SIP FAQ
Q: Why would I want SIP trunks?
A: SIP trunks will dramatically reduce your calling costs by routing your calls over an IP network as opposed to the traditional public telephone network which uses analog or ISDN lines. There are also several other advantages. For example, your telephone numbers moves when you do, and you can answer a London call in Belfast.
Q: What level of voice quality can I expect using VoIP?
A: With the correct configuration you can expect quality equivalent to that of an ISDN line.
Q: How reliable is the service?
A: At the time of publication, we have had no interruptions to service since our first server went live more than a year ago. The only threats to availability are in the network connection between your premises and our data center. Either a line fault on your ADSL, or a core network fault can cause an interruption of service. These risks also apply to traditional telephone trunks, so the SIP service has a reliability comparable with traditional phone lines. It is also necessary that your PBX and ADSL router have power, so without backup power, an electricity outage will imply a phone outage at the same time.
Q: Are emergency services supported?
A: Yes, calls to 999 and 112 will connect you to the emergency services. However this is subject to your phone and internet equipment having power. Your interconnection connection must be switched on and working. If this is not the case, you will need to use a mobile or analog phone to call.
Q: What do I need to get started?
A: You need a SIP enabled device (PBX or handset) and a broadband internet connection, such as ADSL. We recommend the use of our VxDSL broadband product, which includes both the connection and the router.
Q: Can we keep our old number(s)?
A: Yes, old number can be ported so that you can receive incoming calls via SIP.
Q: What telephone number will we be given?
A: You can assign as many numbers to each SIP account as you want. You can select which area code you would like, so that you can have a local number.
Q: Is fax supported?
A: No, not currently. Fax over SIP is a very new technology. We are monitoring developments and plan to add this feature to our service in the future.
Q: DTMF doesn't seem to be working properly, why is this?
A: DTMF can be sent in one of many ways. By default, we support and expect DTMF using RFC-2833. It is possible that your system uses a different convention. To resolve this, review the documentation for your PBX, where you should find details of the method used to send DTMF. If it is configurable on your PBX, please set it to use RFC-2833. If not, please contact our support team and let us know which method your PBX supports, and we will configure our system to recognise that method for your SIP trunk.
Q: The voice quality is bad, why is this?
A: Voice quality is affected by several factors. The most important of these are packet loss, latency, and jitter. These are properties of the network between your device and us. If you are using our VxDSL service to connect, we are able to monitor and control these parameters, and you should contact us immediately if you experience degradation in voice quality. If you are not connecting with VxDSL, we are unable to do anything about your connection and you should contact your ISP for assistance. Please note however that most ISPs will see high latency or jitter as faults since these factors do not affect internet usage, and they do not treat voice differently to internet usage.
Q: What is a STUN server and what are the settings?
A: Simple Traversal of UDP through NAT (STUN) is a network protocol used by devices behind NAT in order to discover their public IP address and port, as well as the type of NAT which is being used.
Q: What codecs are supported?
A: By default we support the high quality G.711 codec using µ-law or A-law. We can also enable G.729 on request. Please contact our support team if would like G.729 enabled.
