Determines the media capabilities of the target endpoint.SIP supports address resolution, name mapping, and call redirection. Determines the location of the target endpoint.SIP then returns a message indicating why the target endpoint was unavailable. If a call cannot be completed because the target endpoint is unavailable, SIP determines whether the called party is already on the phone or did not answer in the allotted number of rings. Determines the availability of the target endpoint.Note SIP for the ATA 190 is compliant with RFC2543. Session management provides the ability to control the attributes of an end-to-end call. Signaling allows call information to be carried across network boundaries. Like other Voice over IP (VoIP) protocols, SIP is designed to address the functions of signaling and session management within a packet telephony network. SIP is an ASCII-based, application-layer control protocol (defined in RFC3261) that can be used to establish, maintain, and terminate multimedia sessions or calls between two or more endpoints. Session Initiation Protocol (SIP) is the Internet Engineering Task Force (IETF) standard for real-time calls and conferencing over Internet Protocol (IP). Installation and Configuration Overviewįigure 1-1 Cisco Analog Telephone Adapter.The ATA 190 also has an RJ-45 10/100BASE-T data port. The ATA 190 support two voice ports, each with an independent phone number. The ATA 190 analog telephone adapters are handset-to-Ethernet adapters that allow regular analog phones to operate on IP-based telephony networks. The section includes a brief overview of the Session Initiation Protocol (SIP).
#Ata chapter 92 software
This section describes the hardware and software features of the Cisco ATA 190 Analog Telephone Adapter (ATA 190). Cisco ATA 190 Analog Telephone Adapter Overview