Siemens Gigaset.Net connection

Discussion in 'UK VOIP' started by Nick, May 18, 2009.

  1. Nick

    Nick Guest

    With their Gigaset range of ip phones Siemens provide a service called
    Gigaset.Net which allows registered phones to initiate VOIP calls to
    each other.

    I'm having a few problems getting this to work. It seems that other
    users can dial my Gigaset.Net number and connect to me but I can't
    connect to them, in fact they can't connect to themselves.

    I have checked all phones are registered and they are. So I'm now
    leaning towards the view that the service is pants, does anyone else
    have any experience.

    The G.722 codec that can be used with this service does appear to sound
    better than a normal voip call.
     
    Nick, May 18, 2009
    #1
    1. Advertising

  2. Nick

    Tim Guest

    Nick wrote:
    > I have checked all phones are registered and they are. So I'm now
    > leaning towards the view that the service is pants, does anyone else
    > have any experience.


    It is a bit pants. Works great if you are a public IP address.

    No so good through NAT - has a tendency for 1 way audio.

    I'm not fussed by this because I have plenty of other SIP servers to use
    which work a lot better.


    > The G.722 codec that can be used with this service does appear to sound
    > better than a normal voip call.


    Indeed. But you can use G.722 with almost any SIP service which doesn't
    use asterisk or otherwise limit codecs. Asterisk can support G.722 but
    needs patching and a few things aren't quite right. I'm told this will
    be addressed in future versions.

    I have an SER SIP server I use for internal calls. G.722 is nice but it
    is disconcerting because you can hear a lot more of the environment at
    the other end.


    Tim
     
    Tim, May 19, 2009
    #2
    1. Advertising

  3. In article <4a11f88f$0$515$>,
    Tim <> wrote:

    >> The G.722 codec that can be used with this service does appear to sound
    >> better than a normal voip call.

    >
    >Indeed. But you can use G.722 with almost any SIP service which doesn't
    >use asterisk or otherwise limit codecs. Asterisk can support G.722 but
    >needs patching and a few things aren't quite right. I'm told this will
    >be addressed in future versions.


    I'm using G722 in Asterisk. (the back-port to 1.4) Works OK, but as you
    say right now there are issues. The ones I've found are mostly to do with
    transcoding and issues after putting a call on hold (and IAX - sticking
    to SIP seems fine), but I need to do more tests. Have one beta-test
    customer using it with polycom phones and they seem OK with it so-far.

    >I have an SER SIP server I use for internal calls. G.722 is nice but it
    >is disconcerting because you can hear a lot more of the environment at
    >the other end.


    It's like video phones - must remember to not pick nose :)

    Gordon
     
    Gordon Henderson, May 19, 2009
    #3
  4. Nick

    Nick Guest

    www.GymRatZ.co.uk wrote:
    > Nick wrote:
    >> With their Gigaset range of ip phones Siemens provide a service called
    >> Gigaset.Net which allows registered phones to initiate VOIP calls to
    >> each other.
    >>
    >> I'm having a few problems getting this to work. It seems that other
    >> users can dial my Gigaset.Net number and connect to me but I can't
    >> connect to them, in fact they can't connect to themselves.
    >>
    >> I have checked all phones are registered and they are. So I'm now
    >> leaning towards the view that the service is pants, does anyone else
    >> have any experience.
    >>
    >> The G.722 codec that can be used with this service does appear to
    >> sound better than a normal voip call.

    > Don't think I've ever had anyone use it as I don'tknow anyone else that
    > has one.
    > You can try calling our gigaset in the shop. You'll get my dull
    > westcountry tones from the gigaset mailbox but the mailbox is set to
    > "verh high quality"
    > Look up "GymRatZ" I think that's the current one. There's an entry for
    > "GymRatZ UK" but that was set up on the original box that went back due
    > to faults caused by crap firmware rather than the box.
    > Try it and see what happens. You won't disturb anyone, the shutters are
    > down and the alarm is on.
    > :¬)

    Yep you are there, I won't phone you know. I can actually call out from
    all the phones. However my own phone is the only one that actually
    receives. When I try to connect to the others I get the dialling sound
    repeated and eventually IP Status Code 408.
     
    Nick, May 19, 2009
    #4
  5. Nick

    Nick Guest

    Tim wrote:
    > Nick wrote:
    >> I have checked all phones are registered and they are. So I'm now
    >> leaning towards the view that the service is pants, does anyone else
    >> have any experience.

    >
    > It is a bit pants. Works great if you are a public IP address.
    >
    > No so good through NAT - has a tendency for 1 way audio.
    >


    I am using NAT but I can't see how it is a NAT issue. The phones are
    registered but do not receive calls. I have my firewalls set up the same
    , I'm forwarding the same ports.

    > I'm not fussed by this because I have plenty of other SIP servers to use
    > which work a lot better.
    >

    Yes I suppose this is the answer. I did look at Vaxalot but I don't
    think SIP registration is included in the free account. Are there any
    free ones.
     
    Nick, May 19, 2009
    #5
  6. In article <0056e766$0$10376$>,
    Nick <> wrote:
    >Tim wrote:
    >> Nick wrote:
    >>> I have checked all phones are registered and they are. So I'm now
    >>> leaning towards the view that the service is pants, does anyone else
    >>> have any experience.

    >>
    >> It is a bit pants. Works great if you are a public IP address.
    >>
    >> No so good through NAT - has a tendency for 1 way audio.
    >>

    >
    >I am using NAT but I can't see how it is a NAT issue. The phones are
    >registered but do not receive calls. I have my firewalls set up the same
    >, I'm forwarding the same ports.


    NAT is the biggest PITA with SIP. There are many ways round it, but
    they're all "ways round it", alas.

    You may be better off NOT port-forwarding and using a STUN server. Also
    make sure your firewall/router/modem device does not have a SIP ALG
    enabled.

    >> I'm not fussed by this because I have plenty of other SIP servers to use
    >> which work a lot better.
    >>

    >Yes I suppose this is the answer. I did look at Vaxalot but I don't
    >think SIP registration is included in the free account. Are there any
    >free ones.


    So what you're after is basically a free, private "intercom" type of
    thing? Why not sign up for a sipgate account for each extension - then
    just dial the internal sipgate numbers..

    I don't think you even have to pay them to just have a basic account.

    Or run your own little asterisk box...

    Gordon
     
    Gordon Henderson, May 19, 2009
    #6
  7. Nick

    Nick Guest

    Gordon Henderson wrote:

    >
    > NAT is the biggest PITA with SIP. There are many ways round it, but
    > they're all "ways round it", alas.
    >
    > You may be better off NOT port-forwarding and using a STUN server. Also
    > make sure your firewall/router/modem device does not have a SIP ALG
    > enabled.
    >


    I'm not quite happy with that answer but I'm not clever enough to
    contradict you. It is a kind of interesting comment but I have other
    things I should be learning even though this question been nagging at me
    for about a year.

    >>> I'm not fussed by this because I have plenty of other SIP servers to use
    >>> which work a lot better.
    >>>

    >> Yes I suppose this is the answer. I did look at Vaxalot but I don't
    >> think SIP registration is included in the free account. Are there any
    >> free ones.

    >
    > So what you're after is basically a free, private "intercom" type of
    > thing? Why not sign up for a sipgate account for each extension - then
    > just dial the internal sipgate numbers..
    >
    > I don't think you even have to pay them to just have a basic account.
    >


    Well I'm just playing really. I had the phone and it worked so well I
    got a few for relatives. I noticed the Gigaset.net account and it works
    really well one way around but not the other so I wondered if there was
    an easy fix.

    I have a Sipgate account so I'll look at that. I don't really understand
    what a SIP registration server does I had naively assumed it was a
    lightweight directory type service that was only used to initiate the
    initial connection. I guess the regular heartbeat style re-registrations
    put to much load on a server to make it free.

    Thanks for the help.
     
    Nick, May 19, 2009
    #7
  8. Nick

    alexd Guest

    Nick wrote:

    > Gordon Henderson wrote:
    >
    >>
    >> NAT is the biggest PITA with SIP. There are many ways round it, but
    >> they're all "ways round it", alas.


    > I'm not quite happy with that answer but I'm not clever enough to
    > contradict you.


    Try his suggestion. See what happens. And roll on IPv6, so we can say
    goodbye and good riddance to NAT!

    >>> I did look at Vaxalot


    Vaxalot sucks ;-)

    > I have a Sipgate account so I'll look at that. I don't really understand
    > what a SIP registration server does I had naively assumed it was a
    > lightweight directory type service that was only used to initiate the
    > initial connection. I guess the regular heartbeat style re-registrations
    > put to much load on a server to make it free.


    The above generally only becomes an issue when NAT is involved. SIP is
    theoretically peer to peer, but when you throw NAT in, the endpoints can't
    see each other, so need a server ['proxy', 'back-to-back user agent'] to
    talk through. Add in buggy NAT and SIP ALG implementations and you've got a
    whole host of reasons for hilarity.

    Sipgate is one option, that gives you a PSTN number for each phone as a side
    effect. pbxes.com is another, although I'm not sure about how their pricing
    structure would work out for you.

    --
    <http://ale.cx/> (AIM:troffasky) ()
    13:26:23 up 12 days, 16:00, 2 users, load average: 0.19, 0.18, 0.14
    A few flakes working together can unleash an avalanche of destruction
     
    alexd, May 19, 2009
    #8
  9. In article <0005472f$0$2244$>,
    Nick <> wrote:
    >Gordon Henderson wrote:
    >
    >>
    >> NAT is the biggest PITA with SIP. There are many ways round it, but
    >> they're all "ways round it", alas.
    >>
    >> You may be better off NOT port-forwarding and using a STUN server. Also
    >> make sure your firewall/router/modem device does not have a SIP ALG
    >> enabled.

    >
    >I'm not quite happy with that answer but I'm not clever enough to
    >contradict you. It is a kind of interesting comment but I have other
    >things I should be learning even though this question been nagging at me
    >for about a year.


    NAT is a big issue with SIP.

    If you port-forward, you can only effectively have one device on the
    inside. You may be able to have multiple accounts on that device though.

    However, you need to tell the device what it's external IP address is.

    The reason for this is that SIP encapsulates the endpoints IP address
    inside the data/command packets it sends out.

    So by default, if the phone is behind NAT, on (eg.) 192.168.1.10, then
    it will encode that IP address inside the command packets. The recieving
    side will take that IP address and use it to send audio back. Since
    you can't get to that IP address except from behind your own NAT, it
    won't work. They may be able to hear you, but you won't be able to hear
    them. One way audio - and how many times have we read that about VoIP ...

    There are several ways forward - one is telling the phone what it's
    external IP address is, so it uses that IP address in the command packets
    it sends out. In this scenario you usually need to port-forward on the
    router too.

    Another way is by using STUN. STUN is just a service out on the 'net
    that allows a device to find out it's external IP address and how
    port re-mapping happens. The phone can then use this information when
    contacting a remote service.

    Another way is by the remote service using a SIP proxy that can do "deep
    packet inspection" and re-write the data packets so that it re-writes
    the IP address to be the same as the IP address it gets the data from
    before passing the packets on to the PBX.

    Some modem/router/firewall devices have what's called a SIP ALG -
    Application Layer Gateway. This tries to do the same "deep packet
    inspection" on outgoing SIP data and do the re-writing, while at the
    same time setting up more "memory" for the returning SIP/audio data.

    I've yet to find a router which has a fully working SIP ALG.

    Some routers even block SIP because they run their own VoiP
    services. Fritz boxes, Draytek 'v' series and others. eg. BT HomeHubs.

    >>>> I'm not fussed by this because I have plenty of other SIP servers to use
    >>>> which work a lot better.
    >>>>
    >>> Yes I suppose this is the answer. I did look at Vaxalot but I don't
    >>> think SIP registration is included in the free account. Are there any
    >>> free ones.

    >>
    >> So what you're after is basically a free, private "intercom" type of
    >> thing? Why not sign up for a sipgate account for each extension - then
    >> just dial the internal sipgate numbers..
    >>
    >> I don't think you even have to pay them to just have a basic account.

    >
    >Well I'm just playing really. I had the phone and it worked so well I
    >got a few for relatives. I noticed the Gigaset.net account and it works
    >really well one way around but not the other so I wondered if there was
    >an easy fix.
    >
    >I have a Sipgate account so I'll look at that. I don't really understand
    >what a SIP registration server does I had naively assumed it was a
    >lightweight directory type service that was only used to initiate the
    >initial connection. I guess the regular heartbeat style re-registrations
    >put to much load on a server to make it free.


    A "lightweight directory type service" is good enough :)

    It's either that, or you dial by IP address...

    Gordon
     
    Gordon Henderson, May 19, 2009
    #9
  10. In article <>, alexd <> wrote:
    >Nick wrote:
    >
    >> Gordon Henderson wrote:
    >>
    >>>
    >>> NAT is the biggest PITA with SIP. There are many ways round it, but
    >>> they're all "ways round it", alas.

    >
    >> I'm not quite happy with that answer but I'm not clever enough to
    >> contradict you.

    >
    >Try his suggestion. See what happens. And roll on IPv6, so we can say
    >goodbye and good riddance to NAT!


    And "Hello" to (lack of) firewall problems because by then everyone
    will have forgotten what firewalls are, having been "protected" by NAT
    for all this time...

    Gordon
     
    Gordon Henderson, May 19, 2009
    #10
    1. Advertising

Want to reply to this thread or ask your own question?

It takes just 2 minutes to sign up (and it's free!). Just click the sign up button to choose a username and then you can ask your own questions on the forum.
Similar Threads
  1. Linker3000
    Replies:
    1
    Views:
    555
  2. Ian Pawson
    Replies:
    16
    Views:
    2,822
    Dave Saville
    Aug 26, 2008
  3. Lobster
    Replies:
    7
    Views:
    6,157
  4. Dave Saville
    Replies:
    2
    Views:
    665
    Dave Saville
    Oct 13, 2008
  5. hammond
    Replies:
    1
    Views:
    1,858
Loading...

Share This Page