Architecture rtspd SIP/RTSP Unified messaging RTSP media server sipum Quicktime RTSP clients RTSP T1/E1 RTP/SIP Telephone SIP/PSTN Gateway switch SIP conference server sipconf Web based configuration Web server SIP proxy, redirect server SQL database sipd e*phone sipc Software SIP user agents Hardware Internet (SIP) phones SIPH.323 convertor NetMeeting siph323 H.323 SNMP (Network Management)
Demo Scenario Web interface SIP-phone to SIP-phone SIP-phone to PSTN phone PSTN phone to SIP phone Device control using SIP Voice mail service (unified messaging) Multi-party conferencing Network management (SNMP) SIP-H.323 translation
Example Call Bob signs up for the service from the web as “bob@cs.columbia.edu” sipd canonicalizes the destination to sip:bob@cs.columbia.edu He registers from multiple phones sipd rings both e*phone and sipc Alice tries to reach Bob INVITE sip:Bob.Wilson@cs.columbia.edu Bob accepts the call from sipc and starts talking Web based configuration Web server Call Bob SIP proxy, redirect server SQL database sipd e*phone sipc Software SIP user agents Hardware Internet (SIP) phones cs.columbia.edu
PSTN to IP Call 1 2 3 5 4 DID - direct and simple PBX PSTN External T1/CAS Regular phone (internal) Call 9397134 1 Gateway Internal T1/CAS (Ext:7130-7139) Call 7134 2 SIP server sipd Ethernet 3 sipc 5 Bob’s phone SQL database 4 7134 => bob DID - direct and simple No-DID - dial extension, supports more users
IP to PSTN Call 4 5 3 1 2 PBX Internal T1/CAS Call 85551212 Regular phone (internal, 7054) PSTN External T1/CAS Call 5551212 5 5551212 Gateway (10.0.2.3) 3 Ethernet SIP server sipd sipc 1 Bob calls 5551212 SQL database 2 Use sip:85551212@10.0.2.3