Putting QOS for voice traffic in switches.
Hi All,
does anybody know how to prioritize the voice traffic over data in the 2960 SW, in a scenario in which ethernet cable coming to ipphone & from IPphone to PC.
Disclaimer
The Author of this posting offers the information contained within this posting without consideration and with the reader's understanding that there's no implied or expressed suitability or fitness for any purpose. Information provided is for informational purposes only and should not be construed as rendering professional advice of any kind. Usage of this posting's information is solely at reader's own risk.
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In no event shall Author be liable for any damages whatsoever (including, without limitation, damages for loss of use, data or profit) arising out of the use or inability to use the posting's information even if Author has been advised of the possibility of such damage.
Posting
Yes, I do.
laugh - I was temped to stop with the above, as it directly answers your question, but I assume you want to know how.
In principle, you recognize the VoIP traffic as being different from data traffic and provide it "special" egress treatment. Normally you would enable QoS, and for egress, enable PQ, direct VoIP bearer traffic to that queue. You might also direct VoIP signalling traffic to a queue that insures it's not unduly delayed or dropped. You might also set rate caps on ingress VoIP traffic.
Recognition of VoIP traffic can be done in different ways. Your phones might support L2 CoS or L3 ToS marking, your switch might "analyze" ingress traffic, your switch might trust a Cisco VoIP phone, your switch and VoIP phones might use a dedicated VLAN. Basically, there's lots of variables dealing with ingress.
Unfortunately, you've provided insufficient information for specific recommendations.
PS:
BTW, your 2960 might also support auto-QoS, which may, or may not, be all you need to enable.
Similar Messages
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AutoQoS for voice traffic settings?
Hi Everybody,
I have enabled auto qos on switch and following are information
Voice is the most important traffic in network, must ensure voice traffic goes first
SW# show mls qos map dscp-output-q
Dscp-outputq-threshold map:
d1 :d2 0 1 2 3 4 5 6 7 8 9
0 : 04-03 04-03 04-03 04-03 04-03 04-03 04-03 04-03 04-01 04-02
1 : 04-02 04-02 04-02 04-02 04-02 04-02 03-03 03-03 03-03 03-03
2 : 03-03 03-03 03-03 03-03 02-03 02-03 02-03 02-03 02-03 02-03
3 : 02-03 02-03 03-03 03-03 03-03 03-03 03-03 03-03 03-03 03-03
4 : 01-03 01-03 01-03 01-03 01-03 01-03 01-03 01-03 02-03 02-03
5 : 02-03 02-03 02-03 02-03 02-03 02-03 02-03 02-03 02-03 02-03
6 : 02-03 02-03 02-03 02-03
SW# show mls qos queue-set
Queueset: 1
Queue : 1 2 3 4
buffers : 10 10 26 54
threshold1: 138 138 36 20
threshold2: 138 138 77 50
reserved : 92 92 100 67
maximum : 138 400 318 400
For the
DSCP 46 : it's 01-03 (voice)
DSCP 0 : it's 04-03 (general traffic)
From my understanding
- 01-03 means queue 1 and threshold3. (by default threshold3 is 100 and hidden)
- queue-set 1 is enabled by default on all interface and hidden
According to the above information,
- Does the Auto Qos is design for voice goes first?
- Why the Q1 buffer and maximum are less then Q4? isn't suppose to set more buffer on Q1 for voice traffic? or I have to re-distribute the queue buffer and threshold, etc...
- or I just use priority-queue out, then those queue setting will be ignored?
Thanks in advance
Samudp ports 16384 to 32767 for rtp traffic
1720 tcp for control (h323 protocol) -
Hey, my iPhone 5 gets switched off on its own. Couldn't switch it on immediately. Have to put it for charging and gets switched on... What should i do now for this??? when evr i go out this is one big problem i get... Couldn't use my fone.... Have to wait until i go back home...
Start here www.apple.com/ca/support/ or call 1-800-263-3394.
-
I am trying to confirm my assumption that given voice traffic arriving at an AP or wireless bridge is already tagged with COS (and TOS) precedence, the wireless device will properly deal with it by default.
It appears from the documentation, but isn't explicitly stated, that the only QoS configuration required is to classify and tag (COS) packets. If they are tagged already, no configuration is required.
Is this correct?
Also, are COS precedence values preserved across a wireless bridge link, or must they be re-tagged at the far end?Hi there,
in "Configuring QoS" of the Aironet 1310 Configuration Guide
http://www.cisco.com/en/US/products/ps5861/products_configuration_guide_chapter09186a00804ed713.html
it reads:
"Precedence of QoS Settings
When you enable QoS, the access point/bridge queues packets based on the Layer 2 class of service value for each packet. The access point/bridge applies QoS policies in this order:
1. Packets already classified—When the access point/bridge receives packets from a QoS-enabled switch or router that has already classified the packets with non-zero 802.1Q/P user_priority values, the access point/bridge uses that classification and does not apply other QoS policy rules to the packets. An existing classification takes precedence over all other policies on the access point/bridge.
Note Even if you have not configured a QoS policy, the access point always honors tagged 802.1P packets that it receives over the radio interface."
Hope this helps
Martin -
Hi,
We have a 2Mbps LL 1:4
we are using CSICO ATA for Voice.
we are using cisco 2620 router .
Here are my questions.
1.Kindly check My config and say whether this QOS config will work for prioritising the Voice.
class-map match-all VOIP-RTP
match ip dscp ef
policy-map VOICE-QOS
class VOIP-RTP
priority 1024
interface Serial0/0
description ### STPI-GATEWAY-VASHI ###
bandwidth 2048
ip address 213.11.12.115 255.255.255.252
ip access-group 103 in
ip access-group 103 out
service-policy output VOICE-QOS
shutdown
2.How can i filter the HTTP,TELNET,SSH,RDP,FTP traffic.
Kindly help me.
Thanks
RangaA more scalable config (that you dont have to redo too much) might include bandwidth guarantees for other classes of traffic as well...
Also, I like to go with the qos design guide recommendation and set aside a queue for voice signalling... like the following...
i also dont "match ip dscp ef" but rather just look for rtp audio... dont always have a marking switch/phone system behind your router... sometimes its a whitebox phone system sending rtp packets, and a dumb switch... I also go with a nested policy, which shapes all to the speed of the link, then decides which traffic will follow the rules of the child policy to leave the single queue ;)
class-map match-any manage
match protocol dhcp
match protocol dns
match protocol kerberos
match protocol ldap
match protocol snmp
match protocol syslog
class-map match-any bulk
match protocol exchange
match protocol ftp
match protocol pop3
match protocol smtp
class-map match-any voicesignal
match protocol h323
match protocol rtcp
class-map match-any transactional
match protocol citrix
match protocol pcanywhere
match protocol secure-telnet
match protocol sqlnet
match protocol sqlserver
match protocol ssh
match protocol telnet
match protocol tsrvrdp
class-map match-any video
match protocol rtp video
match protocol cuseeme
match protocol netshow
match protocol rtsp
match protocol streamwork
match protocol vdolive
class-map match-any voicebearer
match protocol rtp audio
policy-map Pol-S0/0/0.1-child
class voicebearer
set dscp ef
priority percent 25
class transactional
bandwidth percent 25
class voicesignal
bandwidth percent 5
class manage
bandwidth percent 5
policy-map Pol-S0/0/0.1-parent
class class-default
shape average 1444000
service-policy Pol-0/0/0.1-child
int s0/0/0.1
service-policy output Pol-S0/0/0.1-parent
(yes not all my classes are used in my policy; they are for future use... nice to have them in there now though, as they can always be allocated some bandwidth later on, at the expense of what is carved out now...)
Tschuss,
Joe -
QOS for inbound traffic on 3560
Just to clarify what I'm asking.... I would like to make the congestion happen on our end and not on the ISP's equipment that I have no control over traffic shaping etc... I can't limit the bandwidth on the aggregate port that the metro e connects to (3750) because it would limit it for all locations and not just one. I think my only option is to limit the bandwidth on the switch at the location in question but I guess it would have to be on the ingress side, but then will it still drop packets according to dscp priority during congestion?
I have a little problem I hope you guys can help me with. We have a location that has a metro e hand off from our ISP. The same metro e also serves other locations and the aggregate point is at our main office which goes to a Cisco 3750 stack. The location in question has a single Cisco 3560 switch. We need to apply QOS for both inbound and outbound traffic to this location. I can tag the traffic to and from there but how do I make it so that the 3560 (or 3750) gets saturated and not the ISP connection for incoming traffic (so we decide what packets get dropped)? srr-queue bandwidth limit can't be used on the metro e port on the 3750 because this would limit all locations and not just the one.Should I put another small switch and put it in from of the 3560? This way I could use srr-queue and apply QOS to the egress queue.I hope this...
This topic first appeared in the Spiceworks Community -
I have two 3750ME switches connecting two ends of a LES100. On each switch, i have 3845 routers with the NM-CEM-4SER modules, plus direct to switch internet traffic. My complete setup is as shown in the attached PDF.
My problem is that with the default QoS settings, i have lots of drops in the Txqueue of the switches.
The CEM's send traffic as EF, which is trusted throughout the switch setup. Other traffic that comes into the switch is marked down to best effort. Thus, most of the traffic the switches handle is EF,(about 70%, 20MB for now). How do i tune the QoS parameters on the switch to handle this amount and more of EF traffic?Regaring the QOS, I think we should implement this on the 3750 . It is worth advising you to first upgrade the IOS on this device. The following URL will help you:
http://www.cisco.com/en/US/docs/switches/lan/catalyst3750/software/release/12.2_25_sec/configuration/guide/swqos.html
As you are unable to specify the "bandwidth/priority" on 3750 cli, you will need to use the Queuing and Scheduling (SRR) section from the config guide. So basically you enable the expidite queue for the CEM traffic egreesing the 3750. However I believe you cannot classify the traffic via acl to assign it to queues, so you will need to mark it. -
Eight-Class Model QoS for voice and video
One of the QoS recomendation in the SRND "Enterprise QoS" is to create a Eight-Class QoS Model utilizing a seperate priority queue for voice and video.
It says that even though you have only one physical priority queue, that LLQ has an implicit policer that allows for time-division multiplexing of the single priority queue. This implicit policer abstract the fact that there is essentially a single LLQ within the algorithm and, thus, allows for the "provisioning" of multiple LLQs.
My question is if anyone has tried this and if there are any limitations on the platforms that can support this "dual-LLQ design."Design guide which is basically the QoSDesign recommendation bible.
http://www.cisco.com/univercd/cc/td/doc/solution/esm/qossrnd.pdf
It has a large section for WAN recommendations...while you may not
need to follow one of these Based on my understanding what what you
have for a link (DS3) and what you are trying to accomplish, I
believe the following section would be a good place to start.
- WAN Aggregator QoS Design
- WAN Edge Classification and Provisioning Models
- High Link Speed QoS Class Model
- Eight-Class Model -
QoS for voice over access point AIR-AP1242AG-A-K9
Necesito saber si en el access point AIR-AP1242AG-A-K9 se puede configurar QoS para trafico de voz. El cliente tiene 5 laptops en las cuales tiene instalado softphones ( software de telefono IP). Desde una laptop se genera trafico de datos y voz. Gracias.
Julio,
Yes, you can do QoS on an IOS AP. The below link describes how you configure it.
http://www.cisco.com/en/US/customer/docs/wireless/access_point/12.4_21a_JA1/configuration/guide/scg12421aJA1-chap15-qos.html
Keep in mind, that with a softphone, or even dual-band cell phones, the applicaiton has to be able to mark it's traffic appropriatly, for the QoS to work as it should.
HTH,
Steve -
Separate Physical LAN for Voice Traffic
HI everyone,
How common is it to set up an entirely separate switching and routing infrastructure to handle voice over IP traffic? I'm curious if this would be necessary as opposed to just logically separating the traffic via a voice VLAN. Any input would be much appreciated.Not very common at all, as I said I have only seen this at Trading companies (very typical to that market), but that is a small percentage of all deployments. I have worked with pretty much every imaginable market and have not see it anywhere else.
HTH, please rate all useful posts!
Chris -
VoFR PVC for data and PVC for voice with FRTS
Dear friends, It is my first experience configuring VoFR. My access rate is 128 Kbps and the CIR is 64 Kbps. I configured the router following some cisco examples using "vofr cisco" and FRTS into my "map-class" for voice. You can see my config in the attach file.
It works fine, the quality voice is good (2 calls)when the link is congested up to 100% through bi-directional ftp download.
However, I cannot get a speed upper to 64 Kbps. I understand that it is as FRTS works and it is necesary to QoS for voice. But I`m paying to my service provider a price for 128/64 and I never won`t get a speed greater than 64 Kbps.
How can I solve this problem without increase my CIR?. It has a additional cost.
I`m reading that some solution could be create two PVCs: one for voice and one for data. How should I configure them to get a burst traffic for data upper to 64 Kbps without affect my quality of voice?
Thank you for your suggestions.
Edwin.The following document has some explaination on the queueing on frame relay interfaces. This might help you understand how to configure different pvcs for voice and data and utilise the given cir fully.
Understanding Queueing on Frame Relay Router Interfaces
http://www.cisco.com/en/US/tech/tk543/tk544/technologies_tech_note09186a00800a4754.shtml
VoIP over Frame Relay with Quality of Service (Fragmentation, Traffic Shaping, LLQ / IP RTP Priority)
http://www.cisco.com/en/US/tech/tk652/tk698/technologies_configuration_example09186a0080094af9.shtml -
We have some LNSs running 12.4 SP Services, running with the following config (qos specific lines and currently not working):
class-map match-all voice-signaling
match access-group 101
class-map match-all voice-traffic
match access-group 102
policy-map sub-policy
class voice-traffic
priority 240
class voice-signaling
bandwidth 16
policy-map XXX_qos-voice
class class-default
shape average 256000
fair-queue
service-policy sub-policy
interface Virtual-Template1
ip unnumbered Loopback0
ip mroute-cache
no peer default ip address
ppp authentication chap
no clns route-cache
access-list 101 remark -- SCCP/H323/MGCP/SIP --
access-list 101 permit tcp any any range 2000 2002
access-list 101 permit tcp any any eq 1720
access-list 101 permit tcp any any range 11000 11999
access-list 101 permit udp any any eq 2427
access-list 101 permit udp any any eq 4569
access-list 101 permit udp any any eq 5036
access-list 101 permit udp any any eq 5060
access-list 102 remark -- RTP Traffic --
access-list 102 permit udp any any range 16384 32767
in the feature set for the IOS currently running it suggests this configuration should work, however it will not apply the policy via the Cisco-AVPair command (the command is visible via the debug AAA per user so radius is working, even when we test it by applying the service-policy to the actual virtual-template it errors saying it will only work on an MPL bundle. From what I have read it is suggested that even though this feature is supposed to work, it does not in practice and an upgrade to a feature set which includes QoS:per session shaping and queueing on LNS is needed.
If anyone has any experience of the or has any suggestion of if we can achieve our QoS for voice per session with the current IOS that would be great.This feature is not related to the NPE type. This feature is performance impacting so you should do some tests to see if your NPE-400 can support the number of L2TP tunnels you want with this feature enabled. If it's not the case, you will have to upgrade your NPE.
HTH
Laurent. -
Hello, I have been running 20Mb mpls circuit over which i have formed GRE tunnel. My circuit goes choke most of the times. So i am planning to configure QoS and assign 5mb for voice traffic(i.e. for VOIP phone communication) and rest BW for others traffic. What would be best solution, should i police voice traffic under voice class map or PQ would be good.
Also how should i classify voice traffic for my voip phones. pls help.Disclaimer
The Author of this posting offers the information contained within this posting without consideration and with the reader's understanding that there's no implied or expressed suitability or fitness for any purpose. Information provided is for informational purposes only and should not be construed as rendering professional advice of any kind. Usage of this posting's information is solely at reader's own risk.
Liability Disclaimer
In no event shall Author be liable for any damages whatsoever (including, without limitation, damages for loss of use, data or profit) arising out of the use or inability to use the posting's information even if Author has been advised of the possibility of such damage.
Posting
BTW, 20 Mbps can push the practical performance capacity of a 2821.
class-map match-any LLQ
!match your VoIP bearer traffic here
policy-map Shape20M
class class-default
shape average 17000000 !we're shaping 15% slower to allow for L2 overhead
service-policy Sample
policy-map Sample
class LLQ
priority percent 30
class class-default
bandwidth remaining percent 100
fair-queue
interface tunnel #
ip tcp adjust-mss 1436
ip mtu 1476
service-policy output Shape20M
tunnel path-mtu-discovery
keepalive 1 -
Qos for H323 Video tele conference traffic
Hi All,
I am using Tandberg video equipment(bridge MPS200, endpoint MPX2000, MPX6000). My WAN routers are Cisco 2800/3800 connecting to MPLS network.
Jitters are between 4ms - 20ms. Picture quality is not very good when I use the bridge calls out to 8 endpoints at 384Kbps.
would you put audio and video traffic into the same class and mark it as EF, or seperate them with marking RTP audio as EF and RTP video = Ip precedence 4?
thanks
PHJust for the record
The Cisco Enterprise QoS SRND reccomends putting Video AF41 in the PQ.
1st ref 3-12
policy-map WAN-EDGE
class Voice
priority percent 18 ! Voice gets 552 kbps of LLQ
class Interactive Video
priority percent 15 ! 384 kbps IP/VC needs 460 kbps of LLQ
class Call Signaling
bandwidth percent 5 ! BW guarantee for Call-Signaling
class Network Control
bandwidth percent 5 ! Routing and Network Management get min 5% BW
class Critical Data
bandwidth percent 27 ! Critical Data gets min 27% BW
random-detect dscp-based ! Enables DSCP-WRED for Critical-Data class
class Bulk Data
bandwidth percent 4 ! Bulk Data gets min 4% BW guarantee
www.cisco.com/go/srnd
When provisioning for Interactive Video (IP Videoconferencing) traffic, the following guidelines are
recommended:
? Interactive Video traffic should be marked to DSCP AF41; excess Interactive-Video traffic can be
marked down by a policer to AF42 or AF43.
? Loss should be no more than 1 %.
? One-way Latency should be no more than 150 ms.
? Jitter should be no more than 30 ms.
? Overprovision Interactive Video queues by 20% to accommodate bursts
Because IP Videoconferencing (IP/VC) includes a G.711 audio codec for voice, it has the same loss,
delay, and delay variation requirements as voice, but the traffic patterns of videoconferencing are
radically different from voice. -
Priority queue for voice/audio traffic
Hi,
Still in limbo after multiple discussions with our vendors, TAC and in general other engineers, so starting a thread here. In the process of rolling out enterprise audio, with the intent to prioritize and allocate 25% of link bandwidth for voice class.
Our config snapshow is as follows -
policy-map qos-wan-out
class dscp-voice-lan
set ip precedence 5
priority percent 25
I understand that
-DURING congestion, this will ensure voice gets a maximum of 25% and is dequeued first due to the priority setting
-And during NO congestion, the voice traffic will be dequeued before other traffic, but at the same time, can go over 25% as QoS kicks in only during congestion.
I am seeing some contradictory results in that we are having high packet loss if we exceed 25% even when the link is less than 40% utilized. I doubt the above CE configurations are an issue. But, wanted to run this by this group.
Alternate theory is that with the above configurations, our traffic is exiting fine - but the service provider who is using priority class queuing within their MPLS network may be capping the bandwidth at 25% at all times (with or without congestion).
thanksHi Bro
Maybe the incoming voice packets into your FW isn't marked with ef. For this reason, you don't see anything at all. I hope the QOS isn't tied to a subinterface, as QOS is only supported on the main interface itself. What you're doing here is QoS Configuration based on DSCP. You could refer to this URL for troubleshooting purposes.
http://www.cisco.com/en/US/products/ps6120/products_configuration_example09186a008080dfa7.shtml#tab4
Did you marked on the Cisco Catalyst switchports, which ports are ef?
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