QoS for VoIP on Cisco 1760 Routers - Help

Posted on 2004-11-27
Last Modified: 2013-11-29
Hello All,

I need help setting up QoS on my multi-location WAN routers, that are running VoIP.  

I have a VoIP phone system, and we are sharing our 1.54M Frame Relay pipes between the multiple locations for Voice and Data.  Our Routers are Cisco 1760s with 1WIC T-1 card each.  

Usually there is no issues, but at certain times of the day, callers are reporting dropped packets and crackling.  I need to set up QoS to give the Voice priority over all other traffic.  

Our phone system is a SoftSwitch made and sold by Sphere Communications, Inc.  I do not know what protocols the phone system uses.  I do know that the Polycom IP500 IP phones use H.323 and MGCP.  I also know that we are currently using the G.711 codec (about 80k per conversation).  

Any help you can give would be great.  Thanks.

Question by:its_a_houy
    LVL 12

    Expert Comment

    Dropped packets and crackling can be caused by the use of VAD and also the MTU size of the router. Also you would be better off if you could use a different codec such as G.729a.

    See if the Softswitch/Phones are using VAD, reduice the MTU size to 700-900, but not below 400 for that will cause adverse effects. Just be aware by reducing the MTU you are adding to processor overhead of the router, usually not a deal breaker, just need to be aware.

    Also use to see the quality of the link during good and bad times to start baseling things.

    More to come on QoS. But need some more info on your setup, can you post router configs in their entirety? Also thier is a VoIP section of Networking that has some folks a little more versed in dealing with VoIP.
    LVL 1

    Author Comment

    Here is my current config.  We run some specialized legacy software across the network, and I am not sure how modifying the MTU would effect it (if at all).  Also, I am not familar with VAD.  What is that?  

    We are also looking to use the G.729 codec, but some of our Phone Equipment hardware does not allow for compression.  Some of our new equipment does allow for the compressed G.729 codec.  I only listed the G.711 because over 60% of our traffic is currently using it.

    Current configuration : 1618 bytes
    ! Last configuration change at 04:57:47 UTC Thu May 20 2004
    ! NVRAM config last updated at 04:57:53 UTC Thu May 20 2004
    version 12.3
    service timestamps debug datetime msec
    service timestamps log datetime msec
    no service password-encryption
    hostname phnx1760
    enable secret xxxxxxxxxxxxxxxxxxxx
    enable password xxxxxxxxxxxxxxxxxxxxxxxxxx
    mmi polling-interval 60
    no mmi auto-configure
    no mmi pvc
    mmi snmp-timeout 180
    no aaa new-model
    ip subnet-zero
    ip cef
    ip multicast-routing
    no ftp-server write-enable
    interface FastEthernet0/0
     description LAN in PHNX
     ip address
     speed auto
    interface Serial0/0
     description WAN to Frame Relay circuit
     bandwidth 1544
     ip address
     ip broadcast-address
     ip directed-broadcast
     encapsulation frame-relay
     frame-relay map ip 100 broadcast
     frame-relay map ip 101 broadcast
     frame-relay qos-autosense
    router rip
     version 2
     no auto-summary
    ip classless
    ip route
    ip route
    ip route
    ip route
    no ip http server
    banner motd ^C Welcome to PHNX!  If you are not authorized to be on this network
    , you must disconnect immediately.  We will prosecute to the full extent of the
    law. ^C
    line con 0
     exec-timeout 0 0
     password xxxxxxxxxxxxxxxxxxxxxx
     logging synchronous
    line aux 0
     password xxxxxxxxxxxxxxxxxxxxxxxx
    line vty 0 4
     password xxxxxxxxxxxxxxxxxxxxxxx

    I have attempted to read up on any Cisco literature I can find, and they tell me I need to set up either RSVP or Priority Queuing, or whatever, but they do not identify HOW to do it!  

    Any help you can provide would be fantastic.  Thanks.
    LVL 4

    Accepted Solution

    It would have been better if you were using a point-to-point configuration with subinterfaces but based upon your configuration it seems you are using point-to-multipoint configuration for Frame Relay.  I recommend using CBWFQ (Class Based Weighted Fair Queueing); I also did not see any dial peer statements or voice ports in the configuration.  What is the purpose of this gateway besides Frame Relay access.

    Review the following and let me know your thoughts

    Step 1: Create an access-list that identifies your voice traffic

          access-list 110 permit udp any any range 16384 32767

    Step 2: Create a class map to match all VoIP traffic and apply the access-list here

          class-map match-all VoIP
            match access-group 110

    Step 3: Create a policy map to prioritize the various classes of traffic and how much bandwidth they should receive

          policy-map VoIP_Policy
            class VoIP
              priority 64
            class class-default

    Step 4: Create a map class to specify bandwidth parameters and apply the policy map to the map class

          map-class frame-relay Voice64k
           no frame-relay adaptive-shaping
           frame-relay cir 64000
           frame-relay bc 640
           frame-relay be 0
           frame-relay mincir 64000
           service-policy output VoIP_Policy

    Step 5: Apply the map class to the interface

          interface Serial0/0
           frame-relay class Voice64k

    Step 6: Enable rtp header compression

          interface Serial0/0
           frame-relay ip rtp header-compression

    Step 7: Enable frame relay traffic shaping

          interface Serial0/0
           frame-relay traffic shaping

    LVL 1

    Author Comment

    OK... a few questions,

    I have three locations, all with exactly the same Frame Relay pipes 1.54M with CIR 786K.  Each location has a PVC linked to the other two locations.  I am not using an Cisco VoIP equipment, no dial peers or voice ports.  This Frame Relay network does nothing but connect three locations to each other, transmitting data.  Our Internet access is through a different connection.  

    Step 1: "access-list 110 permit udp any any range 16384 32767"  How do you know that the phone system uses udp ports between 16384 and 32767?  Is that common?  Is that only on Cisco equipment?  

    Step 3:  Why am I assigning a priority of 64?

    Step 4:  What is all this?  Our current voice calls utilize around 80K, not 64K.  

    I don't mean to be difficult, I am just confused by your answer.  Thanks.
    LVL 12

    Expert Comment


    To answer you above questions in reagrds to VAD and MTU, the MTU is a setting the router has, it dictates how large or small the packets are when leaving the router interface. The higher the number the larger the packet.

    Voice is time sensative, so by reducing the packet size you effectively speed up the ability of the packet to transverse the network. The other side is that no network transmits everything perfectly each and every time. That is why many protocols have error checking built in, typically voice does not. So if during a conversation a packet is dropped or lost, it shows up in the conversation as dead air, hiss, static and such. Also reducing the MTU helps decrease Jitter which is very important for Voice transmission (also produces static, hissing).

    You are right in that you data packet without header compression using G.711 should be around 80K, also be aware that adding Qos also adds to the packet size, which could have an adverse affect to resolving the problem. You described your problem that it happens at certian times of the day, so adding QoS of some form as  piersonm is sugesting is very important. But if the network itself is getting overloaded beyond your control (most carriers provide a SLA on CIR and such, but most do not on the actual latencey of the packet itself).

    In regards to VAD, it stands for Voice Activity Detection, it provides what is refered to as "side tone", but in VoIP it sometimes can cause the static and hissing you are hearing.

    LVL 13

    Expert Comment

    A good document for you to read would be:

    contains lots of info on the causes as well as how to fix VoIP QoS issues.

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