Here, we hope to address these issues for you to make your transition as seamless as possible.
Should you still have questions, please feel free to contact us and we will go over any extra information you may require.
What is a SIP Trunk?
Calls can be made as normal over a SIP trunk to the public switch telephone network (PSTN), such as landlines, mobile calls, and international calls.
The major telcos are moving customers to SIP trunks (both in Australia and globally), with many already announcing shutoff dates of their ISDN and analog services over the next 5 or so years.
Replacing a users traditional phone system with a SIP trunk allows the provider to save money – but those savings aren’t always being passed on. Customers have the opportunity to weigh up their options and significantly reduce their call and infrastructure costs.
What are the benefits of using a SIP Trunk?
Call costs, for both domestic and international calls are significantly reduced. It’s not unusual for call costs to be reduced by 60% when moving from traditional to SIP services.
Number Portability allows you to move locations and keep the same number (not just forward it, but actually keep it for inbound and outbound calls). This means that you’re no longer limited to the 02 prefix in NSW, and so on.
International SIP Trunks allow you to provide an Australian number to an overseas company or branch office (or vice versa) – and for all calls made from that office to Australia to be treated and billed as local Australian calls.
How does SIP call quality compare with ISDN and traditional call quality?
Will all your SIP calls be high-quality?
That depends on a few things.
The first is your connection itself – see bandwidth recommendations – and the second is the protocol used. SIP calls can use one of several different VoIP protocols – the main ones are G.729, G.711, and G.722.
G.729 uses the highest compression, so is great where there are bandwidth limitations, such as on a low speed internet connection. It is comparable to a mobile phone call.
G.711 is the default for most VoIP communications and uses lower compression. It is comparable to the highest quality analog phone call.
G.722 provides better quality than the public switch telephone network (PSTN), but requires additional bandwidth. For applications where voice quality is key it is the preferred choice.
The other protocols supported by Aatrox Communications are listed below.
How many SIP Trunks do I need?
A general guide is to divide the number of users by 3, but that should only be used as a guide and will vary based on how you use telephony.
It’s important to be aware that a conference call will use a SIP line for every external call participant.
How much bandwidth do I need for SIP Trunking?
Using the G.711A codec (standard) each call needs a minimum of 100kbps upstream bandwidth.
|Number of concurrent calls||Minimum bandwidth (upstream) for G.711A|
Other codecs may have higher bandwidth requirements.
Are there any other internet connection requirements?
Most business grade internet connections will be completely suitable for VoIP.
Some less common connection types like satellite are not suitable for VoIP.
Aatrox Communications can provide more information on whether your connection will be suitable for VoIP.
Can I keep my phone number(s) with VoIP?
Any existing land line number or range of numbers can be ported to Aatrox Communications and attached to your SIP trunk, which allows you to make and receive calls from it.
This includes 1800, 1300 and 13 numbers.
Numbers may be single or in a range of 10 or 100.
To port your number and attach it to your SIP Trunk, we will need the permission of the account holder. Depending on a number of factors outside our control, ports may take several weeks.
While your number is porting you are able to forward it to a temporary number assigned to you by Aatrox Communications, which is never shown to people you call.
How do I use a SIP Trunk?
On your PABX or device you enter these details, which allows the SIP trunk to be “registered” to you.
Calls made to the numbers assigned to your SIP Trunk will be routed over your SIP Trunk, and you can choose where and how they are treated from there (for example ringing a single handset, going to voicemail, or using something like a call queue, hunt group, or IVR).
Note: To get the most out of your PABX and SIP trunks you may choose to configure plenty of different options for routing, codec usage, number masking, and so on.
We’re more than happy to have a chat with you about your specific requirements.
Are Aatrox Communications SIP Trunks expensive to implement?
Our SIP Trunks should reduce both the fixed infrastructure portion of your bill, and the call cost portion.
How do Aatrox Communications SIP Trunks compare to others?
We have worked hard and invested in our infrastructure to ensure that our call quality is excellent and extremely reliable, and in our team to make sure that there is always a highly skilled engineer available to answer your call.
All of our infrastructure is in Tier 1 data centres within Australia to provide the lowest possible latency (learn more about our infrastructure) and all of our team is located in Melbourne and Sydney.
We’re consistently told that the support we provide is second to none.
Feel free to stop by for a visit!
How much expertise do I need to get started with VoIP?
Aatrox Communications can provide assistance with SIP Trunk registration and compatibility – but ultimately you are responsible for running and configuring your PABX.
If this is not something you’re comfortable with, give us a call – we work with some great Hosted PABX providers that we’re more than happy to put you in touch with.
What infrastructure does Aatrox Communications use to deliver SIP Trunks?
What happens if my PABX goes offline?
Aatrox Communications SIP Trunks can be configured to automatically failover if registration to your PABX fails.
This failover can include voicemail to email, a mobile number as an endpoint and so on.
Contact us to discuss high-availability and redundancy strategies.
How does Fax work over a SIP Trunk?
Depending on the configuration and functionality of your PABX, faxes may be able to be delivered to it directly over your SIP trunk.
If this functionality is not available, fax to email can be configured on Aatrox Communications’ SIP infrastructure.
How long does it take to provision a SIP Trunk?
Our Unlimited SIP Trunks take a little longer – but still usually under 24 hours.
Which PABXs are Aatrox Communications SIP Trunks compatible with?
Which phones can I use with Aatrox Communications SIP Trunks?
You can connect any SIP-compliant physical phone or softphone to an Aatrox Communications’ SIP trunk.
That said, it is far more common to use a PABX such as 3CX.
The telephone requirements are then set by the PABX.
Can I connect an analog PABX with Aatrox Communications' SIP Trunks?
By using a PSTN Gateway (also called a VoIP Gateway) you can connect a SIP Trunk to an analog PABX.
The PSTN Gateway is a physical piece of hardware that essentially converts the digital SIP voice signal into an analog one that is compatible with your existing PABX.
This can directly replace one or more IDSN2, ISDN10 or ISDN30 connections, and immediately provides significant cost savings.
Can I use Aatrox Communications SIP Trunks outside of Australia?
You can register and use Aatrox Communications’ SIP Trunks from anywhere in the world (some anti-hacking restrictions apply but can be lifted on your account by request).
We have customers in the US, the UK, Europe, New Zealand, India, South East Asia, and the Pacific Islands, all using Aatrox Communications’ SIP Trunks to make and receive calls as though they were in Australia.
This saves money, and gives them additional flexibility.
Will Aatrox Communications' SIP Trunks work with my softphone (e.g Xlite)?
However as mentioned above, it is far more common to use a PABX such as 3CX. The telephone and softphone requirements are then set by the PABX.
Do Aatrox Communications SIP Trunks have the availability to dial 000?
However, if your internet connection goes down and you have no redundancy, you will not be able to make any calls (including to 000).
What codecs do Aatrox Communications' SIP trunks support?
- G.711A (also known as G.711alaw)
- G.711U (also known as G.711ulaw)
Codec selection is performed during call setup as per RFC3265. All calls are transcoded to G.711A for termination on the PSTN.
Does it cost anything to get started?
Your SIP Trunks are billed monthly in advance (so you would be charged for your SIP Trunk for July on the last day of June).
What payment methods are accepted?
We accept VISA, MasterCard and American Express cards.
How will I be charged for call usage?
We will alert you if your call charges surpass $100 per SIP Trunk, and depending on how long you have been a customer we may require part-payment of your bill.
This number can be lowered by request, or increased by agreement.
Can I view calls and call charges as they happen?
Aatrox Communications provides live dashboards with all SIP trunks that show all calls made, as well as their type, cost, and specific details. Learn more about our VoIP Dashboards.