SIPit Pro has been developed to allow SIP compliant codecs from various manufacturers to communicate with each other at ease, removing many of the networking issues associated with AoIP connectivity. The main benefits of SIPit Pro are: automatic PAT and NAT Transversal, algorithms to aid interoperability between differing hardware manufacturers, support for a vast range of audio codec algorithms and a cloud based control environment for the initiation, monitoring and disconnection of calls.
The SIP protocol is widely supported by audio codec manufacturers. SIP is used to establish real time audio sessions through the internet. SIPit Pro has been designed to ensure end to end compatibility between various SIP manufacturers to provide a seamless connectivity experience with minimal networking knowledge, saving both time and resources during the setup of an event. A SIP configured device is only programmed once and will retain the same ID as it travels the world, just like a mobile phone.
SIPit Pro offers broadcasters enhanced network security when interfacing their internal codecs to the outside world. This is accomplished by eliminating the need for open network ports, static IP addresses or special port forwarding rules. This is achieved by using a fixed point in the cloud that facilitates closed communication with the customer's router, locked down by IP address and port number.
SIPit Pro servers are capable of transversing full cone NAT and PAT gateways without any network reconfiguration. Each codec can register and stream to the SIPit Pro cloud server by simply plugging into a network port with suitable internet access anywhere in the world. Once registration has been establish, audio calls can be managed from either the user's equipment or from the SIPIT Pro WEB interface.
The WEB interface allow you to place your codecs into groups for specific jobs. Once a codec group has been created you can share it with other companies that are registered on the SIPIT Pro server. Once a share request been accepted by the receiving company, your shared codecs will appear in their account as "friends". The receiving company will then be able to see the status of your shared codecs and make calls to you.
The live WEB interface has been designed to make management of your codecs very simple. You can create your own codec instances within your domain and share them with other users as required. Each codec instance with its associated state is displayed. Codecs can be monitored for status and performance. In addition, the environment can be used to remotely initiate and disconnect of calls between registered codecs.
I think the product is amazing and massively advanced compared to anything on the market. Putting the science behind a great GUI. Knowing that I can have a server in almost any location around the world and a service that can communicate any brand of codec to any other brand of codec. SIP is now agnostic and easy. Confidence is easy with a very dedicated level of support.
At the FIBA 2019 Games we were faced with challenges interconnecting different brands of codecs. SIPITpro rose to the task and simplified this process, providing us with a powerful platform within which to manage this.
SIPit Pro has made connecting different vendor codecs hassle and stress-free. No more trial and error finding compatible algorithms and call modes between codecs! With the ability to remotely monitor call statistics, and initiate and drop calls from the web interface, AoIP delivery has never been so simple. Andrew and Martin were always happy to work through issues with us, and their support was invaluable.
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