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1: How to place a call using BIPmd?

When placing a SIP call with BIPmd you will want to make sure that your PBX or device is configured properly using Username / Password authentication or IP address authentication.

An easy way to test a SIP Call with BIPmd is to use a softphone, such as Xlite or Zoiper, and configure a BIPmd trunk directly in the softphone. When making your SIP call from the softphone, you’ll want to be sure to dial the country code followed by the area code and then the number. For example, to dial the BIPmd main line, you’ll want to dial 13053402398. When calling other countries, simply enter the country code, followed by the city code and then the number. There’s no need to dial 011 in front of the number.

2: What kind of Internet connection do I need for your service?

Any broadband Internet connection will work with our service. Cable, DSL, T1-data, or Metro Ethernet are all supported. Unlike some other SIP providers, do not require you to get connectivity from us.

3: How much bandwidth do telephone calls consume on your service?

Telephone calls on our system are by default configured to operate on the G.711 voice codec, which consumes 85kbps of Internet bandwidth up and down. For example, a small DSL connection of 512kbps up and 3M down will have a limiting factor of the upstream 512kbps limit. Take 512 and divide it by 85 to arrive at a total of 6 maximum simultaneous calls on that particular type of Internet connection. That’s the very low end. Most broadband Internet connections these days are much faster, to the point where dozens and dozens of calls can traverse the data connection.

4: Where can I find configuration instructions for your SIP trunking service?

Wondering how to connect your PBX to a SIP trunk? Our Knowledge Base, contains examples of SIP trunk configurations for a variety of systems and devices along with other useful information to help you get started.

5: How many simultaneous calls are allowed on the unlimited SIP Trunks?

Each unlimited 2-way inbound / outbound channel can handle one SIP call (inbound to your local numbers or outbound to US48 / Canada). For example, if you have a PBX with 8 trunk lines, you would select 8 Channels for your rate plan. Our SIP Trunks can handle multiple calls, meaning in this example you don’t need to buy 8 SIP Trunks, you can get 1 SIP Trunk with 8 unlimited channels. If you need to upgrade to 30 unlimited channels in the future, you can do that all on the same SIP Trunk.

6: What ports do I need to forward to my system if it is behind NAT?

We recommend forwarding ports UDP/5060 and UDP/10000-20000 for standard FreePBX/Asterisk based installs. It may be possible to get your service working without port forwarding, but optimal service will be obtained with these ports. You can lock down port UDP/5060 to for additional security. You can not lock down UDP/10000-20000 to any specific IP address as we release the media on all calls to the closest carrier media gateway for optimal performance.

Learn how to Setup SIP Trunk for Asterisk PBX, FreePBX and others in our Knowledge Base.

7: What methods of SIP trunk authentication do you offer?

We offer the ability to do username/password or IP address authentication for our SIP trunks. These settings are configurable on your Control Panel under the Trunks page.

8: Do I need a dedicated public IP address to use your service?

No. You are not required to have a dedicated public IP address to use our service, although we do sell them and it is certainly recommended. If you have an Asterisk/FreePBX behind NAT, you will need to make a small modification to the sip_nat.conf file on your system (exact instructions can be found in our Knowledge Base.)

9: What is QoS (Quality of Service) and how do I enable it?

QoS or Quality of Service allows for the prioritization of traffic on your network and is usually configured in your router settings. It is imperative that QoS be properly enabled to ensure the successful transmission of VoIP phone calls. Without QoS, you run the risk of audio breakup and voice choppiness whenever your broadband Internet connection becomes saturated. For example, if you are downloading a large video file from a website and you do not have QoS enabled and a VoIP telephone call was initiated at the same time the voice packets would get lumped in along with the packets of the download. The audio would break up and in some cases become unintelligible until the download was complete. (To test QoS is properly enabled, you can initiate a large download on your computer while making a telephone call.) Fortunately, there are inexpensive routers with robust easy-to-configure QoS.