SIP Voice Architecture

The IP POTs card supports up to 6 standard Analog POTs interfaces via the 6 tip/ring pairs on the back side of the T1000 chassis.  These are the same pairs used on the R-POTs card.  The IP-POTs card also supports an IP-based SIP and RTP interface to standard Softswitches.  The SIP interface or Session Initiation Protocol allows the Softswitch and IP-POTs card to negotiate call control, codecs, IP Ports, encryption, security, etc.  The RTP or real-time protocol is used to convey the audio data to either the media servers or when direct media is used the parties called IP phones.  The SIP protocol is typically terminated into an SBC or session border controller or SIP proxy on the periphery of the Softswitch.  The SIP traffic is then forwarded onto the Softswitch for processing.

The IP-POTs, SIP and RTP channels are carried across the T1000 backplane to the GbE222 card and terminated as standard ethernet packets into the service provider's IP network.   The IP-POTs card will work with no issues within most standard service provider networks where SIP-based voice is provided.

The IP-POTS card SIP channel can be deployed using UDP, TCP, or TLS(encrypted and secured by certificates) transport of the SIP packets. The RTP or audio data channel can utilize either RTP (unencrypted) or SRTP (encrypted) for audio transport.