SIP Trunking

High quality, low latency

SIP Trunks

Bring your PBX service up to date with low cost SIP Trunking

  • Low cost / High quality
  • Instant routing / failover control
  • Quality and fault monitoring
  • Live Statistics
VoIP SIP Trunk

SIP Trunk


A SIP Trunk is a Voice over Internet Protocol (VoIP) connection from a VoIP telecoms provider to a VoIP capable device such as a PBX (telephone system). Session Initiation Protocol (SIP) is the signalling protocol VoIP devices use to setup the connection for voice traffic between these endpoints. A single SIP trunk can handle multiple calls.

Inbound Routing


Frequently Asked Questions


Our SIP trunking service is extremely efficient, with setup and configuration changes taking less than 1 minute to propagate across our network. We support IP authentication and Peer Registration which make it easy to setup on the client side and avoid firewall modifications such as port forwarding.

Voice over Internet Protocol (VoIP) is a technology that helps in the delivery of communication over the internet. This works by coding the voice signal from an ordinary phone into a digital signal which is then transmitted over the Internet. This technology allows the users to communicate, using SIP as the protocol language, with the global telephone network and gain access to features that aren’t available with traditional phone systems.

VoIP has multiple benefits over traditional telephone technology, and at significantly lower costs, which has made it the de facto voice channel for business of all sizes. A cloud hosted VoIP PBX does not require any on-site equipment, other than the actual deskphones, and does not require a second network for those phones. Employees have more ways to stay connected which greatly enables remote working as calls can be made and received from almost any device (desktop, tablet and mobile); SIP trunk / network routing options and failover settings are accessible, in stark contrast to legacy telephone services (PSTN/ISDN etc.), and VoIP services scale easily as your business grows.

Choosing the right VoIP provider can quickly become a mammoth task with so many offerings now available. To evaluate a service you should consider the following:

  • Redundancy - Speechpath have points of presence in 3 distinct datacentres.
  • Location - Speechpath has servers in Ireland and the UK providing least-hop routing, meaning faster connections and less chance of packet dropping or drops in voice conversations.
  • Security - what fraud protection measures are in place? - at Speechpath we enable destination access control lists, so in the event that your system is compromised your exposure is limited.
  • Quality - in VoIP we use a method called MOS (Mean Opinion Score) to calculate a call's quality. The highest possible score on this scale is 4.5 for standard codecs. At Speechpath we aim for 4.5 on all calls, we also provide you with a web interface so you can monitor your call quality.
  • Costs We offer competitive call rates and can build a package that suits your needs.

Typically allow 100 Kbs in each direction, so a 1Mb connection would be capable of providing 10 concurrent calls. Remember always to account for uploading/upstream speed as well as downloading/downstream capacity when calculating bandwidth, a mistake commonly made.

A VoIP gateway is a device that converts typical ISDN, basic rate, fractional or Primary Rate telecoms into VoIP SIP Signalling. Using a VoIP gateway is a reliable and cost effective method of implementing VoIP without changing the underlying hardware, such as a PBX telephone system.

Yes, we have porting agreements with all Irish and most UK Carriers and are able to port your numbers to our network. This process normally takes 1 day to complete (for Irish numbers) and once ported you will have full control of your numbers via our web management system. To initiate a port request navigate to our online store and submit a Port Request.

Why should you choose a SIP Trunk from Speechpath?


Session Border Controllers are essential in delivering a SIP trunking service. SBCs provide security, topology hiding, NAT traversal and many more functions ensuring your SIP trunk operates smoothly.

We pride ourselves on offering the best support available and with standard support included from 8.30 to 5.30 to optional 24×7 support if needed.

We directly interconnect to multiple local carriers including BT and Virgin Media and this enables us to offer premium quality calls with no additional hops required.

With an interconnect to Bandwidth (formerly Voxbone), we are able to offer telephone numbers across the globe.

We are Comreg registered and can offer our own numbering blocks which are available to use instantly.

We offer freephone numbers in Ireland and across the globe.

Gain access to the most competitive call rates available by connecting to our network, without compromise.

A SIP trunk with integration is unheard of, except when you use our service. We can provide a real-time data feed in JSON format for all your live calls via PubNub's data distribution service. This can be easily integrated into your backend systems for logging or CRM integration.

We monitor every call for quality including jitter and packet loss. This enables us to troubleshoot any issue in minimum time.

Authentication onto our network using your public IP or register to our network, whichever suits your setup.

Control which destinations are callable via your SIP trunk, limiting your exposure should your system be compromised.

Send signalling to our SBC network and RTP media direct to carriers, avoiding extra hops.

We manipulate your caller ids into correct formats for each carrier, ensuring your call always connects.

We support local dial plan assistance. Dublin users need not prefix with 01 nor Cork with 021, instead we match the number of digits dialled and prefix according to your settings.

G722 is a wideband HD audio codec and noticeably better than standard G711. Opus, another HD audio codec, is commonly used with WebRTC as it can handle packet loss and jitter with little effect on voice quality. We support both on our network.

Our network accepts SIP traffic in the form of UDP, TCP or secure TLS. We support all the options available.

Automatic rerouting of your incoming calls to a failover number ensures that you never miss a call should your VoIP device become unavailable.

All of our services, including SIP trunk settings, are configured via an easy-to-use web gui.