Hosted VoIP Network Requirements

SIP phones require a business-class network to function properly. Use these setup requirements as a guide to make sure your network is ready for Hosted Unified Communications service.

Network Hardware

Routers

Service Provider phones are network devices which require a business-class router with a broadband Internet connection. Each phone must be issued an IP address from the router (not modem) via DHCP.

Service Provider recommends an Ethernet Switch capable of QoS (Quality of Service) be placed between your Service Providerphones and your router. A QoS switch helps ensure high-quality audio during calls.

Firewalls

Many routers include a firewall which protects your network from malicious Internet traffic. Your firewall should allow your Service Provider phones to access HTTP, HTTPS, and UDP traffic on your network. Please contact your network IT professional for help in configuring your firewall.

Internet Connection

Internet Service Provider

All Service Provider phones require a quality broadband Internet connection to make and receive calls. Service Provider does not provide the Internet connection. Contact your Internet Service Provider (ISP) for information regarding your Internet connection.

Bandwidth Requirements

Each call on a Service Provider phone requires approximately 100kbps of Internet bandwidth (both upload and download).

For example, if you have a 5 Mbps Internet connection, or 5000 kpbs), you can make up to 50 concurrent calls at any given time. However, in many cases your computers and Service Provider phones share bandwidth, effectively reducing the amount of concurrent calls possible on your network. Service Provider recommends contacting your Internet Service Provider to ensure your bandwidth meets the minimum requirements for your needs.

Network Quality

Before deploying Hosted VoIP phones and/or Unified Communications services, your Internet connection should be pre-tested to verify that it meets minimum quality standards to support VoIP service. Key quality measurements for VoIP include:

• Latency – This is     defined at the amount of time between a request to a network and the     response given by the network. Optimal latency should be 100ms or less.     Latency that exceeds 150ms will have negative results, leading to unclear,     “choppy” calls.
• Jitter – Defined as     the amplitude and frequency that latency operates at; Jitter occurs when     latency either uncontrollably spikes or jumps beyond 20ms (either up or     down), which results in very poor audio quality.
• Packet Loss – This is data     send directly from a business’ network that is lost in the networking     transit. This should never exceed 1%, as this will result in poor audio     quality and/or dropped calls.

If your Internet connection fails to meet any of these quality standards, contact your Internet service provider for resolution.

There are several Web-based Internet Speed and VoIP testing sites that can be accessed to do basic pre-deployment Internet network assessment. Here are links to two such sites…

Internet Speed Test… http://reinvent.sipvoicetest.com/speed.html

Internet VoIP Test… http://reinvent.sipvoicetest.com/voip.html

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