A question often asked is: ‘What infrastructure do I need to get started with OnSIP VoIP service?’ The world of the Internet, clouds, networks and communication is fraught with complicated terminology, so I’ll try to distill the basics down to an easily digestible version.
Evaluating Your Network
Bandwidth is the measure of the resources consumed during an Internet transmission. VoIP requires an Internet connection of a certain speed (bit/s) to carry voice as data. VoIP typically requires anywhere from 24kbps to 90 kbps (~100 kbps would be ideal). Video calling, conferencing and other forms of communication may require anywhere from 3 to 20 times as much bandwidth, depending on the desired quality and number of participants.
The quality of the phone call also depends on the quality of the network. The major factors include:
- Packet loss: Packet loss occurs when some voice packets drop off due to congestion, increased traffic through the network or incorrectly configured hardware. Lost packets result in gaps and stutters in communication, but it is usually not a big concern with broadband networks.
- Latency: Latency is the time it takes a data packet to reach its destination, measured in milliseconds (ms). Obviously, the lower the latency, the better off you are (100ms is acceptable, while higher than 150ms will start affecting quality).
- Jitter: Jitter measures the variation of packet arrival times, or how much latency varies within the network, measured in milliseconds (ms). Jitter is often caused by network congestion or route changes. Jitter greater than 5ms can increase latency and result in packet loss. The effects of jitter can be mitigated by storing voice packets in a jitter buffer upon arrival and before producing audio, although this increases delay.
Network quality, including the above factors, can be tested using our VoIP test or other free online tools. We've found that network stability (no or low values of packet loss, jitter and latency) tends to be more important to VoIP implementations than Internet speed.
Network Recommendations for OnSIP
The Internet speed or bandwidth required for use with OnSIP will depend on the number of simultaneous calls that an organization expects at any given time. If a company expects to have 20 calls at the same time, the company would ideally have 20*100kbps (or roughly 2Mbps) dedicated to these calls.
To deliver great quality of service, OnSIP sets sensitive voice packets to either 0x18 (Low Delay and High Throughput) or 0x10 (Low Delay) (as defined by RFC 791, an international protocol specification). Quality can be further improved by implementing packet prioritization locally. Documentation of OnSIP’s quality of service is available in the OnSIP Knowledgebase.
Since OnSIP calls travel through the Internet, note that firewall settings may affect your ability to place and receive calls.
Firewalls may also perform Network Address Translation (NAT), the process of modifying IP address information in IP packet headers while in transit across a traffic routing device. NAT traversal is a technology most commonly used to allow multiple devices on a LAN with 'private' IP addresses to share a single (or limited number of) public IP address. 1
A block diagram can help understand what happens:
OnSIP uses SIP, a specific type of VoIP protocol (other VoIP providers may use other protocols). One of the technical challenges while implementing a SIP based VoIP solution is making everything work when a firewall and/or NAT is deployed between devices exchanging data. At OnSIP, it is recommended that NAT traversal technologies be turned off, as OnSIP utilizes a complete "server-side" solution to NAT traversal. This solution operates under the assumption that the end user is not employing any "client side" NAT traversal technologies on their devices or firewalls. In some cases, our server side solution can be confused by changes made by client side technologies - the net effect being that NAT traversal fails.
Furthermore, many of today's lower end commercial routers implement SIP ALGs (Application Level Gateways), which are enabled by default. An ALG understands the protocol used by the specific applications that it supports (in this case SIP) and does a stateful packet inspection2 (SPI) of traffic through it. A NAT router with a built-in SIP ALG can re-write information within the SIP messages (SIP headers and SDP body), making signaling and audio traffic between the client behind NAT and the SIP endpoint possible. While ALG could help in solving NAT related problems, many routers' ALG implementations are wrong and break SIP.
The only thing you really need (apart from phones) is a quality broadband connection. Internet speeds of around 5 Mbps are common3, and are sufficient for most small business users. Use our VoIP test to test latency, jitter and packet loss - these should be minimal. If you employ a firewall, OnSIP recommends that the NAT traversal options be switched off, as OnSIP employs a server-side solution.
If you need more technical information, visit our Knowledgebase and our previous blog posts on the subject linked under Resources. Or, our expert support team is just a phone call away!