Solved: Voice Droping in DSL branches - I am Stucking !!!!!!!

eemara949 · ‎10-23-2008

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,

l.mourits · ‎11-04-2008

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?

View solution in original post

sebastiandm · ‎10-23-2008

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...

eemara949 · ‎10-24-2008

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

l.mourits · ‎10-24-2008

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

robertdm1973 · ‎10-24-2008

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)

CHRIS CHARLEBOIS · ‎10-24-2008

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.

l.mourits · ‎10-25-2008

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)

robertdm1973 · ‎10-26-2008

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.

l.mourits · ‎11-05-2008

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

eemara949 · ‎10-25-2008

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

l.mourits · ‎10-26-2008

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

l.mourits · ‎10-26-2008

ps. can you post your access-list (MOA) as it may contain clues about your problem?

Thanks,

Leo

eemara949 · ‎10-26-2008

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

eemara949 · ‎11-04-2008

Any help Mr. Leo

l.mourits · ‎11-04-2008

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?