SIP is the latest technology allowing voice calls to be made over the internet. Voicecom uses SIP to provide low cost, ISDN replacement lines that provide the best possible call quality, inexpensively, with the flexibility that comes from using the internet.
Session Initiation Protocol (SIP)
SIP is split in to three different yet over lapping technologies; Voice, Text and Video. The G3 mobile network in the UK is built on SIP protocols. You may have heard of presence, as in best method of contacting some one, either by voice, text or e-mail, this is also based on SIP, although originally intended for the enterprise application, this has now rapidly moved into the SME market making this technology readily available to the smaller companies as well. Companies such as Microsoft have been actively using SIP as a protocol for a range of applications such as unified communications server.
The Voice World
SIP provides a signalling and call setup protocol for IP-based communications that can support a superset of the call processing functions and features present in the public switched telephone network (PSTN). SIP by itself does not define these features; rather, its focus is call-setup and signalling. However, it has been designed to enable the building of such features in network elements known as Proxy Servers and User Agents. These are features that permit familiar telephone-like operations: dialing a number, causing a phone to ring, hearing ring back tones or a busy signal. Implementation and terminology are different in the SIP world but to the end-user, the behavior is similar.
SIP is the next logical step in the evolution of voice communications; routing voice calls over the internet provides additional features over and above ISDN and analogue lines. They are more cost effective.
A SIP Trunk is the equivalent of an ISDN channel, depending on your telephony equipment you can mix and match the amount of SIP Trunks and telephone numbers/DDI ranges.
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 move when you do, get your broadband service set up at your new location and you are ready to go – no need for call diversion.
- Technically you can have a local Auckland number but be based in Invercargill – no tolls !
- You move across town to another location but your telco says you can’t keep the same number unless you pay to have it redirected because you are in a different exchange area – not with SIP you don’t !
- You get a web based portal to manage your phone line features and billing.
Q: What level of voice quality can I expect using VoIP?
A: With the correct configuration you can expect the quality equivalent to that of a digital 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 111 will connect you to the emergency services. However this is subject to your phone and internet equipment having power. If this is not the case, you will need to use a mobile or analogue 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 VDSL broadband product.
Q: Can we keep our old number(s)?
A: Yes, old numbers 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: Fax over SIP is a very new technology. We are monitoring developments and plan to add this feature to our service in the future. We can offer a fax to email service at very reasonable costs.
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 recognize 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 VDSL 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 VDSL, we are unable to do anything about your connection and you should contact your ISP for assistance. Please note however that most ISPs will not 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 ยต-law or A-law. We can also enable G.729 on request. Please contact our support team if would like G.729 enabled.