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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