Brekeke SIP Server

SDN/OpenFlow

OVERVIEW
BENEFITS
FEATURE LIST
ADVANCED EDITION
SYSTEM REQUIREMENTS
DOCUMENTATION
CASE STUDIES

Brekeke SIP Server manages flow control with SDN/OpenFlow support

SDN/OpenFlow support has been added in Brekeke SIP Server and Brekeke PBX for the version v3.6 or later. This means that Brekeke SIP server acts as a SDN/OpenFlow controller to manage OpenFlow-enabled switches for relaying RTP, blocking malicious IP addresses, and applying QoS.

RTP relay with SDN/OpenFlow

The Real-time Transport Protocol (RTP) is a network protocol used for delivering audio and video over IP networks. When RTP sessions are going through SIP servers for NAT traversal, it adds load on the SIP server. Thus, oftentimes the number of SIP sessions with RTP-relay handled by an SIP server is a common factor to measure the performance of an SIP server.

When the SDN/OpenFlow controller feature of Brekeke SIP Server is used, you can relay RTP packets through OpenFlow-enabled switches. That means the heavy loads introduced by RTP traffic will not be on the SIP server, rather they will be passing through the network over the OpenFlow-enabled switch. As result, it increases the capacity of a single Brekeke SIP Server to handle RTP-relayed SIP sessions.

Blocking malicious IP addresses and anonymous attacks

Brekeke SIP Server acts as a SDN controller: it can block and prevent attacks on SIP services. By defining IP addresses or rules set at the Brekeke SIP Server, the OpenFlow-enabled switch will block access from the specific IP addresses. To further protect SIP services, a unique Block List policy can be set at the Brekeke SIP Server to prevent any attacks to the SIP service.

Quality of Service (QoS)

For any VoIP implementation, especially for service providers, QoS is one of the most important requirements to ensure high quality VoIP/SIP service, and the delivery of voice or video packets without any delay or loss. With a network with OpenFlow-enabled switches, the Brekeke SIP Server can tweak the QoS of RTP packets based on call source and/or destination defined with DialPlan. This ensures the delivery of the QoS as defined and planned by SIP service providers.

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