DID with SIP

Discussion in 'UK VOIP' started by Brian A, Mar 5, 2006.

  1. Brian A

    Brian A Guest

    Anyone know a cheap source of a DID number that can be diverted to SIP
    addresses ?
    Remove 'no_spam_' from email address.
     
    Brian A, Mar 5, 2006
    #1
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  2. Brian A

    Brian A Guest

    On Sun, 05 Mar 2006 21:50:12 GMT, Brian A
    <> wrote:

    >Anyone know a cheap source of a DID number that can be diverted to SIP
    >addresses ?

    I meant to say UK GEOGRAPHIC DID number.


    Remove 'no_spam_' from email address.
     
    Brian A, Mar 5, 2006
    #2
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  3. Brian A

    Paul Cupis Guest

    Brian A wrote:
    > Anyone know a cheap source of a DID number that can be diverted to SIP
    > addresses ?


    Define cheap.
     
    Paul Cupis, Mar 5, 2006
    #3
  4. Brian A wrote:
    > Anyone know a cheap source of a DID number that can be diverted to SIP
    > addresses ?


    Generally, you'll find that most providers worth their salt won't
    redirect to sip address. Mostly because this could be used (depending
    on the platform that's implemented) to create a denial of service on
    the server, either intentionally or unintentionally.

    Calls could be:

    - inadvertantly bounced between supliers
    - sent to an unroutable address
    - sent to an address (sip:) that redirects to another address
    (sip:) that redirects to the first address (sip:), ergo,
    routing loop.
    - plethora of other examples.

    If someone were to flood that number with calls, the services could
    easily overload or run out of memory or whatever, since it can be
    paticularly difficult to filter these calls[1].

    There may be one or two suppliers though. I can't quote any off the top
    of my head.

    ~ Rich

    [1] Some providers remove SIP headers in order to hide infrastructure.
     
    Richard Smith, Mar 7, 2006
    #4
  5. Brian A

    Ivor Jones Guest

    "Brian A" <> wrote in message
    news:
    > On Sun, 05 Mar 2006 21:50:12 GMT, Brian A
    > <> wrote:
    >
    > > Anyone know a cheap source of a DID number that can be
    > > diverted to SIP addresses ?

    > I meant to say UK GEOGRAPHIC DID number.


    www.ipkall.com provide free US numbers (and very good they are too..!),
    but I can't think of a UK equivalent offhand.

    Ivor
     
    Ivor Jones, Mar 7, 2006
    #5
  6. "Richard Smith" <> wrote in message
    news:...
    > Brian A wrote:
    >> Anyone know a cheap source of a DID number that can be diverted to SIP
    >> addresses ?

    >
    > Generally, you'll find that most providers worth their salt won't
    > redirect to sip address. Mostly because this could be used (depending
    > on the platform that's implemented) to create a denial of service on
    > the server, either intentionally or unintentionally.
    >
    > ...


    Am I missing something. I thought the OP was asking for a source of UK
    numbers which could be assigned to a SIP phone/device - like Sipgate,
    Gradwell and countless other UK suppliers do.

    --
    Thomas Sandford
     
    Thomas Sandford, Mar 7, 2006
    #6
  7. Brian A

    Ivor Jones Guest

    "Thomas Sandford" <${thomas/03$}@paradisegreen.co.uk> wrote
    in message news:440d684d$0$1174$
    > "Richard Smith" <> wrote in message
    > news:...
    > > Brian A wrote:
    > > > Anyone know a cheap source of a DID number that can
    > > > be diverted to SIP addresses ?

    > >
    > > Generally, you'll find that most providers worth their
    > > salt won't redirect to sip address. Mostly because this
    > > could be used (depending on the platform that's
    > > implemented) to create a denial of service on the
    > > server, either intentionally or unintentionally. ...

    >
    > Am I missing something. I thought the OP was asking for a
    > source of UK numbers which could be assigned to a SIP
    > phone/device - like Sipgate, Gradwell and countless other
    > UK suppliers do.


    He said SIP *address* not SIP device. I read it as a number that could be
    directed to an *existing* SIP address.

    www.ipkall.com do this for US phone numbers, but I can't think of a UK
    equivalent offhand.

    Ivor
     
    Ivor Jones, Mar 7, 2006
    #7
  8. Brian A

    Paul Cupis Guest

    Richard Smith wrote:
    > Brian A wrote:
    >>Anyone know a cheap source of a DID number that can be diverted to SIP
    >>addresses ?

    >
    > Generally, you'll find that most providers worth their salt won't
    > redirect to sip address. Mostly because this could be used (depending
    > on the platform that's implemented) to create a denial of service on
    > the server, either intentionally or unintentionally.
    >
    > Calls could be:
    >
    > - inadvertantly bounced between supliers


    Can be done with PSTN, what's special about doing it with VoIP?

    > - sent to an unroutable address


    I'm fairly sure you can do this with PSTN.

    > - sent to an address (sip:) that redirects to another address
    > (sip:) that redirects to the first address (sip:), ergo,
    > routing loop.


    You can do this with PSTN.

    > - plethora of other examples.


    Any which are "impossible" to do on PSTN, or which otherwise make VoIP
    special?
     
    Paul Cupis, Mar 7, 2006
    #8
  9. "Ivor Jones" <> wrote in message
    news:...
    >
    >
    > "Thomas Sandford" <${thomas/03$}@paradisegreen.co.uk> wrote
    > in message news:440d684d$0$1174$
    >> "Richard Smith" <> wrote in message
    >> news:...
    >> > Brian A wrote:
    >> > > Anyone know a cheap source of a DID number that can
    >> > > be diverted to SIP addresses ?
    >> >
    >> > Generally, you'll find that most providers worth their
    >> > salt won't redirect to sip address. Mostly because this
    >> > could be used (depending on the platform that's
    >> > implemented) to create a denial of service on the
    >> > server, either intentionally or unintentionally. ...

    >>
    >> Am I missing something. I thought the OP was asking for a
    >> source of UK numbers which could be assigned to a SIP
    >> phone/device - like Sipgate, Gradwell and countless other
    >> UK suppliers do.

    >
    > He said SIP *address* not SIP device. I read it as a number that could be
    > directed to an *existing* SIP address.
    >
    > www.ipkall.com do this for US phone numbers, but I can't think of a UK
    > equivalent offhand.


    Ah - right - I do see the distinction (though I'm not convinced that the OP
    wasn't asking the wring question in that case!). Gradwell _do_ provide the
    type of forwarding you refer to:

    http://www.gradwell.com/voip/asterisk.php

    (it talks about asterisk, but the SIP inbound would/should work with any SIP
    address).
    [I dare say other providers exist, but I _know_ Gradwell do it].

    --
    Thomas Sandford
     
    Thomas Sandford, Mar 7, 2006
    #9
  10. Brian A

    Ivor Jones Guest

    "Thomas Sandford" <${thomas/03$}@paradisegreen.co.uk> wrote
    in message news:440da03c$0$1168$
    > "Ivor Jones" <> wrote in message
    > news:...


    [snip]

    > > He said SIP *address* not SIP device. I read it as a
    > > number that could be directed to an *existing* SIP
    > > address. www.ipkall.com do this for US phone numbers, but I
    > > can't think of a UK equivalent offhand.

    >
    > Ah - right - I do see the distinction (though I'm not
    > convinced that the OP wasn't asking the wring question in
    > that case!). Gradwell _do_ provide the type of forwarding
    > you refer to:
    > http://www.gradwell.com/voip/asterisk.php
    >
    > (it talks about asterisk, but the SIP inbound
    > would/should work with any SIP address).
    > [I dare say other providers exist, but I _know_ Gradwell
    > do it].


    Thanks, I hadn't spotted that. Pity there isn't a free one like ipkall
    though ;-)

    Ivor
     
    Ivor Jones, Mar 7, 2006
    #10
  11. Brian A

    Brian A Guest

    On Tue, 7 Mar 2006 12:07:15 -0000, "Ivor Jones"
    <> wrote:

    >
    >
    >"Thomas Sandford" <${thomas/03$}@paradisegreen.co.uk> wrote
    >in message news:440d684d$0$1174$
    >> "Richard Smith" <> wrote in message
    >> news:...
    >> > Brian A wrote:
    >> > > Anyone know a cheap source of a DID number that can
    >> > > be diverted to SIP addresses ?
    >> >
    >> > Generally, you'll find that most providers worth their
    >> > salt won't redirect to sip address. Mostly because this
    >> > could be used (depending on the platform that's
    >> > implemented) to create a denial of service on the
    >> > server, either intentionally or unintentionally. ...

    >>
    >> Am I missing something. I thought the OP was asking for a
    >> source of UK numbers which could be assigned to a SIP
    >> phone/device - like Sipgate, Gradwell and countless other
    >> UK suppliers do.

    >
    >He said SIP *address* not SIP device. I read it as a number that could be
    >directed to an *existing* SIP address.

    Ivor you are correct! That is exactly what I meant.
    >
    >www.ipkall.com do this for US phone numbers, but I can't think of a UK
    >equivalent offhand.

    Yes, it was something similar I was looking for.
    Remove 'no_spam_' from email address.
     
    Brian A, Mar 7, 2006
    #11
  12. Brian A

    Ivor Jones Guest

    "Brian A" <> wrote in message
    news:
    > On Tue, 7 Mar 2006 12:07:15 -0000, "Ivor Jones"
    > <> wrote:


    [snip]

    > > He said SIP *address* not SIP device. I read it as a
    > > number that could be directed to an *existing* SIP
    > > address.

    > Ivor you are correct! That is exactly what I meant.


    I see from another post in this thread that Gradwell offer this, but it's
    not free (£3/month).

    Ivor
     
    Ivor Jones, Mar 8, 2006
    #12
  13. Paul Cupis wrote:
    > > - inadvertantly bounced between supliers

    >
    > Can be done with PSTN, what's special about doing it with VoIP?


    Sure, however with the PSTN, this doesn't cause that much of a problem,
    mostly due to the fact that PSTN interconnections are limited to a
    fixed number of active channels.

    You buy an E1 and your destination has an E1, assuming you're the only
    person using those links, you're only going to be able to bounce a call
    back and fourth 15 times before you get an engaged tone. So the
    switching platform that the provider uses in between is safe and it's
    unlikely to take out the entire exchange.

    The reason VoIP is a unique case is because typically there aren't such
    limits... And since SIP Packets are negligable network traffic the
    probability of exhausting resources before bandwidth is nearer to 1.

    > > - sent to an unroutable address

    >
    > I'm fairly sure you can do this with PSTN.


    Of course you can, however sending a call to a wrong number gives you
    an NU tone. Whereas with voip, sending a number to an unroutable
    address causes the outbound connection to do DNS lookups. DNS timeout +
    someone flooding a destination with calls = resource shortage, ergo
    Denial Of Service.

    > > - sent to an address (sip:) that redirects to another address
    > > (sip:) that redirects to the first address (sip:), ergo,
    > > routing loop.

    >
    > You can do this with PSTN.


    Yep, see comments above about available channels. On VoIP this will
    loop endlessly until the platform runs out of resources to allocate.
    This is exasperated by the fact that providers remove Via headers from
    SIP packets to hide infrastructure.

    > > - plethora of other examples.

    >
    > Any which are "impossible" to do on PSTN, or which otherwise make VoIP
    > special?


    Pretty much all of them... Mostly due to the PSTN having that small
    property of bandwidth limitation.

    --
    Rich
     
    Richard Smith, Mar 13, 2006
    #13
  14. Richard Smith wrote:

    >>> - sent to an address (sip:) that redirects to another address
    >>>(sip:) that redirects to the first address (sip:), ergo,
    >>>routing loop.

    >>
    >>You can do this with PSTN.

    >

    Err, with SIP isn't there generally a handover, so that only the
    endpoints are talking to each other? (maybe excluding RTP traffic).
     
    Thomas Kenyon, Mar 13, 2006
    #14
  15. Thomas Kenyon wrote:
    > Err, with SIP isn't there generally a handover, so that only the
    > endpoints are talking to each other? (maybe excluding RTP traffic).


    You can set it up that way, however most ITSPs need to maintain control
    of the call signalling, to cut you off when you go over credit for
    example.

    Don't forget that SIP-to-SIP calls cost money too, just not on the
    magnitude of PSTN.

    ~ Rich
     
    Richard Smith, Mar 13, 2006
    #15
  16. Brian A

    Paul Cupis Guest

    Richard Smith wrote:
    > Paul Cupis wrote:
    >>> - inadvertantly bounced between supliers

    >>
    >>Can be done with PSTN, what's special about doing it with VoIP?

    >
    > Sure, however with the PSTN, this doesn't cause that much of a problem,
    > mostly due to the fact that PSTN interconnections are limited to a
    > fixed number of active channels.
    >
    > You buy an E1 and your destination has an E1, assuming you're the only
    > person using those links, you're only going to be able to bounce a call
    > back and fourth 15 times before you get an engaged tone. So the
    > switching platform that the provider uses in between is safe and it's
    > unlikely to take out the entire exchange.
    >
    > The reason VoIP is a unique case is because typically there aren't such
    > limits... And since SIP Packets are negligable network traffic the
    > probability of exhausting resources before bandwidth is nearer to 1.


    So you are saying that a DoS can be caused on both VoIP and PSTN. That
    is what I said.

    >>> - sent to an unroutable address

    >>
    >>I'm fairly sure you can do this with PSTN.

    >
    > Of course you can, however sending a call to a wrong number gives you
    > an NU tone. Whereas with voip, sending a number to an unroutable
    > address causes the outbound connection to do DNS lookups. DNS timeout +
    > someone flooding a destination with calls = resource shortage, ergo
    > Denial Of Service.


    Diverting a number to an unroutable PSTN destination and flooding the
    destination with calls can also cause a resource shortage, unless the
    originating telco is default-free.

    >>> - sent to an address (sip:) that redirects to another address
    >>>(sip:) that redirects to the first address (sip:), ergo,
    >>>routing loop.

    >>
    >>You can do this with PSTN.

    >
    > Yep, see comments above about available channels. On VoIP this will
    > loop endlessly until the platform runs out of resources to allocate.
    > This is exasperated by the fact that providers remove Via headers from
    > SIP packets to hide infrastructure.
    >
    >>> - plethora of other examples.

    >>
    >>Any which are "impossible" to do on PSTN, or which otherwise make VoIP
    >>special?

    >
    > Pretty much all of them... Mostly due to the PSTN having that small
    > property of bandwidth limitation.


    It seems that all the ones mentioned so far can be used to cause DoS on
    both PSTN and VoIP.
     
    Paul Cupis, Mar 13, 2006
    #16
  17. Brian A

    Paul Cupis Guest

    Thomas Kenyon wrote:
    > Richard Smith wrote:
    >>>> - sent to an address (sip:) that redirects to another address
    >>>> (sip:) that redirects to the first address (sip:), ergo,
    >>>> routing loop.
    >>>
    >>> You can do this with PSTN.

    >
    > Err, with SIP isn't there generally a handover, so that only the
    > endpoints are talking to each other? (maybe excluding RTP traffic).


    There can be, but there doesn't have to be.
     
    Paul Cupis, Mar 13, 2006
    #17
  18. (sorry if anyone got this twice, for some reason this message i posted
    6 days ago never seemed to propogate - certianly not to google, so
    re-posting)

    Paul Cupis wrote:

    >> Calls could be:
    >>
    >> - inadvertantly bounced between supliers

    >
    > Can be done with PSTN, what's special about doing it with VoIP?


    You can tell when a call has been diverted over ss7. RFC3261 doesn't
    specifiy a way directly, only via the [CC-]Diversion header (which very
    few vendors support as of yet, and doesn't have a ratified standard
    last time i checked).

    >> - sent to an unroutable address

    >
    > I'm fairly sure you can do this with PSTN.


    what about causing delays in DNS lookups, leading a invite server
    transaction on verrry slowly, and lots of other things i've not thought
    of so that you can consume stupid amounts of resouces?

    combine that with the above, and the below, and you could have a lot of
    "fun"!...

    >> - sent to an address (sip:) that redirects to another address
    >> (sip:) that redirects to the first address (sip:), ergo,
    >> routing loop.

    >
    > You can do this with PSTN.


    Can you explode a single call into 270 calls (that's
    1,180,591,620,717,411,303,424 calls) in the PSTN? i'd be pretty
    impressed if you could. you can with SIP, in very little time indeed
    thanks to forking.

    >> - plethora of other examples.

    >
    > Any which are "impossible" to do on PSTN, or which otherwise make VoIP
    > special?


    Lots and lots and lots and lots and ... lots. mostly all are related
    to DOS attacks and the fact that many PSTN<->SIP gateways are so badly
    designed i'm not going to discuss them here, but will happily explain
    in person next time we bump into each other at a free pissup :)

    PSTN equipment has been around for many centuries, SIP has, in it's
    current revision been around since June 2002 (minus the plethora of
    extensions added since). That's not enough time to work out all the
    security issues that are introduced in the complex networks that are
    being built using SIP.

    If you've got a single SIP device/gateway, you probably don't need to
    care too much. If you have anything more, then you need to take a long
    hard think about security issues SIP introduce, or we're going to be in
    a pretty bad situation in a few years (if not sooner), once the black
    hats realise the potential SIP has in aiding them. and don't just
    think your vendor will think about these for you, because most i've
    spoken to don't seem to care to much, as they're seemingly developing
    with the target customers being in private networks.

    SIP is NOT a Q931/SS7 replacement. don't try and compare the two.

    ~ Theo
     
    Theo Zourzouvillys, Mar 13, 2006
    #18
  19. > PSTN equipment has been around for many centuries

    .... and of course, i did mean decades, not centruries :)

    ~ Theo
     
    Theo Zourzouvillys, Mar 13, 2006
    #19
  20. Paul Cupis wrote:
    > > Sure, however with the PSTN, this doesn't cause that much of a problem,
    > > mostly due to the fact that PSTN interconnections are limited to a
    > > fixed number of active channels.
    > >
    > > You buy an E1 and your destination has an E1, assuming you're the only
    > > person using those links, you're only going to be able to bounce a call
    > > back and fourth 15 times before you get an engaged tone. So the
    > > switching platform that the provider uses in between is safe and it's
    > > unlikely to take out the entire exchange.
    > >
    > > The reason VoIP is a unique case is because typically there aren't such
    > > limits... And since SIP Packets are negligable network traffic the
    > > probability of exhausting resources before bandwidth is nearer to 1.

    >
    > So you are saying that a DoS can be caused on both VoIP and PSTN. That
    > is what I said.


    No, what I'm saying is that it's alot easier to do it and cause severe
    problems from the VoIP provider point of view. A denial of service for
    a customer on an E1 is unlikely to kill the provider.

    My original comments about denial of service and other phenomenon
    stated that it was in direct relation to a service provider, and that
    services providers worth their salt would not allow redirect to SIP
    addresses.

    Dragging my original comments out of context and twisting them around
    to suit your needs, does not make you intelligent enough to understand
    the concept. As Theo said, there are features and extensions to the SIP
    RFCs that are potentially very damaging if taken advantage of. Forking
    for example.

    The amount of problems a user can cause on the PSTN is limited to the
    amount of bandwidth they have available. Yes there are DOS attacks that
    you *COULD* do on the PSTN, but their impact would be limited mostly to
    the victim, and not the provider equipment and the provider would jump
    down their throat as soon as they found out.

    If you can demonstrate a situation where someone with an Analog Phone
    can take out a BT exchange or NTL Exchange, then I will eat my hat[1].

    Don't confuse ignorance with stupidity, and don't try to compare SIP
    and VoIP to q.931/SS7.

    ~ Richard

    [1] Although I may need to purchase one since I don't own a hat.
     
    Richard Smith, Mar 13, 2006
    #20
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