UTStarcom GF210 WiFi VOIP phone

Discussion in 'UK VOIP' started by Mark Ingle, Jan 13, 2009.

  1. Mark Ingle

    Mark Ingle Guest

    This is rather a long posting but someone on this ng was asking about
    the UTStarcom GF210 WiFi VOIP phone. I bought one from sipgate.co.uk,
    and although I have not had it for long, thought I'd provide some
    feedback.

    I thought the price was very reasonable, and the delivery from sipgate
    very quick

    The first issue I had was connecting to my WiFi network. Although the
    phone supports WPA and WPA2 encryption, I just could not get the phone
    to connect to my WPA network; it would see the network but wouldn't
    connect. As I have an older access point with WEP encryption, I used
    that instead, and it connected fine. It is worth pointing out that the
    phone doesn't seem that intuitive when scanning for and configuring
    encrypted access points; it seems to be geared towards use on open
    access points.

    The next hurdle was connecting to a SIP account. I have used Draytel for
    some years now and have found them pretty good, reliable and supports
    the IPKall US number, so opened another account for the UTStarcom phone.

    This worked fine at home, and I tested the phone with the SIP number and
    geographic number for both incoming and outgoing calls, and it worked
    fine. However, when I took it to work and tried to connect through the
    access point, although the screen said 'registered' and I could make
    outgoing calls, incoming calls were diverted to voicemail.

    I then tried a Sipgate account; with the settings as instructed on their
    website, it would not work at all from home unless I switched off the
    NAT/STUN setting. It would only work on the work WiFi network with the
    NAT/STUN setting on. It is slightly inconvenient for a phone that is
    designed to be taken about and used on different WiFi networks to have
    to keep on switching the NAT on or off, but if that's what needs to be
    done...

    As Draytel doesn't have a STUN server, I suppose this means the
    UTStarcom phone cannot be used reliably on many WiFi networks, which is
    a shame especially as I am using two other Draytel accounts for family
    members in another country.

    Overall, despite the various issues with it, the UTStarcom WiFi VOIP
    phone seems exceptional value for money considering the flexibility it
    gives; hopefully in the next revision it will be more intiuitive to use.
    Mark Ingle, Jan 13, 2009
    #1
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  2. Mark Ingle

    Guest

    On Tue, 13 Jan 2009 23:31:44 +0000,
    (Mark Ingle) wrote:

    >Overall, despite the various issues with it, the UTStarcom WiFi VOIP
    >phone seems exceptional value for money considering the flexibility it
    >gives; hopefully in the next revision it will be more intiuitive to use.

    Thanks for posting this I have two F1000 which have been no trouble at
    all but when I first saw the UTStarcom WiFi VOIP I immediately got a
    gut feeling to stay well clear .
    Pleased I did after reading your report above :))) .
    , Jan 14, 2009
    #2
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  3. Mark Ingle

    Graham. Guest

    > As Draytel doesn't have a STUN server, I suppose this means the
    > UTStarcom phone cannot be used reliably on many WiFi networks, which is
    > a shame especially as I am using two other Draytel accounts for family
    > members in another country.


    I rather thought that any STUN server could be used, not necessarily
    the one provided by your provider.

    --
    Graham.

    %Profound_observation%
    Graham., Jan 14, 2009
    #3
  4. Mark Ingle

    Mark Ingle Guest

    Graham. <> wrote:
    >
    > I rather thought that any STUN server could be used, not necessarily
    > the one provided by your provider.


    That's interesting!

    However, by testing further I found out with sipgate if I switch off the
    NAT/STUN setting, it will work on both home and work WiFi networks. Will
    have to do more tests on other WiFi networks, but it looks promising.

    I have had other issues in the past with the Draytel accounts and
    incoming calls, so this may be the opportunity to move stuff to sipgate.
    With the Draytel accounts and certain configurations, incoming calls to
    the PSTN number get diverted to voicemail immediately, yet the IPKall
    number and VOIP number work. Very strange esp as when I configure the
    same account on my FritzBox, everything works fine.
    Mark Ingle, Jan 14, 2009
    #4
  5. In article <1itk0gm.pu23dn1ekc4w0N%>,
    Mark Ingle <> wrote:
    >Graham. <> wrote:
    >>
    >> I rather thought that any STUN server could be used, not necessarily
    >> the one provided by your provider.

    >
    >That's interesting!


    Yes, you can use any STUN server you like. It's purpose is to give the
    phone "hints" about how the NAT gateway it's sitting behind is working
    and what the it's external IP address is.

    This is because SIP encodes it's IP address in the data stream.

    If the phone doesn't put the extnernal IP address in it's data stream,
    but puts the internal IP address (eg. 192.168.x.y) then return audio
    data won't reach it.

    >However, by testing further I found out with sipgate if I switch off the
    >NAT/STUN setting, it will work on both home and work WiFi networks. Will
    >have to do more tests on other WiFi networks, but it looks promising.


    Some ITSPs have fancy proxy servers that can analyse the incoming
    SIP commands and notice: "Ah-a, it's coming from IP address 1.2.3.4,
    but the data has w.x.y.z in it, I'll re-write the data to have 1.2.3.4"
    .... and then pass the data onwards... I think sipgate moved to this sort
    of scheme some time back, but using a STUN server shouldn't make much
    of a difference.

    Some routers have a built in SIP-ALG (Application Layer Gateway) which
    tries to do the same on outbound SIP connections - they look inside the
    data going out and do the IP address re-writing for you. I've not found
    many of these that actually work.

    >I have had other issues in the past with the Draytel accounts and
    >incoming calls, so this may be the opportunity to move stuff to sipgate.
    >With the Draytel accounts and certain configurations, incoming calls to
    >the PSTN number get diverted to voicemail immediately, yet the IPKall
    >number and VOIP number work. Very strange esp as when I configure the
    >same account on my FritzBox, everything works fine.


    Probably because the FritzBox is the router? Therefore it's not actually
    doing NAT on the SIP data as it's on the "outside" of the router.

    Gordon
    Gordon Henderson, Jan 14, 2009
    #5
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