10-23-2008 02:51 AM - edited 03-15-2019 02:07 PM
Dear all,
We have full cisco unified communication enviroment, and everything is running smooth in the head office, but we have two remotes branches one is 1 Mbps DSL connection and the other is 1 Mbps WiMaX connection, we have one ip phones in the first site and 4 in the second one. We are suffering from voice droping in the calls between the head office and the remote sites and between the remote site with external PSTN.
We dont have any special traffic in our remote sites, they are only around 5 PCs and one IP phone in the firs site and 3 PCs and 4 IP phones in the second site.
Please we are suffering from this problem!! any body can help us??
Thanks & Best Regards,
Solved! Go to Solution.
11-04-2008 02:01 AM
Hi, Sorry, I've been busy. ACL looks fine. That leaves the configuration of network regions and locations. How did you configure these? What codec is used? What bandwidth do you allow?
10-23-2008 02:58 AM
Hi,
How does your call routing work ? Are your calls coming on the remote sites locally and breaking out across the WAN or going back out locally. What type of pstn circuit do you have FXO/E1/T1...
10-24-2008 02:09 AM
Hi my dear,
I dont think that the calls breaking out across the WAN because the ping is working fine from the HQ and remote site across the WAN.
We have E1 PSTN ciruit.
Appreciate your support
10-24-2008 03:59 AM
Let's see if I understand your environment first (as I'm sure others will wonder as well).
So you have a CCM cluster in the head office, and two branch offices connected via DSL to the head office. Then the phones in the branch office register to the CCM cluster and transport voice signaling and voice bearer traffic over the wan (bearer traffic for both internal as external calls, since you do not have PSTN on the branch office, in other words, if they dial out they leave out the PSTN in the head office). Correct?
Couple if questions:
How did you implement the VPNs between the offices? I would expect DMVPN like solution. If so, did you ensure the MTU sizings are correct?
What codec are you using between the region locations? Recommended is G729a for WAN links.
How did you configure your QoS on the WAN links (i.g. what queing method are you using, how do you gaurantee bandwidth, et cetera)?
If you happen to have no answer to any of the above questions, I think you need to start reading up about voice over encrypted WAN links, and voice engineering in general.
HTH,
Leo
10-24-2008 12:36 PM
Hi-
I have a similar problem where calls dropped on WAN link. Using the CCAPI debug, here's what showing on the log. What does cause code=38 stands for?
*Oct 24 18:15:32.472: //175/C8E86EA280B0/CCAPI/cc_api_call_disconnect_done:
Disposition=0, Interface=0x653EB79C, Tag=0x0, Call Id=175,
Call Entry(Disconnect Cause=38, Voice Class Cause Code=0, Retry Count=0)
10-24-2008 12:57 PM
Cause 38, I beleive, is a generic bad connection failure. Unfortunately, not very descriptive but it basically means that there was too much delay/jitter/lost packets/etc.
10-25-2008 06:30 AM
Similar to the other issue of the topic starter, the same questions arive if you see a generic code about bad network connectivity:
How are your WAN links provisioned (MPLS network, DMVPN over DSL, leased lines?????)
How did you implement QoS. How are you marking traffic? How do you handle bandwidh admission control? What did you do to ensure bandwidth guarantees?
If you don't know the answer to this, you probably should be better of not routing calls over the wan.
(ps. excuse me being cinical here, but I have seen too many people trying to implement voice over IP without QoS, and it always fails sooner or later)
10-26-2008 01:47 PM
If the WAN link do not support QOS, what's the best work-around to eliminate dropped calls?
Ours is an L2L VPN over a public Internet, both with big pipes (DS3).
Thanks.
11-05-2008 02:41 PM
If your wan link does not supporting QoS you should not even consider routing voice over it, is the one and only correct answer to this question.
But, if you have a DS3 your routing platform is probably a 3800 series or similar and you should be able to at least configure the correct priority queues. You could limit the bandwidth per class. If that is not an option, your second best is probably to implement car on your router (committed access rate).
HTH,
Leo
10-25-2008 11:31 PM
Dear Leo,
Correct as you said, we have the same enviroment.
we did NOT implement VPNs between the offices.
QoS already configured as following:"
class-map match-any VOIP
match access-group name MOA
!
!
policy-map QOS
class VOIP
priority percent 60
set ip precedence 5
class class-default
fair-queue
"
kindly advice
10-26-2008 12:57 AM
Hi,
K, setting priority and precedence is one thing, bandwidth admission control is the second item.
If I had to do such install, I would limit the class to a specific amount of bandwidth (number of simultaneous calls * 30kbps), and use region/location configs in CCM to select the correct codec and control bandwidth.
HTH,
Leo
10-26-2008 09:02 AM
ps. can you post your access-list (MOA) as it may contain clues about your problem?
Thanks,
Leo
10-26-2008 02:38 PM
this is my access list MOA
Extended IP access list MOA
10 permit tcp any any eq 1720
20 permit udp any any eq 5060
30 permit udp any any eq 2427
40 permit tcp any any eq 2428
50 permit tcp any any eq 2000 2002 (168311 matches)
60 permit tcp any any eq 5060
11-04-2008 01:54 AM
Any help Mr. Leo
11-04-2008 02:01 AM
Hi, Sorry, I've been busy. ACL looks fine. That leaves the configuration of network regions and locations. How did you configure these? What codec is used? What bandwidth do you allow?
Discover and save your favorite ideas. Come back to expert answers, step-by-step guides, recent topics, and more.
New here? Get started with these tips. How to use Community New member guide