delay of 70ms

Discussion in 'VOIP' started by bob@coolgroups.com, May 28, 2006.

  1. Guest

    I have a pair of Cisco 3725 routers that I'm talking through
    using VOIP. I measured a mouth to ear delay of 70ms with a
    standard config. Is this what I should be getting? Is there a way
    to make the delay 0?
     
    , May 28, 2006
    #1
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  2. R-Guy Guest

    <> wrote in message
    news:...
    >I have a pair of Cisco 3725 routers that I'm talking through
    > using VOIP. I measured a mouth to ear delay of 70ms with a
    > standard config. Is this what I should be getting? Is there a way
    > to make the delay 0?
    >


    Are they connected by LAN or WAN?
    What codec and packetization period are you using?
    What type of voice terminal are you using?
     
    R-Guy, May 28, 2006
    #2
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  3. Guest

    LAN,

    G.711 ulaw 20ms,

    standard POTS phone


    R-Guy wrote:
    > <> wrote in message
    > news:...
    > >I have a pair of Cisco 3725 routers that I'm talking through
    > > using VOIP. I measured a mouth to ear delay of 70ms with a
    > > standard config. Is this what I should be getting? Is there a way
    > > to make the delay 0?
    > >

    >
    > Are they connected by LAN or WAN?
    > What codec and packetization period are you using?
    > What type of voice terminal are you using?
     
    , May 29, 2006
    #3
  4. <> wrote in message
    news:...
    > R-Guy wrote:
    >> <> wrote in message
    >> news:...
    >> >I have a pair of Cisco 3725 routers that I'm talking through
    >> > using VOIP. I measured a mouth to ear delay of 70ms with a
    >> > standard config. Is this what I should be getting? Is there a way
    >> > to make the delay 0?
    >> >

    >>
    >> Are they connected by LAN or WAN?
    >> What codec and packetization period are you using?
    >> What type of voice terminal are you using?

    >
    > LAN,
    >
    > G.711 ulaw 20ms,
    >
    > standard POTS phone


    With 20ms packetization, it will be impossible to get less than 20ms delay,
    because that's how much data the gateway has to collect before sending a
    packet. There's also typically 10-20ms burned in the codec/DSP's pipeline.
    On top of that, the receiver's jitter buffer will add a few tens of ms; most
    jitter buffers, even adaptive ones, are tuned with a minimum of a few tens
    of ms because it doesn't hurt, and 40ms or so of buffering will hide all but
    the worst unless you have no QoS at all over a skinny WAN pipe.

    70ms isn't bad, and no human should ever notice it. Don't worry about
    latency until it hits 150ms.

    S

    --
    Stephen Sprunk "Stupid people surround themselves with smart
    CCIE #3723 people. Smart people surround themselves with
    K5SSS smart people who disagree with them." --Aaron Sorkin


    *** ***
     
    Stephen Sprunk, May 29, 2006
    #4
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