The Trouble with NAT and VOIP

Discussion in 'VOIP' started by Rick Merrill, Feb 4, 2005.

  1. Rick Merrill

    Rick Merrill Guest

    This may shed some light for those who are trying to estabilish VOIP
    behind a NAT router. - RM

    from http://www.voip-info.org/wiki-NAT and VOIP
    :

    The Trouble with NAT and VOIP
    "In addition, the way in which conventional VoIP protocols are designed
    is also posing a problem to VoIP traffic passing through NAT.
    Conventional VoIP protocols only deal with the signalling of a telephone
    connection. The audio traffic is handled by another protocol and to make
    matters worse, the port on which the audio traffic is sent is random.
    The NAT router may be able to handle the signalling traffic, but it has
    no way of knowing that the audio traffic is related to the signalling
    and should hence be passed to the same device the signalling traffic is
    passed to. As a result, the audio traffic is simply discarded.

    "At first, for both the calling and the called party everything will
    appear just fine. The called party will see the calling party's Caller
    ID and the telephone will ring while the calling party will hear a
    ringing feedback tone at the other end. When the called party picks up
    the telephone, both the ringing and the associated ringing feedback tone
    at the other end will stop as one would expect. However, the calling
    party will not hear the called party (one way audio) and the called
    party may not hear the calling party either (no audio).

    "The issue of NAT Traversal is a major problem for the widespread
    deployment of VOIP. Yet, the issue is non-trivial and there are no
    simple solutions."
    Rick Merrill, Feb 4, 2005
    #1
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  2. Rick Merrill

    Jim Hatfield Guest

    On Fri, 04 Feb 2005 10:44:44 -0500, Rick Merrill <> wrote:

    >"The issue of NAT Traversal is a major problem for the widespread
    >deployment of VOIP. Yet, the issue is non-trivial and there are no
    >simple solutions."


    UDP hole-punching seems to work pretty well.

    See: http://www.pdos.lcs.mit.edu/~baford/nat/draft-ford-natp2p-00.txt

    --
    Jim Hatfield
    Jim Hatfield, Feb 4, 2005
    #2
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  3. Rick Merrill

    Dilbert2004 Guest

    Re: The Trouble with NAT and VOIP (solution)

    There is a simple solution! Edgewater Networks has a "box" that enables
    you to put your phones on private / NAT addresses. You don't have to
    punch holes in your firewall !

    http://www.edgewaternetworks.com

    Dilbert!


    Jim Hatfield wrote:
    > On Fri, 04 Feb 2005 10:44:44 -0500, Rick Merrill

    <> wrote:
    >
    > >"The issue of NAT Traversal is a major problem for the widespread
    > >deployment of VOIP. Yet, the issue is non-trivial and there are no
    > >simple solutions."

    >
    > UDP hole-punching seems to work pretty well.
    >
    > See: http://www.pdos.lcs.mit.edu/~baford/nat/draft-ford-natp2p-00.txt
    >
    > --
    > Jim Hatfield
    Dilbert2004, Feb 4, 2005
    #3
  4. Rick Merrill wrote:

    > "The issue of NAT Traversal is a major problem for the widespread
    > deployment of VOIP. Yet, the issue is non-trivial and there are no
    > simple solutions."


    'Simple Traversal of UDP Through NATs' (STUN) is pretty basic to
    configure. See
    http://www.zyxel.com/support/supportnote/p2002/app/ata_nat.htm
    for example.

    Arnold.
    Arnold Ligtvoet, Feb 4, 2005
    #4
  5. Re: The Trouble with NAT and VOIP (solution)

    On 4 Feb 2005 13:31:25 -0800, "Dilbert2004" <>
    wrote:

    >There is a simple solution! Edgewater Networks has a "box" that enables
    >you to put your phones on private / NAT addresses. You don't have to
    >punch holes in your firewall !
    >
    >http://www.edgewaternetworks.com


    It's just a little bit expensive. Per user, it seems to be cheaper to
    replace the end user's ATA with one on a public IP.

    There are no universal, cheap solutions. Even stuff like STUN and pin
    holing doesn't work for all.

    peter

    --
    peter gradwell. gradwell dot com Ltd. http://www.gradwell.com/
    -- engineering & hosting services for email, web and voip --
    -- http://www.peter.me.uk/ -- http://www.voip.org.uk/ --
    Peter Gradwell, Feb 4, 2005
    #5
  6. Rick Merrill

    stephen Guest

    "Rick Merrill" <> wrote in message
    news:...
    > This may shed some light for those who are trying to estabilish VOIP
    > behind a NAT router. - RM
    >
    > from http://www.voip-info.org/wiki-NAT and VOIP
    > :
    >
    > The Trouble with NAT and VOIP
    > "In addition, the way in which conventional VoIP protocols are designed
    > is also posing a problem to VoIP traffic passing through NAT.
    > Conventional VoIP protocols only deal with the signalling of a telephone
    > connection. The audio traffic is handled by another protocol and to make
    > matters worse, the port on which the audio traffic is sent is random.
    > The NAT router may be able to handle the signalling traffic, but it has
    > no way of knowing that the audio traffic is related to the signalling
    > and should hence be passed to the same device the signalling traffic is
    > passed to. As a result, the audio traffic is simply discarded.


    not true. the signalling protocol carries the info about which ports the
    specific connections will use - otherwise how is the call going to get
    connected correctly to the end point?

    for example, cisco PIX has a fixup protocol that does this for you for H323
    and SIP:
    http://www.cisco.com/en/US/products...s_configuration_example09186a00801fc74a.shtml

    >
    > "At first, for both the calling and the called party everything will
    > appear just fine. The called party will see the calling party's Caller
    > ID and the telephone will ring while the calling party will hear a
    > ringing feedback tone at the other end. When the called party picks up
    > the telephone, both the ringing and the associated ringing feedback tone
    > at the other end will stop as one would expect. However, the calling
    > party will not hear the called party (one way audio) and the called
    > party may not hear the calling party either (no audio).
    >
    > "The issue of NAT Traversal is a major problem for the widespread
    > deployment of VOIP. Yet, the issue is non-trivial and there are no
    > simple solutions."


    this is actually an issue for a firewall software supplier who cannot be
    bothered to write code to handle this particular protocol rather than a
    problem with the protocol.

    the long term fix is simple - if your firewall cant handle the voice
    protocol you need - take it back, and / or complain to the supplier.

    a year or 2 of that and the software will get fixed.
    --
    Regards

    Stephen Hope - return address needs fewer xxs
    stephen, Feb 5, 2005
    #6
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