Unlike traditional ISDN and PSTN installations, SIP trunking doesn’t require any on-site installation. Instead, there are a few considerations that must be accounted for.
SIP trunks require an internet connection to run over. SIP can theoretically operate over any form of internet connection. However, it is always recommended to have a dedicated circuit or separate VLAN specifically allocated for your SIP traffic.
This will allow the SIP calls to receive unchallenged internet and operate at peak performance. The threat of other applications sharing your internet often leads to degraded performance. Jitter and packet loss are examples of symptoms experienced when sharing an internet connection with other business applications.
Where SIP is required before your dedicated connection is ready, you can route SIP traffic over temporary or backup circuits. ADSL and FTTC are good examples of these. These types of connection do not allow for segregation, so you cannot apply QoS or guarantee any level of service. Hence, they are never a long term recommendation.
When dedicating a connection, or portion of your existing internet to your SIP trunk, you need to know how much bandwidth is required.
SIP works on a concurrent call basis. This means rather than needing multiple trunks, you can have one SIP trunk with as many sessions as you require.
To work out how many concurrent calls you need, just calculate the maximum amount of people that could be on the phone at the same time. For example, if you are a busy call centre, you could have 100 employees that need to be on the phone at the same time. However, a hospital could have 100 employees but only 10 people making calls at any one time.
Once you have worked out the maximum concurrent calls, simply apply the methodology of 1Mbps per 10 concurrent calls. For the instance of 10 calls, your SIP trunk will be configured to allow up to 10 calls at any one site. This requires 1Mbps of bandwidth. The bandwidth you allocate on your internet connection needs to reflect this, in order to allow all 10 calls to initiate.
On top of your internet connection, your network must be configured to allow SIP traffic. Furthermore, it must be configured to let SIP perform at optimum performance.
Ensuring latency within your network is no less than 150ms will ensure that repeated speech and stutter does not impact your call quality
The same applies for levels of jitter. Ensuring your internal network jitter is less than 100ms will guarantee you do not experience voice transmission delay.
When network configuration is not ready for SIP traffic, call quality typically suffers as it must compete with other applications and services.
If your network is not currently operating at these standards, there are a few things to investigate before your SIP go-live. Network changes, upgrades and proactive monitoring are just a few things for you to look at as you prepare for your SIP installation.
If your PBX is not IP enabled
A common misconception with SIP is that you need an IP-ready PBX to utilise the technology. Natively, this is correct. You cannot connect SIP trunks to a legacy PBX. However, there is a way around this.
If your existing PBX is not IP Enabled, SAS can provide a media gateway to connect to the SIP Trunking service. We provide media gateways from Cisco and Sangoma. These gateways convert your non-IP PBX to an IP ready PBX, ready to be plugged into your SIP trunks.
Operating as an extension of your PBX, these gateways enable your PBX to connect to SIP.
For those businesses with access to a data centre, another option is to install SIP trunks into a data centre. Centralising SIP in one data centre allows each of your locations to be connected via the internet.
The connection of sites over the internet enables free calls between sites as all traffic is carried on-net. There are pros to hosting SIP in your data centre, especially if you already have the estate to leverage. However, it does leave your SIP service with a single point of failure.
The most common, and recommended method to connect to SIP is over your dedicated internet connection and directly into your PBX. This allows you to benefit from SIP connectivity, with the peace of mind that if you PBX fails, you can utilise the cloud for disaster recovery scenarios.
IP ready PBXs
If your PBX is IP-enabled, getting ready for SIP trunking is simple. Assuming your internet and network are ready for SIP, the next step is planning your migration.
The first step is user acceptance testing. Here, a SIP engineer will run through the most common calling scenarios with you on your PBX using a set of test numbers. Once testing is complete, you are ready to move your existing numbers from ISDN to SIP.
Number porting generally takes around four working weeks. You will need to consider timings and set user’s expectations.
If there is a period in the working day where calls are at their quietest, you will want to schedule your number porting for then. Sometimes, it’s hard to establish a best period for switching your numbers over. Generally, if you have a firm communications plan in place, users will be aware of a change in their telephony. Moving to SIP doesn’t incur any scheduled downtime but informing users that something is happening will ensure they are aware of any changes – and less likely to cause a stir when they notice any changes.
If your network and PBX are ready to kick off your move to SIP, it’s time to book a call with a SIP specialist. SAS regularly manage SIP migrations, as well as looking after the connectivity and network elements. For an expert approach that covers all angles of SIP implementation, book a free half hour chat with one of our experts.