Our Business VoIP Platform

Superior business communications services, delivered over a dependable, seamlessly scalable platform.

Our philosophy

OnSIP was built from the ground up with redundancy and scalability in mind. We use redundant servers, routers, Tier 1 Internet connections, and upstream carriers to provide an extremely reliable VoIP service. Today, we have two data centers, located in renowned carrier hotels at 60 Hudson St, New York, NY and One Wilshire Blvd, Los Angeles, CA

OnSIP in action

Above is a Google Maps mashup with our real-time call data. The pin drops represent calls being answered across the world using our platform.

The OnSIP network

OnSIP Hosted PBX is built using a substantially different paradigm than competing services, resulting in industry-leading reliability, geographic scalability, and low cost. Unlike competing services, OnSIP is not a collection of customer PBXs deployed in a datacenter. The latter setup is associated with limitations such as sub-optimal call routing (affecting call quality) through the customer’s dedicated box, service outages if the particular box fails for any reason, and scaling inflexibility.

simplified hosted pbx network diagram

The OnSIP network diagram: Many open source projects such as OpenSIPS, FreeSWITCH and Asterisk are used in the implementation of the current OnSIP platform. The "core" of the OnSIP platform consists of a geographically distributed cloud of SIP proxies. These SIP proxies appear as one entity, sip.onsip.com. Restated, OnSIP presents itself as a single SIP server, sip.onsip.com, to the outside world, which is a key characteristic: Instead of registering to a single box, OnSIP phones will connect to the geographically closest SIP proxy in the cloud. These proxies also handle NAT and firewall traversal on the server side, freeing customers from excess NAT or firewall administration.

Geographically distributed for maximum uptime

OnSIP's Location Service, a geographically distributed database, charts the shortest possible path for call routing while also ensuring enhanced reliability. All of our servers are redundant across our data centers to prevent network downtime caused by any single point of failure.

Tier-1 peering for optimized performance

We peer directly with all Tier-1 service providers, which lets us avoid all of the inter-carrier network hops that can compromise call quality.

A separate real-time network to avoid congestion

We send all real-time multimedia data traffic over a physically distinct network. This ensures that your media is not mixed in with bursty traffic that can cause congestion.

An undersubscribed network to prevent bottlenecks

We over-provision capacity and take a proactive approach to preventing congestion by restricting our network to self-limiting and fixed rate protocols ( SIP and RTP ) that do not attempt to use all available network bandwidth.

SIP standards-based for interoperability

The OnSIP platform is implemented with full conformance to RFC 3261 SIP: Session Initiation Protocol and the current Internet RFC standards for SIP, providing interoperability with a wide variety of devices and systems. We have completed extensive testing with phone manufacturers and SIP application developers from around the world.

Our low cost of service means you save

Because we do not pay licensing fees for our software or rely on third party proprietary technologies, we are able to charge no fee per seat, per extension, or phone, and instead offer services on a Pay-As-You-Go basis.

No hoops to jump through

OnSIP can be used with a wide variety of routers and has been tested with manufacturers including, but not limited to: Apple Airport, Cisco, D-Link, Juniper, Linksys, Netgear, and more, which can be found in our knowledgebase. OnSIP has been designed to handle NAT and firewall traversal with a server-side (cloud) solution. With OnSIP, no additional NAT or firewall administration is necessary.

OnSIP can be leveraged with any Internet Service Provider, provided that the customer’s network meets the minimum bandwidth and low latency, jitter, and packet-loss requirements for the VoIP deployment.

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