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CX SIP Proxy

CX SIP Proxy streamlines call routing within Session Initiation Protocol (SIP) networks. This highly scalable and stateless call routing engine optimizes network performance and simplifies troubleshooting processes. CX SIP Proxy is a stateless routing control between SIP network elements. It acts as a load balancer, session controller and a security gateway for anything related to voice.

In a SIP call flow it can relegate to a SIP B2BUA (such as Cisco CUBE or EF Media server), switch a call from one to the other (if self-service is required) or execute SIPREC forking for recording/ transcription/ voice biometrics. It also ensures failover if any of the active components (SIP B2BUA) should fail. It can also forward a call to the EF Media server for initial (Conversational) AI treatment, and then execute an explicit transfer/ forward to a third party voice component.

For high availability, CX SIP Proxy can be deployed in a cluster mode with two or more instances.

CX SIP Proxy can act as a replacement of CUSP. For more information see CX SIP Proxy as a CUSP replacement



Proxy for SIP Unified Communications signaling

Provides proxy functionality for SIP-based communication, allowing for the routing of signaling messages between different endpoints in a network. This functionality is being provided through the use of the core tm (Transaction Management) module and the dispatcher module for load balancing and routing of SIP messages to endpoints.

Signaling Support

Supports a wide range of signaling protocols for voice. 

Address Resolution

Provides support for address resolution using DNS for translating phone numbers into SIP addresses. 

TCP, User Datagram Protocol (UDP), and Transport Layer Security (TLS)

Supports both TCP and UDP for SIP message transport, as well as TLS for secure transport. This is provided through the use of the proto module for protocol handling and tls module for TLS support.

Redundancy and high availability options

Provides support for redundancy and high availability through the use of load balancing and failover mechanisms. 

SIP header manipulation

Provides powerful SIP header manipulation capabilities, allowing for the modification, addition, and removal of SIP headers. This is achieved through the use of the hdr module that provides various functions for accessing, modifying, and manipulating SIP headers.

Support for IPv4 and IPv6

Supports both IPv4 and IPv6 network protocols for SIP communication, allowing for seamless communication between different network types. This is achieved through the use of the proto module that supports both IPv4 and IPv6 transport protocols.

Call Admission Control

Provides support for call admission control. This allows for the control of call traffic and resource allocation. This is achieved through the use of the CAC (Call Admission Control) module that provides various functions for monitoring and controlling call traffic.

Dynamic Call Routing and Load Balancing

Provides support for dynamic call routing and load balancing, allowing for the intelligent routing of call traffic based on various parameters such as caller ID, user location, and call quality. This is achieved through the use of the dispatcher module, which provides advanced routing and load balancing capabilities.

QOS Support

Provides support for quality of service (QoS) through the use of differentiated services code point (DSCP) and IP precedence marking. This allows for the prioritization of SIP traffic and the allocation of network resources based on traffic priority. This is achieved through the use of the textops module, which provides functions for setting and manipulating DSCP and IP precedence values.

Monitoring and reporting through syslog, SNMP

Provides support for monitoring and reporting through various methods such as syslog, SNMP, and SNMP traps. This allows for the monitoring of SIP traffic, call quality, and system performance. This is achieved through the use of the syslog module for logging, and the snmpstats module for SNMP support.

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