As an entrepreneur you are constantly on the lookout for options to cut costs and to review expenses. Today’s digitized world requires to act with foresight as well as a fast and global network. Internet telephony is a great option to do this. Possibly you may have heard of the term SIP trunking already or one of your colleagues is already saving money with this technology.
In this article you will learn everything you need to know about SIP trunking so that you are ideally prepared when talking to a provider about your own business telephone system.
Hold on tight, we’re taking a trip down memory lane. Back then traditional telephony was analog only. For that a cable connection had to connect both callers and remain in place until the end of the call. All voice data was thus transmitted in an analog form.
Digitization came along with the evolution of ISDN, which on the one hand brought better performance, and on the other hand the capacity could be increased to the transmission of two calls over one line. This laid the foundation for SIP trunking.
In 1995, Internet telephony was already known, but not yet ready for mass use. At that time Internet access could only be used via ISDN or through an analog telephone connection. This meant having to deal with narrow bandwidth and usage (and payment) by duration.
Of course, this infrastructure was not working for real-time applications such as VoIP. Where possible, voice data was compressed so much that speech intelligibility and quality dropped.
In 1996, SIP (Session Initiation Protocol) was invented with the purpose of creating connections between multiple participants, or to enable multimedia applications on the network.
Since then this technology has been continuously developed, and by now the protocol used to establish a connection can also be used for data transfer or streaming.
By means of a simple example we would like to explain how SIP trunking works and what benefits it provides to companies.
Our sample company, UTG Heizungsrohre AG, is located in Augsburg, but has branches and production facilities in Taiwan, Uganda and China. For complex projects, manager Spitzer has had good experiences setting up weekly video meetings where the project managers of all branches talk to each other. However, Spitzer has been annoyed for years that the connection always breaks down and it is rarely possible to fit all participants into one conversation. He also worries about the security of the conversations and about data privacy. Last but not least, he is irritated by rising telephone costs, and he is dissatisfied with free video Internet services anyway.
This is where SIP trunking comes in.
SIP (Session Initiation Protocol) is a network protocol for communication between two or more participants using voice over IP (VoIP). This application layer protocol is required for audio or video real-time sessions between two endpoints.
Simply put, SIP is the technology that allows you to hold sessions with one or more participants within an IP network. There is no loss of quality, whether it is a call between two parties or a conference call.
A trunk is a virtual telephone line that connects the telephone system to the network provider. SIP trunking is therefore a type of telephone connection, but based on a data line – usually for VoIP telephony.
A SIP trunk is the virtual version of an analog phone line. A SIP provider can connect one, two, or twenty channels to your PBX by means of SIP lines, allowing you to make local, long-distance, and international calls via the Internet.
If you have a local PBX system in your office, a SIP trunking service provider can connect to you and you can make outgoing calls on your existing system without limiting the number of simultaneous calls.
Typically, your existing PBX system will be connected to Multiconnect‘s SIP trunk. Thereby it is possible to just handle individual phone numbers and terminals if desired, but based on your needs also entire phone number blocks and any number of extensions can be handled.
The customer has several options:
Hybrid solutions with an existing ISDN solution are also possible.
Multiconnect supports all conceivable end devices, from hardphone to softphone to smartphone. In most cases, we can integrate the technology invisibly into your company.
We are fully integrated into many systems, e.g. Freshcaller from Freshdesk. More information can be found here.
We are also a 3CX Platinum Partner – so our SIP trunk works seamlessly for 3CX.
SIP trunking costs vary depending on business requirements. Below you find an overview of the most important costs:
We are happy to create your individual offer – just send us your request via the contact form at the bottom of the page!
The metered SIP trunking is delivered when used and charged in such a way that a fee is incurred for each minute. This plan is very flexible as the number of simultaneous calls is unlimited and only one charge is made for each minute of each call. Metered services offer businesses the flexibility to add calls dynamically and pay only for the additional usage.
How many simultaneous calls can be handled on a SIP trunk with metered data? And what about bandwidth?
Call restrictions depend on your Internet bandwidth. Each non-compressed call requires a bandwidth of approximately 85 to 100 Kbps. For most Internet connections the upload speed is slower.
If you want to determine the number of calls your connection can support, you should use your upload speed as a starting point. For example, if your upload speed is 4 Mbps, the maximum number of calls is 47 (4,000,000 / 85,000).
In order to use SIP trunking, you need an IP PBX and an Internet connection with sufficient bandwidth. Different providers work with different manufacturers.
As a telecommunications service provider and network operator, we have our own telephone network with mass load capability. As a full system integrator we can thus offer you your entire telecommunications from a single source.
Moreover, if you wish, we can also connect to your existing SIP-enabled telephone system.
Multiconnect SIP trunking is tested and certified by 3CX
This means that you can easily connect your existing 3CX to our SIP Trunk via plug-and-play without any further configuration – leaving your existing processes unchanged.
We also offer cloud solutions. Learn more here.
We believe that SIP trunking is the future. ISDN switch-off is already being discussed before 2018 and could be replaced by VoIP telephony in the near future. Especially in the long term cloud based telephone systems are the right choice.
Familiarize yourself with the peculiarities of the system. The technology around SIP trunking can sometimes cause problems. However, with careful planning they can be avoided.
In fact, most problems are not related to the technology itself, but to the wrong system configuration.
If you take the following steps to prepare for SIP trunking, you will soon be able to enjoy the described benefits with minimal effort.
The most important requirement for successful SIP trunking is that also small businesses have sufficient bandwidth.
As you know, a typical voice call today requires a bandwidth of 85-100 kbps, if the audio data is not compressed by codes.
Follow the steps below to determine the total bandwidth you will need.
Multiply the total number of simultaneous calls that your company makes and receives when the maximum call volume is reached by 100. This will show your total bandwidth for voice traffic.
A simple calculation:
Number of most simultaneous calls during business hours
85 (100) kilobytes per second
For SIP trunking required bandwidth in megabits per second.
Add this number to the total bandwidth your business is already consuming when reaching maximum data usage. This will give you an estimate of how much bandwidth you will need for voice and data combined.
We recommend a router with a feature called Quality of Service (QoS). This feature serves as a “traffic cop” for your voice and data traffic. When a call comes in, the router notices that it must have first priority. It then allocates bandwidths automatically: all other users who are currently using YouTube and Facebook are automatically allocated a slower speed.
This ensures that audio quality and connection are maintained in good quality. This makes sense because you don’t want call quality to suffer or your customers to notice how overloaded your system is at the moment.
This ensures that audio quality and connection are constantly maintained in good quality. This makes sense because you don’t want call quality to suffer or your customers to notice how overloaded your system is at the moment.
In our opinion, this is a particularly important step. Various phone system manufacturers prioritize features differently. Some are custom, some are industry standard.
This means that you should find out which features are carried via SIP connection and which are not.
In addition, not all non-voice devices that use the telephone network to transmit data (e.g. fax machines) are compatible with the standards.
The best way to solve this complex problem is to take a thorough inventory of the connection to the phone system.
Explain to your provider:
“This is the data we want to transmit over SIP lines. Can you handle it? If any problems may arise, please let us know before you register.”
The best way to get to know a SIP trunking system is to test it extensively. Take advantage of the test phase.
Session Initiation Protocol (SIP) is a network protocol for connecting two or more participants. SIP also has other functions such as file sharing and video telephony.
The trunk is a virtual telephone line or channel that connects the provider to the telephone system. You can either integrate a SIP trunk into an existing ISDN network or supplement it.
A SIP-capable PBX system and sufficient bandwidth. The rule of thumb is 100 kbps per call.
The answer to this question also depends on your Internet connection. You can secure your VoIP telephony with encrypted SIP trunks. You will achieve the greatest security against eavesdropping if all callers make calls on the same network.
The discussion about VoIP security and encryption is absolutely justified. However, one must also keep in mind that ISDN connections were not encrypted at all. A hacker could therefore have obtained the data relatively easily.
A VoIP connection is always Internet based, and of course a capable hacker can gain access to data even in this case. The question is how to protect it. This is ideally done by means of encryption, since only non encrypted data is of any use to an eavesdropper.
Since a phone call always takes place between two connections, the provider must ensure that its connections are protected. For VoIP telephony, this works as follows.
To ensure that a telephone call is encrypted, the provider protects the SIP trunk (the IP connection) with capable encryption protocols. These include TLS (Transport Layer Security) or SRTP (Secure Real-Time Transport Protocol).
They are very secure. We are happy to advise you on this.
You can continue to use your existing phone numbers by taking them with you to the new provider. However, you can also add new number blocks at low cost.
Multiconnect‘s SIP trunking service is ideally suited for all companies from small to large that already use or want to use VoIP telephony and want to benefit from the low costs, the high scalability and flexibility.
If you are interested in digitizing your communication solutions, we will be happy to advise you by phone at +49 (89) 44 288-000 or you can send us your inquiry using the following contact form:
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