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Internet Telephony – Capability Negotiation

Internet Telephony – Capability Negotiation. ENTC 345 Dr. Ana Goulart Assistant Professor. Announcements. Background section Exam#2 . Questions about the project? . Introduction and Background Network Model – WLAN Model Delay Analysis analytical model for estimating the delays

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Internet Telephony – Capability Negotiation

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  1. Internet Telephony – Capability Negotiation ENTC 345 Dr. Ana Goulart Assistant Professor

  2. Announcements • Background section • Exam#2

  3. Questions about the project? • Introduction and Background • Network Model – WLAN Model • Delay Analysis • analytical model for estimating the delays • Simulations • Metrics used to support your decision • How you’ve found some of the values in the table • Results • Final table • Conclusions

  4. Announcements Today: • SIP and capability negotiation

  5. Basic SIP call setup

  6. Capability negotiation • Allows end users to negotiate the media capabilities and flows that will be in the multimedia session • SDP-based offer/answer model (RFC 3264) • SDP = Session Description Protocol

  7. Offer/Answer Model – Type 1

  8. Offer/Answer Model – Type 2

  9. A typical SIP Message:header +body SIP header: INVITE sip::professor@ece.gatech.edu SIP/2.0 Via: SIP/2.0.UDP gtrep.edu:5060 From: Student <sip::student@ece.gatech.edu> To: Professor <sip::professor@ece.gatech.edu> Call-ID: 10000001@ece.gatech.edu Cseq: 1 INVITE Subject: Meeting Contact: Student <sip::student@ece.gatech.edu> Content-Type: application/sdp Content-Length: 160

  10. A typical SIP Message:header + body Body: SDP – Session Description: v=0 o=student 45345932 32847243 IN IP4 gatech.edu s=Session SDP c=IN IP4 130.207.230.189 t = 0.0 m=audio 9160 RTP/AVP 0 a=rtpmap:0 PCMU/8000 m=<media> <port> <transport> <fmt list> a=rtpmap:<payload type> <encoding name>/<clock rate>[/<encoding parameters>]

  11. Example: SIP and Cellular Access • The session description below might be sent by a SIP user agent using a cellular access. The user agent supports GSM on port 30000 and AMR on port 30002. When the remote party sends GSM, it will send RTP packets to port number 30000. When AMR is the codec chosen, packets will be sent to port 30002. Note that the remote party can switch between both codecs dynamically in the middle of the session. However, in this example, only one media stream at a time carries voice. The other remains "muted" while its corresponding codec is not in use. (RFC 3388)

  12. Example: SIP and Cellular Access v=0 o=Laura 289083124 289083124 IN IP4 two.example.com t=0 0 c=IN IP4 131.160.1.112 a=group:FID 1 2 m=audio 30000 RTP/AVP 3 a=rtpmap:3 GSM/8000 a=mid:1 m=audio 30002 RTP/AVP 97 a=rtpmap:97 AMR/8000 a=fmtp:97 mode-set=0,2,5,7; mode-change-period=2; mode-change-neighbor; maxframes=1 a=mid:2 RFC 3388

  13. SIP and ResourceReservation(QoS) Capability negotiation (RSVP) Capability negotiation

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