SIP Tunnel is specially developed to provide a communication tunnel for SIP (Session Initiation Protocol) and RTP (Real-time Transport Protocol) packets over the Internet. This is a great solution for VoIP, Audio and Video broadcast in real time and other applications, that are used on machines behind firewalls, NATs or proxies.
SIP and RTP are Internet protocols for the transport of real-time data. They are used for media-on-demand, VoIP calling phone numbers in PSTN and other Internet services. They use UDP packets to transport the data, which makes communication impossible when user’s machine is behind a firewall, NAT or proxy that restricts UDP communications. This is where SIP Tunnel can help – to provide a secure and fast communication tunnel for your application.
How it works?
There are different scenarious depending on how the user is connected to the Internet and the way your program is working. Our goal is to make the communication as fast as possible and avoid decreasing of the quality of the transfer data ( voice, video, etc. ).
One possible situation is when your application is a VoIP program, calling phone numbers in PSTN. If the user’s machine is behind a firewall, NAT or proxy, the communication might be impossible. In this situation you can use SIP Tunnel Client to establish a TCP tunnel for the SIP/RTP connection. This will let your VoIP application to use SIP/RTP protocols without being restricted from the firewall, NAT or proxy.
SIP Tunnel was created in joint development with Dingotel – a VoIP solutions provider. This cooperation gave us opportunity to test SIP Tunnel in real life cases and ensure its high quality and conformance to the SIP standard.