VoIP question

Discussion in 'VOIP' started by gmcleveney, Apr 1, 2004.

  1. gmcleveney

    gmcleveney Guest

    Is it possible to assign different level of service for a higher
    paying customer in a VoIP post paid scenario, based on the caller's
    telephone number.
    The end point telephones are regular POTS lines.
    Intent is to give different quality and service to subscribers based
    on the perminute charges they pay.
    If yes please explain how it is done..
     
    gmcleveney, Apr 1, 2004
    #1
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  2. gmcleveney

    SPD Guest

    Humm. Interesting one. If the endpoints are POTS phones
    do they connect to an Ethernet/IP network through some
    sort of terminal adapter, like an ATA186? I'll assume
    they do.

    You could set different 802.1p COS values for different
    customers and different IP precedence or diffserv code
    point values simultaneously at the terminal adapter for
    out bound calls.

    Your ingress Ethernet switch will need to be able to
    honor, enforce and remark the above markings. Also any
    other electronics in the path of the traffic will also
    need to respond to these markings appropriately.

    And of course inbound traffic, from the network core to
    the terminal adapater attached phone will need to be
    marked at it's source.






    gmcleveney wrote:
    > Is it possible to assign different level of service for a higher
    > paying customer in a VoIP post paid scenario, based on the caller's
    > telephone number.
    > The end point telephones are regular POTS lines.
    > Intent is to give different quality and service to subscribers based
    > on the perminute charges they pay.
    > If yes please explain how it is done..
     
    SPD, Apr 2, 2004
    #2
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  3. gmcleveney

    Hank Karl Guest

    On Thu, 01 Apr 2004 20:36:05 -0500, SPD <> wrote:
    > Humm. Interesting one. If the endpoints are POTS phones
    >do they connect to an Ethernet/IP network through some
    >sort of terminal adapter, like an ATA186? I'll assume
    >they do.
    >
    > You could set different 802.1p COS values for different
    >customers and different IP precedence or diffserv code
    >point values simultaneously at the terminal adapter for
    >out bound calls.
    >....

    Or you could allow those paying the highest rate to use G.711, and
    force those paying the lowest rate to use G.723.1 or some worse codec.
     
    Hank Karl, Apr 2, 2004
    #3
  4. gmcleveney

    SPD Guest

    Hank Karl wrote:
    > On Thu, 01 Apr 2004 20:36:05 -0500, SPD <> wrote:
    >
    >> Humm. Interesting one. If the endpoints are POTS phones
    >>do they connect to an Ethernet/IP network through some
    >>sort of terminal adapter, like an ATA186? I'll assume
    >>they do.
    >>
    >> You could set different 802.1p COS values for different
    >>customers and different IP precedence or diffserv code
    >>point values simultaneously at the terminal adapter for
    >>out bound calls.
    >>....

    >
    > Or you could allow those paying the highest rate to use G.711, and
    > force those paying the lowest rate to use G.723.1 or some worse codec.


    True although I assumed the author was concerned about
    variability on the IP network affecting call quality.
    The codec selection helps but may or may not provide the
    consistent user experience associated with a given
    fee level. COS,IP Prec & DiffServ may not either but
    in my experience this approach is more consistent.

    We have many different pieces of hardware, OS's and
    loads on the processor/memory on desktop devices. The
    encoding and decoding delay associated with some codecs
    combined with network congestion has created quality
    issues for users where other users of the same codec
    do not have problem (The other users have different
    hardware on their desktop)

    That's been my experience for what it's worth :)
     
    SPD, Apr 3, 2004
    #4
  5. gmcleveney

    Hank Karl Guest

    On Fri, 02 Apr 2004 20:21:52 -0500, SPD <> wrote:

    >Hank Karl wrote:
    >> On Thu, 01 Apr 2004 20:36:05 -0500, SPD <> wrote:
    >>
    >>> Humm. Interesting one. If the endpoints are POTS phones
    >>>do they connect to an Ethernet/IP network through some
    >>>sort of terminal adapter, like an ATA186? I'll assume
    >>>they do.
    >>>
    >>> You could set different 802.1p COS values for different
    >>>customers and different IP precedence or diffserv code
    >>>point values simultaneously at the terminal adapter for
    >>>out bound calls.
    >>>....

    >>
    >> Or you could allow those paying the highest rate to use G.711, and
    >> force those paying the lowest rate to use G.723.1 or some worse codec.

    >
    >True although I assumed the author was concerned about
    >variability on the IP network affecting call quality.
    >The codec selection helps but may or may not provide the
    >consistent user experience associated with a given
    >fee level. COS,IP Prec & DiffServ may not either but
    >in my experience this approach is more consistent.
    >

    I read it a different way: the author wanted to connect two analog
    phones, so his service would be doing the codec.

    Codec selection absolutely affects call quality. See the MOS table at
    http://www.cisco.com/warp/public/788/voip/codec_complexity.html
    Cisco claims their gear has:
    G.711 has a MOS of 4.1 (other estimates place it at 4.2 or 4.4) and
    compression delay of .75 mSec.
    G.723.1 at 5.3K has a MOS of 3.65 and compression delay of 30 mSec.
    You'll also have some delay in collecting the samples, which varies
    with codec and is sometimes a user-specifiable option.

    Delay affects perceived voice quality, and often a delay of under 100
    milliseconds is the goal for toll-quality VoIP.

    So G.711 sounds better and may have a lower delay, assuming no jitter
    or packet loss, and equal network delays between the two endpoints.

    >We have many different pieces of hardware, OS's and
    >loads on the processor/memory on desktop devices. The
    >encoding and decoding delay associated with some codecs
    >combined with network congestion has created quality
    >issues for users where other users of the same codec
    >do not have problem (The other users have different
    >hardware on their desktop)
    >
    >That's been my experience for what it's worth :)

    I agree, but will add that the network congestion causes packet loss,
    and excess jitter (which leads to packets being discarded by the
    jitter buffer). Packet loss and discard affect different codecs in
    different ways. It's unclear how much control over his network the
    author had. If you own the network, you can manage it so that jitter
    and loss are not problems.
     
    Hank Karl, Apr 3, 2004
    #5
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