Hello ,
I am trying to send a test packet to the router for which my router has
received an update packet containg the route. So that i can test the
route and check its validity for any false route information. also it
will be helpfull to keep eye on security issues. Plz comment. Is this
Assuming your providers give you a new modem which is already NAT'ing
the LAN side of the modem and you are plugging that into multiple NIC's
on your linux router like;
-modem-pub - modem-priv - linux-eth0
-modem-pub - modem-priv - linux-eth1
-linux-eth3 - LAN switch
1) Configure VRRP
Hi, NANOGers.
Just a FYI - we at Team Cymru are upgrading some of our infrastructure
today. This will result in partial and complete outages for most of
the day. We will be back online, new and improved, by the end of the
day.
Thanks!
Rob, for Team Cymru.
--
Rob Thomas
Team Cymru
One thing to note here is that while VoIP flows are low volume on a
bits-per-second basis, they push substantially more packets per
kilobit than other traffic types - as much as 50pps per 82Kbps flow.
And I have seen cases of older line cards approaching their pps
limits when handling
Chris Woodfield wrote:
One thing to note here is that while VoIP flows are low volume on a
bits-per-second basis, they push substantially more packets per kilobit
than other traffic types - as much as 50pps per 82Kbps flow. And I have
seen cases of older line cards approaching their
On 18/12/05, Chris Woodfield [EMAIL PROTECTED] wrote:
One thing to note here is that while VoIP flows are low volume on abits-per-second basis, they push substantially more packets per
kilobit than other traffic types - as much as 50pps per 82Kbps flow.And I have seen cases of older line cards
Joe Maimon wrote:
Chris Woodfield wrote:
One thing to note here is that while VoIP flows are low volume on a
bits-per-second basis, they push substantially more packets per
kilobit than other traffic types - as much as 50pps per 82Kbps flow.
And I have seen cases of older line cards
On Sun, 18 Dec 2005, Joe Maimon wrote:
Something about intelligent edges? The payload length of voip applications
often has a lot to do with rtt. Adapting payload length to the actuall
average rtt could have a positive effect on pps throughput.
What is your suggestion? High latency
Jay Hennigan wrote:
VoIP by design will have high PPS per connection as opposed to data flows.
At 20 ms sample rates you have 50 pps regardless of the CODEC or algorithm.
Increasing the time per sample to 40 ms would cut this in half but the
added
latency would result in degraded
Mikael Abrahamsson wrote:
On Sun, 18 Dec 2005, Joe Maimon wrote:
Something about intelligent edges? The payload length of voip
applications often has a lot to do with rtt. Adapting payload length
to the actuall average rtt could have a positive effect on pps
throughput.
What is your
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