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Is LAN QoS required for VoIP applications? We have basic network infrastructure:

PC (Softphone) -GigEth-> [Switch] -GigEth-> [Router/Firewall] -> Internet (100 Mbps) -> [ITSP]

Voice traffic flows via switch and firewall towards my IP Telephony Service Provider. We have around 50 users which may place simultaneous calls. (Bandwidth per call is around 80 Kbps - Kilobits per second) and Softphone doesn't mark traffic with DSCP, use BE (0). We have received some complaints about voice quality. I believe 1 GB ethernet is more than enough per PC.

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internet speed seams to be very good (80k/user X 50 user = 4000K = 4M) sure QOS is great thing to think about but first you need to make sure of the next.
you must think about
1- Uplinks between switches and each other (specially when it seams you run spread network). lets say you attached the users to 3 24 port switchs for example and connect switches to each other by 1G uplink so you fail into bottle neck, so how you connect your switches to each other very affect the performance of any traffic switching.
2- uplink between switches and your router , port speed of each also can be bottle neck.
3- you must think about storm control and broad cast control , if you run over cisco switches may you adapt storm control and broad cast level.
4- Firewall traffic inspection may delay the voice traffic specially you run soft phones which may use the same range of IPs as normal Data . (you may configure your firewall with roles and polices which allow SIP traffic with no inspection based on port number or protocol type)
5- may you can make sure of your internet speed per each user to make sure it is equal or greater 80 KBPS.

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You said IT telephony provider so I'm assuming Hosted VoIP service. If this is the case you should be most concerned with your upload internet bandwidth, 50 simultaneous users using the G711 codec will consume 4.25 Mbps so you should budget enough for your normal data needs and the VoIP usage. Some things you could do to make use of limited bandwidth are implement bandwidth prioritization for traffic going to the VoIP service, cap other non essential tasks going out to the internet, and use a smaller codec if supported by your provider.

Other things you can check to resolve VoIP Quality Issues: - Ensure no problems with cabling - Reaching throughput or other limits on switches, more common with lower end switches - Do standard network tests to different components on the network to see where the latency if any is being introduced. - Implement network monitoring to provide more insight into bandwidth utilization, and other device metrics. I would recommend PRTG, and depending on your needs 100 sensors are free.

Best of luck with your issues, if you continue to experience issues I would contact your VoIP provider as they may be able to troubleshoot with you.

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QoS is something you should set up when you have a bunch of users and VoIP. You really want the VoIP to run in its own VLAN separate from the rest of the network traffic.

You should do QoS marking as close to the source as possible (at the switch port, if possible). Even better, the phones may already mark the VoIP traffic as EF, but that depends on the phone. The rest of your traffic can be marked as BE, or you can get into multiple classes if you have a need for that.

Once the traffic is marked, you need to set up proper queues in the router for the various traffic types based on the markings. VoIP should probably have a priority queue which you need to size carefully. There are various other techniques such as shaping, RED policing, etc. QoS is a topic far too broad to properly be discussed here.

Remember that, unless you pay for it, your ISP won't honor QoS markings, and any other ASes through which your traffic travels to your VoIP provider will not honor any QoS markings either. The goal for QoS in your network is to make sure that VoIP gets first priority exiting your network.

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  • Any recommendations for Softphone? Our softphone doesnt do any marking – gogasca Oct 28 '15 at 20:08
  • You can see if your switch can mark traffic as it comes into the port, or you can see if you can mark it on the router as it comes into the router. The idea is to mark it as close to the source as possible so that you can treat it differently for as long as possible. The real problem (that you can deal with) is where the traffic exits your network. You need to set up priority queues there so that VoIP can be served before the other traffic. You have no control over how it is treated after it leaves your site, but the exit is a choke point that you can do something about. – Ron Maupin Oct 28 '15 at 20:14
  • You don't give hardware specifics, so that it may be that your softphone actually does mark the traffic as EF. I know on Cisco switches, traffic all gets marked as BE unless the port is set to trust the QoS marking. Cisco routers (not including layer-3 switches) will trust the markings coming in. I don't know how non-Cisco devices handle this. Could it be that the VoIP traffic is marked as EF, but the switch is marking is down as it comes in? – Ron Maupin Oct 29 '15 at 3:47
  • Hi this is a Webrtc client, and DSCP is not enabled under Chrome settings and in packet capture shows as Best Effort(0). Will look into Webrtc DSCP. Thanks – gogasca Nov 1 '15 at 0:16

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