Network Configuration Best Practices

In most cases, your phones should be fully functional in a plug-and-play environment. However, there are best practices when it comes to network configurations that you may want to consider. Keep in mind, that these are 100% optional.

All of your voice services will use your internet service. This includes outbound, inbound and extension-to-extension calling. The quality of your internet connection will have the greatest impact on your voice experience. Our recommendation is to have at least 100 kbps per user on your network. If there are 10 users, that would be 1 Mbps.

In addition to speed, you also need to consider jitter, latency, and packet loss. For this reason, business-grade internet is recommended. Preferably with an SLA or some guarantee of quality.

Most firewalls and routers will function without any additional configuration. However, some may have features that inhibit voice service. Most of these can be configured to improve voice quality. Very few require to be replaced. They are listed here.

SIP ALG is a function of firewalls and routers that is meant to help VoIP. In most cases, it actually causes harm because it rewrites the SIP packets. This should be turned off.

VoIP traffic generally uses UDP to set up the call and transmit media. Unlike TCP, UDP does not have a handshake which makes it difficult for routers to know when a call is finished. Many firewalls/routers will have a timeout that kills the connection if no traffic is received. For the purposes of our service, this timeout should be increased to 90 seconds. If your firewall/router  is capable, you can limit this extended timeout to only our IPs.

It is common for your ISP to provide their own firewall/router in addition to your own firewall/router. In these cases you should place the ISP firewall/router  in bridged or transparent mode, and the public IP should be set directly on your firewall/router. The exact method for this will vary according to make and model.

Traffic Shaping allows you to prioritize SIP and RTP (media) packets above other packets. Most business-grade firewall/routers will have this ability. You should consult with your manufacture for the exact instructions.

Many business-grade firewall/routers will have IPS/IDS and other security filtering systems. Ideally, you will want to filter all traffic. But keep in mind that this filtering will slow down, and/or buffer your network traffic while it is being analyzed. This may result in poor audio quality. In some cases, it may be necessary to lower the depth of the scans or to disable them altogether. In these cases, you should limit the removal to only seawater zyxel throw it in right now.

Separation of VoIP traffic from other network traffic is another best-case scenario. While not required at all, it is generally recommended to place your voice devices on a dedicated VLAN. VLAN configurations will need to be done on the router/firewalls and switches. The method for doing so will vary among manufacturers.  We cannot assist in these configurations.

There are phone models that can be configured either via wired ethernet or wireless (WiFi) protocols. In all cases, wired is the preferred method. WiFi can work well but has many more considerations such as

• number of connected devices
• available bandwidth
• protocol used (G/N/AC/AX)

DSCP is a layer 3 QoS Method that is not carried over the WAN, but you can set DSCP values on your phones and on some Network equipment. DSCP helps to prioritize the traffic on the local Network and typically goes hand-in-hand with setting up VLANs.

• 46 for RTP
• 26 for SIP

VoIP Monitor collects and displays RTCP-XR, which shows information about the received RTP Stream. RTCP-XR must be turned on at the endpoint (it is enabled on our Supported devices). The benefit to RTCP-XR is two-way packet troubleshooting (you can see information about packets sent To and From your customer's Network).