VOIP help needed

Discussion in 'VOIP' started by john, Dec 6, 2004.

  1. john

    john Guest

    Hello all:

    I have a few questions and would appreciate
    some assistance:

    We have a T1 between a router in the Philippine
    and US.

    1. Using G723 Codec, how many concurrent calls
    can be supported over this T1, not looking for
    theoretical answer, more like practical real
    life answer.

    2. Since this is a "direct T1" (The Telco told us so!) from
    there to here, any "compression" "queuing" "QOS" that
    I need to do or would like to do to improve call
    quality? I'm thinking maybe doing cRTP on the routers
    to improve bandwidth.

    Thanks,
    John.
     
    john, Dec 6, 2004
    #1
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  2. john wrote:
    > Hello all:
    >
    > I have a few questions and would appreciate
    > some assistance:
    >
    > We have a T1 between a router in the Philippine
    > and US.
    >
    > 1. Using G723 Codec, how many concurrent calls
    > can be supported over this T1, not looking for
    > theoretical answer, more like practical real
    > life answer.
    >
    > 2. Since this is a "direct T1" (The Telco told us so!) from
    > there to here, any "compression" "queuing" "QOS" that
    > I need to do or would like to do to improve call
    > quality? I'm thinking maybe doing cRTP on the routers
    > to improve bandwidth.
    >
    > Thanks,
    > John.
    >
    >


    Now I coulda sworn you said they said the t1 line was:
    > It was technically provisioned via our Freeway Service which is our direct
    > connection to LA.
    >
    > Freeway runs on ATM over C2C-JUNC cable routes.


    Which means it definitely is NOT a direct line. It goes over ATM. If it
    goes over ATM it ain't a direct line. The two are mutually exclusive.

    Now as far as tweaking goes, what is best depends heavily on
    1) what type of VOIP are you running? (ie: h323, sip. etc) and
    2) what type of routers are on both ends of the T1? What type of QOS do
    they support? Diffserv, 802.1q priority?
    Basically you want to set up the routers to look for characteristics in
    the packets that will identify them as VOIP traffic (ie: upd packets to
    or from specific ports)and make them higher priority, and put them into
    a faster queue. How this is done varies greatly depending on the type of
    VOIP and type of router.

    BUT, if you are still getting > 250ms ping times over the T1, your call
    quality is going to be pretty crappy. Some of it may be helped if you
    crank up the jitter biffers on the VOIP system to compensate. But with a
    delay of a quarter second, users WILL notice a lag in the converstation.

    QUESTION: when the providor tells you the T1 is a direct line, do they
    mean direct line to the end site at the US? Or direct to the internet or
    some other endpoint? what you want is direct to the end site, no ATM in
    between.
     
    T. Sean Weintz, Dec 6, 2004
    #2
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