WIKI VoiP VoIP Call Relay


Call Relay

VoIP Call Relay


More intellectual way of VoIP switched connections between different IP networks 
Тenor Call Relay™ ensures the organization of the single central VoIP portal between IP networks, thus allowing achieving VoIP connections trough the segments of IP networks. All the calls are connected through the segments of IP networks, using only one compression and decompression of the voice signal. As a result, while overcoming VoIP packets, a shorter delay and higher quality of telephone connection is achieved.
Tenor Call Relay allows end points of VoIP, which are behind a Network Address Translation (NAT), to establish connections with the external VoIP networks. This enables companies and suppliers of the services to broaden its VoIP networks for the clients, as well as using the Internet for this purpose. 

Tenor Call Relay provides centralized disposal of calls, administration and the security of your VoIP network.
Due to the unique intellectual switching of VoIP-network, Tenor Call Relay allows you to easily and securely extend the scope of the VoIP calls. Call Relay is available in two versions: Call Relay 120 – supports up to 120 simultaneous calls and Call Relay SP – supports up to 672 simultaneous calls.

Much higher level of intellect means higher quality of communication and shorter delay.
When using a circuit switched connections, independent VoIP networks unite, multiple processes of compression and decompression voice packet increase the transmission delay, and are the reason behind the loss in the quality of conversation. Thankfully to its direct IP connection between VoIP networks, eliminates those delays and therefore increases the quality of the conversation.
More intellectual level eliminates the firewall problems
Using the NAT Access technology, Tenor Call Relay may direct VoIP traffic transparently through the NAT firewall IP networks.

Higher intellectual level means lesser system overload
Due to Packet Saver™ technology, Tenor Call Relay reduces the load time of VoIP connections by the way of multiplexing them into the one and the same packet, thus reducing the overall usage of the bandwidth up to 57% after compression and masking.

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