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1) Yes, and we had some of the same issues.
The biggest was that we had to hard code the codec on all interfaces.
For some reason the default codec G729ar8 wasn't always being selected or
detected and since it's the default you can't hard code it. We had to opt
for G729r8. We had point to point 56K circuits so it was much easier to
deal with. Oh yea, we also tried to do a voice class to change the default
codec for a whole router, but we had a couple of routers that ignored the voice
class config, which is why we ended up hard coding every interface.
2) I think this is more an issue that IOS is
a work in progress. We've always had to use an Early Deployment release to
fix an issue. ( and in the process broke something else...)
3) Do it anyway.
4) Yep, and you'll be sorry you did. Even on
full point to point T1's, I've seen issues.
Just a suggestion, but have you considered putting
voice cards in the 7206's and going VOFR?
Rodgers Moore
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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