Start Your FREE TRIAL TODAY! SIP Trunking SIP trunking is a service that allows companies to connect their onsite IP PBX (Phone system) to the Internet Telephony Service Provider (ITSP) and use its services; inbound phone numbers and outbound call termination. It is effectively a replacement of old PSTN phone lines. A SIP Trunk service can be used via public internet as well as via private IP connections.

Advanced functionality

SIP trunk service is part of our mPBX system. This enables some unique features to be configured that are not normally available for SIP trunk services.

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SIP trunking
Frequently asked questions

SIP trunking is a way to deliver voice services over the internet. It works with a SIP enabled communication systems and handsets. SIP trunking is used to replace standard phone carrier services like ISDN, PRI and PSTN. SIP Trunk is mostly used if you have VoIP/SIP enabled onsite phone system.

VoIP is a broad term which refers to any call made over the internet rather than over a standard phone line. VoIP depends on IP (data connectivity) to transmit voice packets. SIP is a protocol used to initiate, maintain, and terminate voice communication sessions in VoIP applications. SIP protocols support the signalling and management of voice, video, and electronic messaging applications. SIP (Session initiation protocol) is used in VoIP (Voice over IP) technology.

SIP Trunks is a way of getting your onsite phone system connected to the outside world. SIP trunk replaces older PSTN and ISDN phone lines and uses the Internet to transmit voice data. Key benefits of SIP Trunks is lower cost compared to older PSTN / ISDN connections, ability to easily scale up or down with the number of channels (or simultaneous calls required). It is also easy to relocate as you could use your SIP Trunk anywhere you can get access to the internet. SIP Trunks can be configured to encrypt all voice data before transmission and allow for HD voice codes with significantly better voice quality to be enabled. SIP Trunks are also significantly cheaper compared to older PSTN or ISDN lines.

A SIP channel is used for one incoming or outgoing call. You can think about it as a sort of a standard phone line. Every SIP trunk can support multiple SIP channels. Number of channels is limited by your Internet connection speed, but most of the modern broadband connections are capable of supporting dozens of channels. E.g. 10Mbps connection could support about 100 simultaneous calls.

SIP trunking was designed to effectively replace older T1 or PRI lines. They allow you to connect your onsite IP phone system to the outside world. Rather than routing voice calls over copper lines and dedicated circuit-switched networks, SIP trunking involves sending the voice calls over data networks like the Internet or enterprise local area network.

Number of channels supported by SIP Trunk is limited by your Internet connection speed, but most of the modern broadband connections are capable of supporting dozens of channels. E.g. 10Mbps connection could support about 100 simultaneous calls. You would also need to order required number of SIP channels via customer billing portal.

VoIP (Voice over IP) requires data connection between your premises and our servers. In most cases it would be public internet. However you only need a tiny amount of bandwidth – connection speed in order to use VoIP. Each channel requires 80Kbps (0.08 Mbps), or using compressed codec even smaller – 0.03Mbps. Quality of VoIP will highly depend on the reliability of your internet connection. It does not have to be very fast, but it must be reliable and stable. We also recommend using dedicated internet link for setups with over 5 phones.

We have multiple solutions that can make sure you can continue to receive calls even when your internet goes down. You can have calls diverted to your mobile phone if you are offline, this way you can continue to receive calls without any internet. You can also have multiple internet connections for redundancy so if one connection fails you can continue to use the other.

Simply speaking – it is better than using traditional phone lines. We use G722 (High Definition Voice Codec). Connection will be crystal clear and better compared to normal PSTN services. Also, unlike many other services, all our equipment is hosted locally, meaning that there will be no delays in voice transmission.

Our systems are optimized for reliability, not price. We guarantee 99.9% SLA backed uptime on all our systems. All parts of the system are fully redundant, meaning that there is no single point of failure. This is not just words. We use independent monitoring company called hyperspin.com to monitor our server uptime. Visit our system uptime page.

We offer lower rates for larger call volume accounts. For services with minimum monthly spend we deduct the minimum monthly spend amount on the 1st of each month and this amount becomes call credit on your account. There are no extra charges for calls up to the minimum monthly spend, calls over are charged at the call plan rates.

Our head office is in Melbourne Australia. All sales and support is handled by our highly trained staff