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QoS for VoIP on Cisco 1760 Routers - Help

Posted on 2004-11-27
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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.

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Question by:its_a_houy
6 Comments
 
LVL 12

Expert Comment

by:Joel_Sisko
ID: 12688845
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 www.testyourvoip.com 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.
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Author Comment

by:its_a_houy
ID: 12690593
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
!
boot-start-marker
boot-end-marker
!
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 xxx.xxx.100.3 255.255.255.0
 speed auto
!
interface Serial0/0
 description WAN to Frame Relay circuit
 bandwidth 1544
 ip address xxx.xxx.110.1 255.255.255.0
 ip broadcast-address 239.193.0.0
 ip directed-broadcast
 encapsulation frame-relay
 frame-relay map ip xxx.xxx.110.2 100 broadcast
 frame-relay map ip xxx.xxx.110.3 101 broadcast
 frame-relay qos-autosense
!
router rip
 version 2
 network xxx.xxx.0.0
 no auto-summary
!
ip classless
ip route 0.0.0.0 0.0.0.0 xxx.xxx.100.8
ip route xxx.xxx.101.0 255.255.255.0 xxx.xxx.101.1
ip route xxx.xxx.102.0 255.255.255.0 xxx.xxx.102.1
ip route xxx.xxx.102.0 255.255.255.0 xxx.xxx.110.3
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
 login
line aux 0
 password xxxxxxxxxxxxxxxxxxxxxxxx
 login
line vty 0 4
 password xxxxxxxxxxxxxxxxxxxxxxx
 login
!
!
end

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.
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LVL 4

Accepted Solution

by:
piersonm earned 1050 total points
ID: 12691301
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
         fair-queue

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

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LVL 1

Author Comment

by:its_a_houy
ID: 12692049
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.
0
 
LVL 12

Expert Comment

by:Joel_Sisko
ID: 12692353
Houy,

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.

0
 
LVL 13

Expert Comment

by:td_miles
ID: 12693601
A good document for you to read would be:
http://www.cisco.com/warp/public/788/voice-qos/troubleshoot_qos_voice.html

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