How is SIP Trunking set up?
When migrating to SIP trunking, it’s important that you know what is going on behind the scenes. Once you’ve selected a provider to move you from ISDN to SIP, there are several key stages along the way.
As an IT Manager, you are the point of reference in your business for everything SIP. Whilst you will have the full support of your new service provider, the expectation is that you are always in the loop. We’ve outlined the five key elements of setting up SIP trunks to provide you with a schedule of events you should expect to experience when setting up your SIP connection.
The requirements gathering phase is the most crucial part of any SIP setup. You are likely migrating from ISDN. This means your legacy telephony setup will restrict your number of calls to the amount of ISDN lines you have run into your building.
When setting up SIP, you have the freedom to dictate the amount of concurrent calls you require. If your internet connection has bandwidth that can be made available, your SIP connection can offer as many calls as you genuinely need, rather than as many calls as your ISDN trunks can offer.
Capturing this requirement can be achieved by analysing your highest peak in your concurrent calls, then allowing for some overage to leverage a higher amount of lines available.
As you change from ISDN to SIP, another major requirement is the interoperability of your existing equipment. Firstly, you need to check your current phone system is compatible with SIP. Secondly, you may have a selection of other devices running off your ISDN and PSTN connections. Check for faxes, alarms and credit card terminals to make sure these are SIP-compatible or need an adapter or gateway.
Once you have satisfied these requirements, you will need to ensure you have enough bandwidth available and a readily configured network.
Bandwidth and network
To work out how much bandwidth you need, use the simple methodology of 1MBps per 10 concurrent calls. This is the amount of bandwidth you should ideally dedicate to your SIP traffic. Your internet service provider should be able to assist with creating a dedicated VLAN for voice traffic. By providing this, you ensure your voice calls are not impacted by other applications that require heavy bandwidth.
Further to your bandwidth allocation, your network must be configured to allow SIP traffic to perform without interference from other network components. Here are some key items to watch for when setting up SIP trunking:
- Ensure your network latency is no less than 150ms – this will ensure call quality does not suffer from repeated or delayed speech
- Ensure your internal network jitter is less than 100ms – this will guarantee you do not experience voice transmission delay
- Make sure your router and firewall are SIP compatible – older models may not be configured to pass through SIP traffic
If your network is not yet at this level, this is a crucial pre-requisite before you setup your SIP trunks. Network changes, upgrades and proactive monitoring are just a few things for you to look at as you prepare for your SIP installation. It is recommended that you contact your service provider to see if they can assist with a network readiness assessment.
When your requirements have been captured and your network is ready to handle SIP, your trunks can be configured. Unlike ISDN installations that you may be used to, SIP is configured remotely by your service provider.
You will need to connect your local internet connection to your PBX. This will involve simply running an ethernet cable from your router or firewall to your PBX. Once connected, the SIP configuration will route all voice transmissions over your internet connection to your phone system. When a dedicated SIP VLAN is established, the call traffic is uninterrupted.
Here you can apply quality of service (QoS). QoS is the description or measurement of the overall performance of a SIP service. This application enables your network to always pass through the best performance available from your network.
Before your numbers are migrated over to the live service, a dedicated testing environment can be setup for your SIP trunks. SIP testing is conducted to ensure every given call scenario is proven for your SIP environment.
Whilst SIP technology is tried and tested, it is important to test your dedicated partition. By completing a series of common calling scenarios, you have the peace of mind that your PBX is ready to use with your SIP trunks. If you have any other devices such as fax or alarm, you can test to make sure the SIP connectivity will work once all your numbers and devices are migrated over to the live connection.
The final stage when setting up a SIP trunk is porting your numbers. This is the process where your phone numbers leave their current ISDN infrastructure and migrate onto your new provider’s hosted SIP platform.
The porting planning phase requires interaction between your old and new service provider. Information such as address, numbers to be ported and an agreed date are exchanged.
On the day of porting, numbers will start to migrate across to your new service. Your provider will be on hand to provide updates as changes occur. Your numbers will start to move from one network to another at the given port time. Dependent on the volume of numbers being ported, this could be a short or long process.
Once all numbers have been transitioned from one network to the other, your ISDN installation will be made redundant by the network. All devices previously connected to your PSTN and ISDN services must be changed over to continue service.
Now that you are familiar with the SIP trunking setup process, it’s time to tailor a package to your requirements. Read on for information on how to choose a SIP trunking provider to best match your needs and to see what packages are available.