whether KPhone should reject calls from unknown URIs Besides that it can reroute incoming calls to a specific URI which is handy for users who desk hop With the exception of the missing doc umentation KPhone makes a good impression Although the option to use STUN is insecure it is often the only pos sibility to access phones behind a NAT gateway And video conferencing sup port
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