Review of the Citel C-4110 IAX (and SIP phone)

by Ruben Email

Some time ago I did a posting about VoIP Supply announcing that they would begin to sell a IAX phone. At the time of release, they had a contest where one could win one of these phones. Usually I am not very into contests, but when someone asked me nicely if I was willing to write a "Why is IAX {cooler,better,more suiteable} than SIP", and submit this to their contest, I did oblige.

Given that VoIP Supply is one of the few companies around that I consider myself to be a fan of, I had no problem doing this. VoIP Supply are very good, and if they did ship their hardware with Euro-type power adapters I would probably be buying most of my stuff from them. Unfortunately they do not.

Now back to the Citel C-4110 VoIP Supply sent me back in May 2009.

Eco friendly, but very nice, packaging

We all like eco friendly packaging. You know when your eagerly awaited hardware comes in grey boxes. The box I received was a grey package - but to my surprise they had put on a nice drawing on the top of the box. Inside the box was the usual suspects: Phone, hand set and US power adapter. Even if the phone it self is on the cheap side, price wise, they had put in a Ethernet cable. A nice and welcome touch. No CD-rom, which is good for the environment. The only form for end user instruction was a small yellow paper which told me to go to a given address to download the phone's manual. The phone comes with 4 dedicated user assignable keys, and the Citel guys have included various ready made labels to put onto these buttons.

What did really annoy me is that there where no user name nor password written on the yellow paper. I had to download the user manual, before getting the unit configured.

Physical appearance

I am always amused when taking a look at unknown telephone brands. The reason for this is what I have coined the Far East Syndrome. Short explained, the Far East Syndrome is that even if a unit is extremely interesting at first sight, it may have a one (or more) shortcomings making the device very unusable. Once I got my hands of a quite capable phone - which had a button labeled redil and a hand set which could be easily broken in two.

The C-4110 did not amuse me in respect to the Far East Syndrome. The build quality is sturdy. When I try to twist the relatively heavy hand set, I can not do it. It does not budge. It does not squeak. I am sure I could kill an elephant with the hand set, if needed.

The keys on the telephone are pretty solid.

I am quite impressed with the build quality of the phone.

What does NOT IMPRESS ME is the colour of some of the keys. Why is is so that a lot the smaller brands in VoIP degrade them self, design wise, to mess up the appearance of their goods with colourful buttons.

Colourfull buttons and keys may make the unit look cheap. On the C-4110 we have grey keys, a Viagra coulured rls button, two pistachio ice cream coloured line buttons, one washed out orange Hold button and lastly a stop-red button with speaker/head set symbols.

On top of this we have the keys for attaching various paper labels.

One could get the impression that the phone has a messy key interface. Frankly I am quite happy with the key interface. A few dedicated keys on a desk phone makes the phone more usable, than having to wade through soft button menus.

As noted above we have buttons for Rls which in fact a release button. Other dedicated keys not mentioned above are Mute, Transfer, Menu, Callers and volume control. 3 soft keys and a up-down-left-right pad along with 4 dedicated user assigned keys makes up the rest of the keypad. The pre-defined ready-made labels from Citel are Pick-up, Message, Conf/Trans, Call-Fwd, DND, Ring Again, Transfer, 2 x IP: and a slew of blank labels. My question is: There is a dedicated Transfer button on the phone, why should I add my own?

The phone has provisions for a relatively big speaker area, but as with a lot of units from the Far East, the actual speaker is much smaller. With the C-4110, the speaker is ca 75% of the designated speaker area. I have not opened the phone to see if the reason for this is space constraints inside the unit.

Sound quality

The unit supports G.711{a,u}, G.723, G.729 and surprisingly G.722. At the next VoIP Users Conference I am going to try using the G.722 codec and dial directly into the bridge.

The hand set has excellent quality. The downloadable manual does not mention G.722 at all, nor if the hand set if Wideband Audio ready.

This is not a true speaker phone. I have not done extensive testing on this feature, but the preliminary tests shows that the built in speaker phone microphone is not very good. I had to sit very close to the phone for the mic to pick up my voice.

Web configuration interface

I was pleasingly surprised when I fired up the web GUI. It is very clear and really to-the-point. It does not scream of Far East Syndrome at all.

I will not go into great details on the various configuration options, but just high-light a few important features.

Network configuration

The phone has a built in micro switch which one can plug in a computer, or other network devices. This switch can either be set in bridged mode - or let the phone assign an address. I have not tested, but according to the manual, the phone can also be set up to be a DHCP server and a router (NAT enabled).

The phone has support for VLAN and QOS settings. Given the price range this phone falls into, I was pleasantly surprised on the level of details regarding QOS parameters.

VoIP configuration

It is possible to configure one IAX account and two SIP account. Both the SIP and the IAX configurations are pretty straight forward, except for the input parameter Phone number.

Of course, my own provider (which is my self), does not use phone numbers as SIP accounts. For inexperienced VoIP people this will become a problem. A small glitch in the firmware I am currently using (i.e. not the latest) is also annoying: The SIP registrar information is not shown in an existing config. This can of course be quite confusing if this is your first SIP phone.

A very nice feature is what Citel has named as Dial Peer. In effect you enter any phone number you wish, and then the SIP or IAX address for that phone number. Let's say I want to enter *11 when dialing the VoIP Users Conference. I can then configure the Citel to actually dial 200901@login.zipdx.com. This feature is very versatile and I have not seen this in any of the main stream phones available on the market.

Phone configurations

Up to 4 codecs can be prioritized, and a few (most) interesting DSP parameters can be set (i.e. payload length).

A very nice settings is the possibility to disable the #-feature. It is usable, and have become a "de-facto" standard on most hand sets today: Press the #-key at the end of a phone number to initiate the call. I really detest this "standard". The #-feature is just due to lazy programmers.

What a lot of other providers mis-name a "dial plan", Citel call this "Digital map & rule" configuration.

On the negative side: I have not found any documentation stating how to upload a phone book, or have the phone subscribe to a distributed phone book.

Phone Maintenance

The only thing worth mentioning here is the ability to export (and import) the phone's configuration parameters. Interesting observation is that as far as I have investigated, there are several parameters in the config file, which are not found in the GUI. This should thus mean that one can change "hidden" parameters.

For new readers of this blog: One of the first things I check out when dealing with a new VoIP unit is the possibility of remote mass provisioning. The C-4110 has in fact auto provisioning over FTP, TFTP and HTTP. The Citel guys has also added the ability to encrypt the configuration file. No HTTPS, but given the encryption ability, this might not be needed unless you need to verify the client talking to the provisioning server. I will be the first to admit that the auto provisioning is not as elaborate as the one found in the Linksys/Cisco SPA devices - but for most uses it does suffice in plenty.

Security settings

Given the phone's micro switch and the possibility to make the phone behave as a router - it does of course(!?) have a built in firewall. Not very advanced, but good enough for the basics.

A small surprise in the security settings was the VPN settings. The manual is very sparse, but taking a look at the parameters there it at least support for L2TP. There is also something called a UDP Tunnel.

A big surprise: I found a shell prompt

A very welcome feature not found in many other phones is the ability to get shell-prompt on the phone. It's not SSH, but Telnet (unfortunately). However, if the phone is deployed behind a firewall with well behaved users, telnet should not cause a problem.

Why I am getting exited to find a shell prompt? The phone becomes so much easier to remotely manage. Everyone who has tried to automatically script phone GUIs from the command line knows how painful this is. Scripting a telnet session is a walk in the park compared to scripting a web GUI.

Final remarks

Does this phone compare to the SPA offerings from Linksys/Cisco, or the various phones produced by Snom? The short answer is absolutely a big YES.

It is so good that the Linksys SPA 942 I am using in my home office will be replaced by the Citel as the main phone and the SPA will become the lab-phone.

The phone is good enough for the office environment, but not something you would like to put into the living room.

I would also have no problem deploying this phone for a hosted PBX (Asterisk) system - in fact I would probably deploy this phone over a pure SIP phone due to it's support for IAX.

3 comments

Comment from: Gary V. [Visitor]
I have some remote polycom sip phones and was wondering how a remote sip phone would compare, bandwidth wise, to this iax phone? I would guess the iaxy would save on bandwidth over the sip? I would also guess that this iax phone would be easier to deploy as a remote extension and would not have to open all the ports like sip?
21/09/09 @ 19:00
Comment from: Ruben [Member] Email
Hi Gary,

Bandwidth wise, IAX2 is a leaner protocol regarding overhead. However, the savings for only two channels (i.e. a two-way conversation) is negligible compared to SIP. We are talking about a few bytes.

However, if you do trunking with more than a few channels, using IAX in Trunking mode will save on bandwidth.

Example using the Wideband Audio codec G.722 in 64 Kilobit per second (Kbps) mode, 2 channels (i.e. 1 simultaneous call) we get the following numbers:

IAX2 Regular will consume 153 Kbps.
IAX2 Trunked will consume 157.68 Kbps.
SIP will consume 159.26 Kbps.

If we up this to 10 simultaneous calls, IAX2 Regular will consume 1530 Kbps, IAX2 Trunked 1309.68 Kbps, and SIP 1592.5 Kbps.

We see that using IAX2 Trunked gives us a saving of 0.28 Megabit per second. Put it another way, IAX2 Trukned will allow for 2 extra Wideband Audio calls.

To calculate your own bandwidth usage, I urge you to use one of the many bandwidth calculators out there. One such calculator is http://www.asteriskguru.com/tools/bandwidth_calculator.php , another is http://www.bandcalc.com/

As for your 2nd question if using a IAX2 phone would have to open less ports than SIP this is also true (unless you do some tunneling magic). If you are using SIP you need to open "a lot of ports for RTP", and just mentioning open "a lot of ports" on the firewall to your network administrator will probably get him, or her, to loose a good nights sleep.

The problem with IAX2 phones so far has been the quality, but with the Citel C-4110 a lot of quality issues are solved.
22/09/09 @ 01:06
Comment from: Alex [Visitor]
Cheapest lowest cost IP Phones with IAX IAX2 + SIP Support ,Thats amazing.Makes me feel to buy Citel C4110 Phones.
26/09/09 @ 18:37

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