How to set Qos for VOIP in head office and branches

Discussion in '3CX Phone System - General' started by aslam, Jun 11, 2016.

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  1. aslam

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    We have 3CX phone system installed our head office and PSTN lines are connected to the system.Our branches are connected to head office through site to site VPN and head office running Internet with speed of 75 Mbps and branch is 12 Mbps. We are facing audio clarity issues at times and seems that happens in branches more frequently. Hope setting up Qos will help to rectify the problem. What network zone I suppose to choose if I set Qos for Head office voip service and Branches. Is it LAN to WAN or LAN to VPN?
     
  2. lneblett

    lneblett Well-Known Member

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    It might help if we knew the router make and model. However, I assume Sonicwall and I also assume the same in use at each remote site.

    QoS may or may not solve the issue depending on the cause. It is not clear if the connections are symmetrical or asymmetrical and if the call issues are heard by the caller, callee or both. If all remote sites are the same size and if all have the same issue and if the VPNs are used to route all traffic or merely the voice and/or some other subset of data.

    You may need to use BMW in conjunction with QoS to insure that there is enough bandwidth set aside for the phones.

    I have attached an article that explains how to set QoS for Net2Phone use. This may help you, but of course, the ports are different. I am guessing that you want QoS for voice along all paths for both ingress and egress so the set-up for the main and the remote would be different due to the interfaces used. I am by no means a Sonicwall expert, so if you have a maintenance or support agreement with Dell on the products, I suggest you engage them.

    My simple take on it is that you need both ingress and egress rules at all locations and at the main site I tend to think this would involve the LAN, WAN and VPN zones (all the interfaces are used for voice) and at the remote sites the LAN and VPN zones.

    Once you have the proper tagging in place (EF), then you should be able to use Wireshark to confirm the results.

    Not sure that I addressed the need, and as stated, I am no expert with Sonicwall, but hopefully the article will shed some light.
    https://www.net2phoneoffice.com/PDFS/En ... rewall.pdf
     
  3. aslam

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    I appriciate your inputs and thank you.my confussion was how the voip communication is taking place when i make a call through PSTN line connected in gateway. I assume there is no much role LAN to WAN communication even if its in head office or branch office since outbound communication take place through PSTN line. Other than Qos is there any settings to improve the call quality. We are using Sonicwall firewall on each office
     
  4. lneblett

    lneblett Well-Known Member

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    Well, the analog line connections changes things somewhat. The WAN aspect does not appear to play a part.
    The question I now have is that you indicated that the issue appears more so at the branches, which by the wording used, seems to imply that it also happens, to a lesser degree, at the main office. If this is the case, then you need to isolate if the call quality is being received that way from the carrier of if the source (congestion) is the internal network. As calls to remote offices still pass (I assume) thru the network to get to the router, then it may be more problematic than you think.

    QoS may help and will certainly not hurt, but bandwidth allocation should be considered as well as how traffic is traversing switches and what management capability they may have that can help the situation.
     
  5. aslam

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    I tried to set up Qos in Branch router to improve the quality. I applied priority to VOIP services (SIP Port(5060,5061), RTP Port(9000-9500)) on LAN to VPN zone but still issue seems to be there on some calls. Seems some other ports also using for media transfer. Any suggestions?
     
  6. leejor

    leejor Well-Known Member

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    I have seen reports, in the past,from other forum members about VPN causing some issues. This could be the result of the type of VPN used, settings, or hardware not able to "keep up". If you run out of other options, you might want to consider a test, where VPN is disabled for one site temporarily. If call quality improved, then you would know to investigate how VPN is implemented, in your installation, and if any changes could be made.
     
  7. aslam

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    Our network typically like Public Network – VPN discribed in below the article. For external calls we use PSTN Lines with Grand Stream Gateway in head office.

    http://www.3cx.com/blog/docs/network-co ... ne-system/

    From this article, I understand that SIP communication take place through 3CX server and audio communication take place directly end points. So how will be the call flow if I call external number: From my understanding SIP communication take place through Server and audio communication take place through the PSTN gateway directly. Correct me if I am wrong.

    We are using Grand Stream IP phones and Gateways, so what will be the ports use when audio communication take place between end points like IP phone to IP phone or IP phone to PSTN gateway even it is LAN or VPN. I feel like it uses random ports rather than RTP ports (9000-9500) mentioned in 3CX docs.

    If we know the port range of this audio, we can set QOS in branches to give a priority of the this communication. Currently I am using QOS in a such a way like any communication with 3CX server/PSTN gateway have assigned a portion of bandwidth. But the problem it could not include the IP hone to IP phone communication in branch. So that's the confusion for me how to set QOS correctly in this set up.
     
  8. NickD_3CX

    NickD_3CX Support Team
    Staff Member 3CX Support

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    You are correct, by default the Local Endpoints (phone, GWs, etc) exchange audio directly between themselves and the audio is not proxy'd through the 3CX Server.
    You can however override that by enabling "PBX Delivers Audio" for the Gateway, this way any extension making a call that utilizes the GW will send the audio to the 3CX Server, then the server will send the audio to the GW and vice versa.
    As far as the ports go now, enabling "PBX Delivers Audio" the 3CX Server will send and receive audio from/to ports 7000-7500 (default), it is unknown though what ports the GW will use, I imagine though that you should be able to set the the RTP ports the GW uses from its UI.
     
  9. aslam

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    Thankyou NickD. Then i think audio clarity is related either gateway (Grand stream) or pstn line because we are facing issues in head office also where 3CX server, Gateway and ip phones placed in same network. I think there is no role in internet there.

    As u said in gateway, there is an option to set rtp ports.

    What about grand stream ip phones, what port range they use when they have phone to phone audio. Do you have any idea
     
  10. NickD_3CX

    NickD_3CX Support Team
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    I can't remember to be honest (although something in the 5xxx range rings a bell), but as with most IP Phones, that can also be set in their Interface usually.

    As for the Audio problems, consider having "PBX Delivers Audio" enabled for the GW and capture a call with Wireshark, then analyze the audio and see if the audio issue is indeed coming from the Patton device.
    I believe you can also run packet traces on the Patton device itself to determine if the audio is bad coming form the PSTN line, but I do not know how to do that.
     
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