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Patton Electronics SmartLink 4050/2 SIP VoIP phone
[January 23, 2006]

Patton Electronics SmartLink 4050/2 SIP VoIP phone

Patton’s SmartLink 4050/2 SIP VoIP Telephone review

Patton Electronics Co
7622 Rickenbacker Drive
Gaithersburg, MD 20879 USA
Phone: 301/975-1000
Fax: 301/869-9293

Web site:

RATINGS              (0–5)
Overall Rating: A-

SIP (session initiation protocol) based IP phones have ushered in low cost telephony. Full-function SIP phone offerings from various vendors have made connectivity to a VoIP service provider using a standard VoIP protocol easy and affordable. Thanks to technologies like SIP, IP phone connectivity is becoming commoditized and sooner or later we will end up owning one. Viewpoints may differ on how to deploy IP phones with some advocates for SIP, others for MGCP and still others for H.323. For me the underlying motivation to look at a SIP IP phone was seeing a slew of SIP user agents introduced in the market on a regular basis and to top it all the fact that currently VoIP is really big business.

In the VoIP SIP IP phone space one such offering in the market is from Patton Electronics. They offer a VoIP gateway, VoIP router etc. and to complement these VoIP products they have SIP telephones called the SmartLink 4050 series. In this series Patton offers you the SmartLink SL4050/10 (Multi-line support-up to 10 concurrent Calls) and the SmartLink SL4050/2 (Multi-line support-up to 2 concurrent Calls). Speaking of the SmartLink SL4050/2-for a network interface it boasts RJ45 x 2, 10/100Base-T ports and in tandem with a SIP server promises support for features such as call transfer (unattended, blind & announced), call forward (busy, no answer, unconditional),anonymous call blocking, out-of-band DTMF,message waiting indicator, call park/pickup etc. For a voice codec you get flavours such as G.711μ-law, G711a-law, G.723.1, G.729a etc. In phone functions you get features such as multi-user (up to 4 SIP accounts),full duplex speakerphone, handset and speakerphone volume adjustment, speed dial (10 records), phone book (200 records), a redial key which provides information for redial, missed and received calls, message key for checking voice messages, conference key for a 3-way conference, with the hold key you can actually place the person on the other line on hold, the speaker key on the phone allows a handset free operation, the alpha-numeric keypad allows inputting IP, phone number and alphabet characters. Mute/Func. Key can be used to disable user’s handset microphone so that the person on the other line cannot hear anything, the Menu key gives access to the phone menu and the Phone Book key provides access to the phone book. The unit as a standard comes with a 2x16 characters LCD display which displays menu, time, clock, name, phone number, call status etc. The phone users can connect to the required SIP Server by setting a registrar server and an outbound proxy server.


As to dialling you could attempt either a direct IP call without SIP registration or dial a registered number via SIP server or dial a url from the phone book or even use the speed dial facility. When it comes to voice quality there is inherent support for VAD (Voice Activity Detection), CNG (comfort noise generation) and AEC (acoustic echo cancellation), G.168 and jitter buffer. NAT transversal is made possible using STUN & UPnP and IP assignment thanks to Static IP, DHCP and PPPoE options. Configuration of the phones can be done using the keypad and LCD display via the menu key and or using your PC’s web browser. The TFTP server facility allows transfer of files from a computer to the phone and FTP client facility allows the IP phone to download files from the FTP server and update the firmware automatically.


Before the curtain rings down on the discussion let us test and see how the unit actually performs in the real world. With this is in mind I invited Patton to submit their Smart SL4050/2 for a spin in the labs and as an extension of the tests decided to further carry out a 3rd party SIP protocol VoIP gateway inter-operability lab test.


Operational Testing

In my test bed the SL4050 2 line VoIP SIP Phone was paired with Multi-Tech’s SIP VoIP gateway and a NEC PABX. I connected the phone’s Ethernet port labelled as LAN to the Ethernet switch and the port labelled as PC to the test PC. Switched the phone on and with the menu key enabled DHCP. The quick start guide was quite handy for this. Next, I used the web interface to manage and configure the phone. The phone uses user name and password to check my credentials before letting me in. The main web interface window is well laid out with all functions arranged neatly under headings like-management, network settings, SIP settings, SIP account settings, STUN & UPnP Settings, Voice settings, phone settings, call tracing log, phone book, and speed dial. The phone book allowed 200 entries. The phone settings is where I could select the tone setting, ringer type and either enable or disable features like do not disturb, call waiting, anonymous call reject or set attributes for anonymous call and call forward. Voice settings are where I could prioritise the selection of codecs to be used for a call. The most used codec to the least used codec in terms of priority with 4 priorities and the choice was between G.711μ-law, G711a-law, G.723.1, and G.729A.Further if required you could set the real-time transfer protocol (RTP) packet length, turn VAD on or off, set the required DTMF Method for the IP phone, under QoS set the voice TOS value and then enable or disable VLAN and then its required priority and ID. The SIP account settings allowed multi-user setting (up to 4 SIP accounts) —that is, the SIP phone can receive calls from up to four different phone numbers. In SIP settings it has fields where I could enter values for the SIP phone port number, registrar server, outbound proxy server, message server, park server etc.The network settings is where you get to input your DNS server value and select between DHCP, PPPoE or Static IP operation.

I could dial a SIP number or dial an IP address or even go to the phonebook and scroll up or down as needed and then dial the required number.


In a mixed multi-vendor environment after establishing the correct setup between the SL4050, the 3rd party SIP VoIP gateway and the NEC PABX inbound and outbound phone calls were made successfully. The remarkable thing about the SL4050 is that it is intuitive to set up and use. The SL4050 is best suited for a SIP server environment (that is what a SIP phone is supposed to do) nevertheless with some tweaking we were able to do without a SIP server and make do only with a VoIP gateway and a run of the mill PABX.

The SL4050 scored well on ease of installation, phone features, product documentation and integrated help. The buttons on the phone are clearly labelled as Menu, Phone book, Mute/Func.,speaker phone, conference, message, transfer, hold, redial etc. including the buttons for menu access and arrow keys for navigating the menu indicating Patton’s clear focus on ergonomics and design.

Room for Improvement

It would be good if there was a way to verify and change Ethernet port settings on the phone (could prove useful if ever there is a situation of mismatch between the Ethernet ports of the phone and the connecting PC or switch). I would also like to see call quality diagnostic information on the phone and suggest that Power over Ethernet (802.3af) which for now is offered as an optional accessory be offered as a standard part of the phone offering.

Nevertheless my recommendation for the 4050 stands good for those looking towards a simple, well featured and affordable SIP phone.


The SmartLink 4050 from Patton’s stable strives to strike a balance between simplicity, price and features and overall comes out as a simple but effective budget priced communication tool.


Biju Oommen is a Telecommunications & Networking Solutions Consultant with a special focus on enterprise products and solutions

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