Frequently Asked Questions

Is my network VoIP ready? How much bandwidth do I need?

VoIP calls typically require 24kbps to 90kbps of bandwidth (200kbps is ideal), but video calling and conferencing can require 3 to 20 times more bandwidth. In planning for your network, estimate how many simultaneous calls your organization expects to have at any given time and multiply those calls by 200 kbps.

Our average customer has about 10% of the organization on the phone at any given time. As an example, if you have 20 employees and anticipate 2 simultaneous calls at any given time, the network would ideally have 2*200kbps, or 400kbps of bandwidth dedicated to VoIP calls. Note that if your company heavily utilizes your network for bandwidth-intensive activities such as streaming video, you should account for that traffic as well. No need for a QoS device; we simply suggest you account for your bandwidth needs.

This test is a general approximation of the quality of your VoIP connection. It does not test whether there are any active issues with your router that are impairing the quality of your connection.

Here are the typical issues associated with a network with under-allocated bandwidth:

  • Packet Loss generally occurs when there’s not enough bandwidth on your network (usually <24 kps) for an acceptable call to take place, though hardware issues and Internet connectivity problems can also lead to this. Packet loss cause gaps and stutters in your call.
  • Latency refers to the amount of time it takes a data packet to reach its destination. The lower the latency, the clearer your call will be.
  • Jitter measures how much latency varies within your network, and it is often caused by network congestion and route changes.

Firewalls can complicate communications over a VoIP network. Firewalls sometimes perform Network Address Translation (NAT), a process OnSIP already takes care of with a “server-side” solution. We do not recommend disabling NAT traversal technologies, because of security concerns. But multiple NATs, SIP ALGs, SIP helpers, and any process that inspects packets can disrupt the quality of your call. We also do not recommend securing traffic by port range, but we do encourage users to open the router to our IP blocks for UDP traffic only, since the port for every call for audio will vary.

How do I keep my existing telephone number?

Porting a number to OnSIP typically takes anywhere from 10 business days to 6 weeks. To begin the porting process, log in to your OnSIP admin account and select 'Port an outside number into OnSIP' under 'Phone Numbers'. Fill out the form and we will get back to you on whether or not your numbers can be ported. We try to work with all telephone service providers. However, certain local or regional guidelines may preclude your current provider from releasing your number.

Porting costs $15 per number. While you wait for your number to port, we can offer you a temporary OnSIP phone number to forward your calls to so your customers can still reach you at your existing phone number.

How do I get a new phone number?

We have phone numbers available for OnSIP customers in almost every region of the United States. An updated list of these numbers can be found on our phone number availability page. We also offer vanity numbers.

How do I enable international calls?

Log in to your OnSIP admin account. Click on 'Accounts' in the top navigation, and then select 'Enable international dialing' under 'Support'. Download and complete the form. You can email it to billing@junctionnetworks.com or fax it to 215.754.4477.

Calls to international countries are not included in the per seat unlimited calling plan. They will be billed separately per minute.

A full list of international calling rates can be found here. Note that you may not need international dialing. If countries you need to call are in our standard calling area, they are included in per seat unlimited. If your company is on the pay as you go plan, calls to the 20+ countries in our standard calling area cost 2.9 cents per minute.

What phone should I use with OnSIP?

We allow you to bring your own SIP phones to our service. We don't sell phones, but we do review them. To see a list of the phones we use in our own offices, click here.

What is your standard calling area?

OnSIP's standard calling area is a list of 20+ countries where inbound and outbound calls are charged at a standard rate of 2.9 cents per minute for pay as you go customers. If your company is on the per seat unlimited plan, you get unlimited calls to any country within our standard calling area.

The area includes the lower 48 states in the US, and most of Canada. To view the entire list of countries, click here.

How do I participate in the introductory webinar?

We host an interactive webinar twice a week, 2:00 PM Eastern every Tuesday and 12:00 PM Eastern every Thursday, for everyone who’s just getting started. The webinar lasts about 30 minutes, longer than the time it will take you to set up your OnSIP phone system, and is followed by Q&A. Click here to join and dial 1-888-864-0686 at the start of the session.

Are there any contracts to sign or long-term commitments you’re not telling me about?

You can cancel at any time, without ever having to worry about penalties.

How does the 30-day free trial work?

If you decide to cancel your service within 30 days, you will not be charged for any of our hosted services (voicemail boxes, auto attendants, groups, etc). Calls to and from the public switched telephone network are not included in the trial. We accept Visa, Mastercard, and American Express. Please have a valid billing address in the United States.

How many simultaneous calls will my OnSIP phone number support?

All of the phone numbers you get from us (both toll and toll-free) support up to 25 simultaneous calls. You’ll find that this is in stark contrast to competing services that advertise “unlimited inbound calling” but restrict their numbers to just a single call at a time. With OnSIP, your customers will never hear a busy signal.