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e*phone

rtspd. Quicktime. RTSP media server. RTSP. sipconf. Telephone. RTSP clients. SIP conference server. sipum. Telephone switch. SIP/RTSP Unified messaging. Web based configuration. sipd. T1/E1 RTP/SIP. SIP proxy, redirect server. SQL database.

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e*phone

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  1. rtspd Quicktime RTSP media server RTSP sipconf Telephone RTSP clients SIP conference server sipum Telephone switch SIP/RTSP Unified messaging Web based configuration sipd T1/E1 RTP/SIP SIP proxy, redirect server SQL database SIP/PSTN Gateway SNMP (Network Management) e*phone Hardware Internet (SIP) phones SIPH.323 converter Web server NetMeeting siph323 H.323 sipc Software SIP user agents httpd

  2. 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

  3. Web based configuration Call Bob sipd SIP proxy, redirect server SQL database e*phone Hardware Internet (SIP) phones sipc Web server Software SIP user agents 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 cs.columbia.edu

  4. PBX Gateway PSTN Internal T1/CAS (Ext:7130-7139) External T1/CAS Call 9397134 Call 7134 Ethernet 1 2 4 5 3 Regular phone (internal) SIP server SQL database sipd sipc Bob’s phone 7134 => bob PSTN to IP Call • DID - direct and simple • No-DID - dial extension, supports more users

  5. PBX Gateway (10.0.2.3) PSTN External T1/CAS Internal T1/CAS Call 5551212 Call 85551212 Ethernet 4 5 2 3 1 5551212 Bob calls 5551212 Regular phone (internal, 7054) SIP server SQL database sipd sipc Use sip:85551212@10.0.2.3 IP to PSTN Call

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