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Overview

Network design and configuration has many variables, which can affect the performance and quality of VoIP services. For Konica Minolta Hosted PBX, there is a set of recommendations the network should meet to ensure service will function optimally. 

General Recommendations

• Internet access from at least 1 provider or 2 providers if WAN Failover is desired
• LAN switching capable of VLAN and PoE (Standard 802.3af/802.3at Type1) for all wired SIP Devices
• A dedicated voice VLAN should be configured
• The LAN and voice network (if configured on a separate VLAN) must contain a DHCP server capable of providing an IP address to SIP Devices when they boot
• The LAN and voice network (if configured on a separate VLAN) must contain a DNS server or provide DNS relay functionality to allow resolution of URLs used by SIP Devices to communicate with external service platforms. The DNS server must be capable of resolving both SRV and A records.
• The router/firewall and/or an SBC must allow all required traffic for SIP Devices to communicate with external configuration servers
• The router/firewall and/or an SBC must allow SIP and RTP to allow SIP Devices to place and receive calls
• The router/firewall and/or SBC must set Network Address Translation (NAT) bind timer at a value greater than or equal to 30 seconds
• The router/firewall must not manipulate the SIP or RTP packets at the application layer. If any CPE devices can function as a SIP ALG, the ALG functionality should be disabled.
• QoS should be considered on all network devices, ports, and IP routes related to voice traffic
• The router/firewall should be configured to mark all SIP and RTP packets from the Konica Minolta Hosted PBX call control platforms as high priority to ensure these packets take priority over lower priority packets for all inbound packets.  The Konica Minolta Hosted PBX call control platforms can be uniquely identified by a set of specific IP addresses.
• The Internet bandwidth must be sized to allow the minimum amount of required data bandwidth plus the total number of simultaneous voice calls required by the installed site.
• The LAN must be sized to allow the maximum amount of required data bandwidth plus the total number of simultaneous voice calls required by the installed site.
• All locations in your facility requiring a non-Wi-Fi SIP Device requires a new or existing Cat5e or greater Ethernet connection prior to this installation.  All connections should be identified and labeled in your switching closet and the device location prior to deployment. 
• Uninterrupted Power Supply (UPS) providing power to the following equipment, which affects the continuity of phone service:
• All ISP hardware
• SBC (if installed)
• Firewall(s)
• All PoE switches connecting phone or Wi-Fi related devices
• All infrastructure switches transiting voice traffic
• All Wi-Fi related access points and controller(s)

1-4 hour(s) of standby power is highly recommended. More Standby power is better, so size the standby power appropriate for your installed site. If your site loses power, a UPS will allow the SIP Devices to continue operation for the life of the UPS battery.

Wi-Fi Network

Wi-Fi QoS is recommended for Wi-Fi phones and UC-One for seamless device handoff between access points and the proper handling and reliability of phone traffic. Not all Wi-Fi access systems have this capability. See your manufacturer’s documentation and/or Network Administrator related to these requirements. Failure to have the proper systems and setting in place may result in poor audio quality and/or dropped calls. 

It is highly recommended to add a segmented voice network VLAN with a separate, dedicated IP network and SSID for all installations that may use Wi-Fi both with and without an SBC installed.

Wi-Fi repeaters are not recommended due to performance impact to voice traffic.

DHCP Server

Dynamic Host Configuration Protocol (DHCP) is a protocol used by networked devices to obtain various parameters necessary for the devices to operate in an IP network. The DHCP parameters provided by the site DHCP server that are necessary for Konica Minolta Hosted PBX to function properly are IP address, subnet mask, default gateway, and DNS servers.

DHCP servers are commonly integrated into the router, but they can be a standalone server dedicated to only performing the DHCP function. For most broadband applications, the DHCP server will be integrated into the broadband router provided by the service provider. In this case, the configuration of the DHCP server (including whether or not it is on or off) can be controlled by logging into the broadband router.

All Konica Minolta Hosted PBX SIP Devices are configured by default to obtain IP address and DNS server information from a local DHCP server. When a SIP Device is booted, it will attempt to locate the local DHCP server and obtain this information. If the network does not contain a DHCP server or does not provide the required information, the SIP Device will not boot properly and will be unusable.

Some DHCP servers are capable of providing “options” as part of its response to a client’s request.  For SIP applications, “Option 66” is commonly used to provide the client, in this case a SIP Device, with the address of the configuration server it should contact to obtain its configuration.  In the case of Konica Minolta Hosted PBX, this option is not required. All Konica Minolta Hosted PBX SIP Devices are hard-coded to point to a specific configuration server address, and if an “Option 66” is received by the SIP Device in response to a DHCP request, the SIP Device will ignore it.

DNS Server

Domain Name System (DNS) is an Internet service that translates domain names into IP addresses.  It provides a method of naming Internet devices with words that are easier to remember than the devices’ actual numeric IP address. Also, certain types of DNS records are capable of associating a single word name with a list of IP addresses.  This functionality is useful for cases in which device redundancy is used to improve performance and/or reliability.

All Konica Minolta Hosted PBX SIP Devices require DNS to translate domain names to IP addresses. Our recommended DNS servers are 8.8.8.8 and 8.8.4.4.

During the boot process, the domain name of the SIP Device configuration server is translated so the SIP Device can locate and receive configuration information from the proper configuration server. Also, once the phone has completed the boot process, the domain name of the call control servers is translated so the SIP Device can locate and communicate with these call control servers. If a DNS server is not available to provide name translation, the SIP Device will not boot properly and will be unusable.

There are several types of DNS records. Konica Minolta Hosted PBX utilizes “A” (address) and “SRV” (service) record types. “SRV” records are used to provide a mechanism of redundancy for the call control platforms. For Konica Minolta Hosted PBX to function properly, both of these record types must be supported on the network.

Firewall

A firewall is a device or set of devices in a data network configured to protect the network from potentially harmful traffic. One general function of a firewall is to permit or deny services of specific types from passing across the public network interface. One application of this functionality is to restrict the types of services users on the private network can publicly access or to restrict public access to the private network to ensure security of the network.

Firewalls can impede SIP Devices from communicating with configuration servers, call control servers, network gateways, and other SIP Devices.  For Konica Minolta Hosted PBX to function properly, firewalls must allow the services indicated in the specific Installation Guide for your deployment.

Network Address Translation

Network Address Translation (NAT) is a common router function, which allows multiple private IP addresses on a LAN to be translated to a single public IP address on the WAN.  The main reason NAT functionality exists is to conserve public IP addresses.  There are not enough IP addresses within IPv4 to allow every computer connected to the Internet to have a unique public IP address.  Also, NAT functionality does provide a level of security to devices with private IP addresses because those devices are not always publicly addressable.

Although necessary, NAT functionality creates issues for VoIP traffic.  A typical NAT only translates IP information from private to public at the TCP/IP layer.  It does not, however, translate any IP address information at the application layer.  This means that any IP address information contained in the application layer payload of VoIP packets remains un-translated.  Since these addresses are private, they are not routable in a public domain and are effectively unreachable.  In the case of SIP, the IP address and port the SIP Device wishes to advertise for establishing a connection is contained in payload of SDP attached to SIP messages.  If this information is not translated, the far end will not be able to communicate with the SIP Device.  This usually creates a phenomenon commonly referred to as one-way RTP (voice path is only available in one direction).

Another issue with NAT functionality is that private devices are not reachable publicly unless a translation, commonly referred to as a bind, is created between the private IP address and the public IP address. This is done dynamically each time a private device attempts to communicate with a public device. The act of requesting communication causes the NAT to create a temporary bind between the private IP address requesting the communication and the public IP with which it is attempting to communicate.  Bind duration is controlled by a timer, which will expire and cause the bind to be removed if there is a period of inactivity on the bind equal to the length of the timer.  During the time the bind is active, public to private communication is possible, but once the bind becomes inactive, the private device is no longer publicly addressable.  The most common duration for this timer is between 30 and 60 seconds.  Also, binds can often be statically configured in a NAT.  This functionality is often referred to as port forwarding.  When this is done, the NAT is configured with a permanent bind between a private and public address.

With Konica Minolta Hosted PBX, the challenges presented by the presence of a NAT are addressed. A technique called NAT Traversal is used to overcome the issues created by the presence of a NAT.  Part of the Konica Minolta Hosted PBX call control platform is responsible for maintaining constant communication with all SIP Devices.  This constant communication ensures that the NAT bind timer never expires, effectively making the dynamic bind permanent.  Without this, a SIP Device in a private network would not be able to receive calls. Also, the Konica Minolta Hosted PBX call control platform uses a technique called Media Relay to overcome the issue where the NAT does not manipulate application layer information. This functionality allows the call control platform to discover the public IP address and port of the RTP stream once the SIP Device sends out its first RTP packet. The call control platform performs this function on both ends of a call and bridges the two legs of the call together, effectively relaying the traffic from one device to another.

Application Layer Gateway

Application Layer Gateway (ALG) is a method of manipulating IP address and port information at the application layer.  It is similar to NAT functionality in that it typically translates private IP and port information created by a SIP Device on a private network to public IP and port information on the WAN side of the router performing the ALG function. If done properly, this functionality negates the need for Media Relay functionality because all information advertised in the application layer is publicly routable.

Although this functionality is intended to improve the processing of VoIP traffic, not all ALG devices perform the application layer translation of packets properly.  In many cases, portions of the packet are modified when they should not be which causes interworking problems between the SIP Device and the call control platform.  When this occurs, the ALG causes the SIP Device to not function properly.

With Konica Minolta Hosted PBX, it is recommended that all ALG functionality between the SIP Device and the call control platform be turned off. Doing this eliminates the potential for the ALG to improperly translate packets, which could render service unusable.

Quality of Service

Quality of Service (QoS) refers to the ability to provide different priority to different applications over a data network connection to ensure higher priority traffic takes precedence over lower priority traffic.  
A voice conversation is real-time and traffic associated with a voice call must process efficiently or issues such as clipping or choppy audio will occur.  On the other hand, normal Internet traffic is best-effort. If packets are dropped or delayed, service is usually not noticeably disrupted.  As a result, voice traffic is typically considered to be higher priority traffic than data traffic.

Konica Minolta Hosted PBX utilizes Differentiated Services Code Point (DSCP) as the mechanism for marking packet priority.  Each SIP Device automatically sets every packet it sends as high priority. However, this does not ensure that all data network equipment in the traffic path will honor the setting and ultimately allow voice traffic to take priority of data traffic.

To ensure voice packets take priority over data packets, the Router(s), Firewall(s), and Switching must be properly configured to handle DSCP.  This functionality is sometimes referred to as Class of Service (COS) or priority queuing. In either case, it is recommended that the router be configured with strict priority queuing allowing packets marked with higher DSCP values to have higher priority.  If this is not done properly, perceived call quality could noticeably deteriorate during peak traffic times.

Packets set with high priority by SIP Devices only addresses traffic sent from the SIP Device to other devices outside of the customer’s network.  It does not address packets inbound to the SIP Device. These packets are normally not marked with a higher priority when received by the Router and/or Firewall because priority values are normally not maintained across a WAN.  As a result, without additional configuration these packets will not be prioritized over normal data traffic.  To accommodate this case, it is recommended that priority rules be established to allow all inbound SIP and RTP traffic to have higher priority than all other traffic.  

To assure QoS for SIP Devices on your internal switching DiffServ should be turned on and DSCP be enabled for all ports in all switches prioritizing QoS for Voice RTP Expedited Forwarding EF (DSCP 46).  Some manufactures use Auto or Simplified QoS settings that configure QoS to known best practices making the process easier. Please consult your switch manufacturer and/or Network Administrator for further information and configuration of QoS.

Bandwidth

Internet Bandwidth

Internet bandwidth is the amount of capacity available for Internet traffic on a network.  This amount is determined by the service provided by the Internet Service Provider.  The amount of bandwidth available will determine the amount of simultaneous voice calls and data traffic that the Internet connection will support.  If properly sized and with the proper QoS settings in the router, Konica Minolta Hosted PBX will function properly.  However, if undersized or if QoS is not provisioned correctly, perceived call quality could noticeably deteriorate during peak traffic times.  The following information provides information and guidelines for properly sizing voice service for a given Internet bandwidth.

To determine the number of phones that can be supported over a given bandwidth, the maximum number of simultaneous calls that can be supported must first be calculated using one of the following formulas.

1. Worst Case Calculation (No Compression)

   Max Calls = Available Voice Bandwidth (Kbps) / (SimCalls * 80Kbps)

   Available Voice Bandwidth (Kbps) – is the maximum amount of bandwidth allowed for voice traffic.  This value is equal to the lower of the connection download and upload speeds minus an amount reserved for processing data traffic. Sites with routers provisioned to prioritize voice traffic over data traffic can process voice calls at up to 100% of total connection bandwidth without jeopardizing call quality.  However, at sustained high call volumes, data traffic quality will be impacted.  As a result, it is recommended that calculations for maximum calls and maximum phones be done assuming only a portion of the overall bandwidth can be used for voice traffic.

   SimCalls – the number of simultaneous calls coming out of a site

   80Kbps – is the bandwidth required for a fax/modem call

2. Best Case Calculation (With Compression)

   Max Calls = Available Voice Bandwidth (Kbps) / ((Phone * 24Kbps)+(Fax * 80Kbps))

   Available Voice Bandwidth (Kbps) – is the maximum amount of bandwidth allowed for voice traffic.  This value is equal to the lower of the connection download and upload speeds minus an amount reserved for processing data traffic.  Sites with routers provisioned to prioritize voice traffic over data traffic can process voice calls at up to 100% of total connection bandwidth without jeopardizing call quality.  However, at sustained high call volumes, data traffic quality will be impacted.  As a result, it is recommended that calculations for maximum calls and maximum phones be done assuming only a portion of the overall bandwidth can be used for voice traffic.

   Phone – the number of simultaneous phone calls with compression coming out of a site

   24Kbps – is the bandwidth required for a phone call with compression

   Fax – the number of simultaneous fax calls (no compression) coming out of a site

   80Kbps – is the bandwidth required for a fax/modem call

There are certain call flows in Konica Minolta Hosted PBX that do not support compression, such as calls to voice mail or to the conferencing service.  Therefore, the actual amount of bandwidth required will vary between the best and worst case calculations.

The maximum number of phones that can be supported over a given bandwidth can now be calculated using the following formula:

Max Phones = Max Calls * Users per Simultaneous Call

Max Calls – is the amount of simultaneous calls that can be supported over the given bandwidth.

Users per Simultaneous Call – is a statistical approximation of the total number of users that can share one call path with non-blocking results.  The value of 4 is recommended for average site usage.  However, this number could vary drastically depending on the type and size of the site.

The following two tables provide estimates for two different site applications.  The first provides estimates for an average usage site, and the second provides estimates for a high usage site.  The actual values for a give site application will vary depending on actual usage requirements.

Note: Sites with routers provisioned to prioritize voice traffic over data traffic will be able to process more voice calls without jeopardizing call quality. However, if call volumes are extremely large, data traffic quality could be impacted. As a result, we recommend that bandwidth engineering be done considering only a portion of the overall bandwidth being available for voice traffic.

Local Area Network Bandwidth

Local Area Network (LAN) bandwidth is the amount of capacity an internal network can support.  This amount is determined by the throughput specification of the LAN infrastructure.  In most applications, the LAN infrastructure is a single layer 2 switch.  The amount of bandwidth available will determine the amount of simultaneous voice calls and data traffic that the LAN will support.  If properly sized, Konica Minolta Hosted PBX will function properly.  However, if undersized, perceived call quality could noticeably deteriorate during peak traffic times.  It is vital to properly size the network to support the addition of VoIP traffic to the network.

Common Terms and Acronyms