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Some things that might help:
1. Are you using CODEC like G.729 that works at an
8Kbps bit rate? If not, you'll be hating life.
2. Bandwidth at full rate & Multilink PPP
(MLPPP) or FRF.12 will still be 17.2kbps.
3. You need QoS, even if you can burst to port
speed at 56kbps.
4. Consider LLQ for Frame Relay for your QoS, which
is now supported in IOS as of 12.1(2)T. We had to use this to get good
voice quality on a moderately saturated FR link with a 1.544mbps
CIR.
5. You may need QoS on your LAN if the links are
congested near the 3640. Look at QoS as an end-to-end
solution.
6. Perform ping tests from the 3640 to your remote
sites to get a better idea of how many ms the round trip is. Try pings
from both sides.
Things to keep in mind:
- VoIP overhead will kill you - you need to do
testing to figure out exactly how much bandwidth you are using.
- Good luck doing this over 56kbps, even if you
use the full line.
----- Original Message -----
Sent: Friday, November 10, 2000 1:08
PM
Subject: VOIP Troubles
All right guys I need some help....
I have been working with Cisco for a
while now on a VOIP issue. The problems lies in both call disconnects
and voice distortion. We have followed all the steps for traffic shaping
(QOS) and rtp header-compression but these do not seem to help. We have
150 remote sites all running 2600's with FXS modules that all come back into
the host site where we have 2 7206's. Each of these links are 56k
frame-relay links with 16k CIR running very few applications mostly small
transactions and Citrix clients. The call must then traverse two
internal Ethernet segments, routing through our 6509 backbone switch and then
into a 3640 before hitting the PBX. Cisco seems to think that we need to
increase our bandwidth to support the voice traffic, however, that is not
something I have been able to sell to the "powers that be". We sold this
idea on cutting cost and in our estimations for upping the CIR to even 32k
will be significant cost increase. Right now I am shaping to 16k with an
8k committed burst so at any one time I should be able to burst to
24k. Assuming that I am able to burst to port speed (56k) why would
I have call distortion unless there is some latency coming through the
ISP's switch? We also have another company site that also
comes back in this way and we have no problems with those calls.O.K.
that being said (and hopefully not too confusing to follow) here come the
questions:
1. Has anyone else implemented VOIP in
slow links successfully?
2. Is anyone else having QOS problems with
their VOIP implementations?
3. Do I need to prioritize the voice
traffic through the local network?
4. Has anyone tried turning off traffic
shaping and letting the voice and data compete for bandwidth?
Thanks in advance for your feedback!
Thanks,
Chris Boyd, CCNA
Network Support
Alex Lee, Inc.
120
4th Street SW
Hickory, NC 28601
(828) 323-4103
http://www.alexlee.com
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