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  • Call quality issues and MPLS

    Ok...got some call quality issues and wanted to see if anybody here had some ideas. Background info on our network.

    Got two sites up. MPLS connection between both sites via T1.

    Main site has the the following:
    PRI trunk
    Three T1's for MPLS and internet connectivity
    Sonicwall NAS2400 for firewall
    Cisco 2824 for the T1's
    Adtran Netvanta Switch
    SGT1k
    SG90

    Remote site has
    DSL for outbound internet traffic
    Sonicwall TZ180 for firewall
    T1 for the MPLS
    Cisco 1841 for the T1
    Adtran Netvanta Switch
    SG30

    Site to Site calls are great. Outbound and inbound calls to and from main site is great. Outbound and inbound calls to and from remote site is poor.

    They tested our MPLS circuit and everything came clean. Double checked the QOS on the MPLS circuit and it looks fine. Should I be looking any where else?

  • #2
    How are you routing the traffic from the LANs to the MPLS? Are you going through the Sonicwall or are you using your Layer3 device to route that to the MPLS router?

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    • #3
      Clarify how your remote site off-system calls are handled. Do they route throught the main site? Or do they use local trunks at the remote site (PRI, analog, etc). No SIP, right?

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      • #4
        LANS to MPLS is through the Sonicwall.

        Remote site calls are routed to the main site, and out of the main site through PRI trunks.

        Might have found a solution...and I'm looking at the DiffServ / ToS Byte (0-255) setting. The enginer said to enter in 184 as the value but for environments "Cisco routers to forward voice packets, this will translate to 46 (decimal)."

        Question is...what does that last part mean? The "(decimal)" is what confuses me. Do I enter in 46 as the value?

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        • #5
          Originally posted by DustySlider View Post
          LANS to MPLS is through the Sonicwall.

          Remote site calls are routed to the main site, and out of the main site through PRI trunks.

          Might have found a solution...and I'm looking at the DiffServ / ToS Byte (0-255) setting. The enginer said to enter in 184 as the value but for environments "Cisco routers to forward voice packets, this will translate to 46 (decimal)."

          Question is...what does that last part mean? The "(decimal)" is what confuses me. Do I enter in 46 as the value?
          You enter 184.

          Your problem may be how the Sonicwall is passing the traffic, and not itself prioritizing the traffic.

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          • #6
            Traffic

            Do you really need to go through the sonicwall at the remote site to get to the other office?

            It is normally not necessary to firewall MPLS point to point connections........

            if you really do go through the sonic wall, your issue will most likely be there. The cisco router and the mpls network may be honoring your QOS values, but the sonicwall may not be setup to do so.......

            easy test would be to bypass the sonicwall and see if the problem goes away.

            A more traditional setup would be

            lan to mpls through cisco router

            lan to internet through sonicwall

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            • #7
              Ok...it looks like a combo....

              Set the ToS to 184 and plugged the MPLS directly into the Adtran, calls are sounding better! I'll keep you guys posted next week...

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              • #8
                Ok. So calls generally are good. There are still some instances where the the company caller's voice will trail off a bit for a second or two. On the other hand, the company caller can hear the outside caller crystal clear all the time.

                Any suggestions, or is this about as good as it will get?

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                • #9
                  Qos

                  Look in the event log of the server.

                  are there messages about excessive packet loss? if so, see if they follow the pattern of your call issues.

                  If you have packet loss, the call quality will be hit and miss. It is only fixed by proper QOS.

                  once we know if you have packet loss reported, we can look further into the QOS config.

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                  • #10
                    Ok. No packet loss on the event viewer recently. I looked back for days and there was an event log for packet loss, but it was weeks ago. Since then, been clean.

                    Any thoughts?

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                    • #11
                      call

                      on the calls that still have a bit of quality issues, are the users on headsets, speakerphones, etc etc?

                      also, are these calls staying in the site, and going out of a t1 pri or analog line, or do they go over the wan, to another site, and then to the outside world?

                      What codec are you using for intra and inter site calls?

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                      • #12
                        Regular handsets.

                        These calls are going on MPLS to our main office, then from there out on a T1 PRI line.

                        Trying to find that out actually. I'm in the Director in the Call Control Options where the codecs used should be, but there's nothing there indicating what's in use right now. Is there somewhere else I can look for it?

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                        • #13
                          BTW, I set teh Max Inter-Site Jitter buffer to 20 and looks like quality improved a little bit. Are there any cons to setting it this low, or even lower?

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                          • #14
                            Mpls

                            If you still have any call quality issues, they are most likely going to be on the mpls side.

                            Most MPLS circuits have a "cap" on how much traffic they will prioritize. For example, some circuits are 256k cos. That means that if you are prioritizing your voice traffic, it only works until you hit the 256k mark. Everything after that is fair game.

                            you can usually pay more to bump it up to like 768k or so.

                            also, are you sure the carrier is even honoring your QOS tagging?

                            this could explain why you are having audio issues one way. it most of your data is going from the headquarters down to the branch, then your audio going that way will suffer. The audio going the other way may be just fine.

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                            • #15
                              Had 2nd level tech at Paetec monitor it and the voice data was tagging correctly on the QOS. Let me check on this voice cap priority.

                              I also switched to the PCMU8000 and call quality seems to have improved significantly. Only getting some complaints here and there, but generally seems to have really gotten better.

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