1 / 23

Presented by: Philipe BC Da’Silva

SESSION INITIATION PROTOCOL. Presented by: Philipe BC Da’Silva. WHAT IS SIP?. SIP is open, non-proprietary standard. Developed by IETF (Internet Engineering Task Force).

ringe
Download Presentation

Presented by: Philipe BC Da’Silva

An Image/Link below is provided (as is) to download presentation Download Policy: Content on the Website is provided to you AS IS for your information and personal use and may not be sold / licensed / shared on other websites without getting consent from its author. Content is provided to you AS IS for your information and personal use only. Download presentation by click this link. While downloading, if for some reason you are not able to download a presentation, the publisher may have deleted the file from their server. During download, if you can't get a presentation, the file might be deleted by the publisher.

E N D

Presentation Transcript

  1. SESSION INITIATION PROTOCOL Presented by: Philipe BC Da’Silva

  2. WHAT IS SIP? • SIP is open, non-proprietary standard. • Developed by IETF (Internet Engineering Task Force). • The Session Initiation Protocol (SIP) is the IETF standard for the establishment of multimedia sessions such audio, video, instant messaging, or real-time data communication sessions. • Seamlessly integrates with the internet. • SIP is a text-based protocol, similar to HTTP and SMTP, for initiating interactive communication sessions between users.

  3. WHAT IS SIP? • SIP is truly an internet-model protocol using ADCII message based on HTTP. • SIP support URL (with DNS). • Easy to decode and troubleshoot , it means web-type applications can support SIP services with minimal changes. • Easy integration and ability to inter-operate with other protocols and vendors. • SIP is flexible and scalable VoIP solutions. • Call setup is quick and incurs low overhead. • The handshake and data exchange to initiate a call is minimal compare to H.323.

  4. Integration with IETF • Other IETF protocol standards can be used to build a SIP based application. SIP can works with existing IETF protocols, for example: • RTP Real Time Protocol -to transport real time data and provide QOS feedback. • RTSP Real Time Streaming Protocol - for controlling delivery of streaming media. • SAP Session Advertisement Protocol - for advertising multimedia session via multicast. • SDP Session Description Protocol – for describing multimedia sessions. • MIME – Multipurpose Internet Mail Extension –standard for describing content on the Internet. • HTTP – Hypertext Transfer Protocol - HTTP is the standard protocol used for serving web pages over the Internet.

  5. SIP Architecture SIP Gateway • RTP carries the media directly between the endpoints

  6. SIP Header • SIP borrows much of the syntax and semantics from HTTP. • A SIP messages looks like an HTTP message – message formatting, header and MIME support. • An example SIP header: • ----------------------------------------------------------------- • SIP Header • ----------------------------------------------------------------- • INVITE sip:5120@192.168.36.180 SIP/2.0 • Via: SIP/2.0/UDP 192.168.6.21:5060 • From: sip:5121@192.168.6.21 • To: <sip:5120@192.168.36.180> • Call-ID: c2943000-e0563-2a1ce-2e323931@192.168.6.21 • CSeq: 100 INVITE • Expires: 180 • User-Agent: IP Phone/ Rev. 1/ SIP enabled • Accept: application/sdp • Contact: sip:5121@192.168.6.21:5060 • Content-Type: application/sdp

  7. SIP Phone- UIP200 1) Call Origination (Make a call) 2) Call Termination (Answer a call) 3) Talk 4) Disconnect (End a call) • Cheap and inexpensive compare to IPT Phones

  8. Features and Specification

  9. Connection UIP200

  10. IPT phone connection

  11. IP Network – (IPT & SIP AnyWhere) 1) Call Origination (Make a call) 2) Call Termination (Answer a call) 3) Talk 4) Disconnect (End a call) Singapore CIX Thailand CIX SLT (LIPU-X1A) (BIPU-Q1A) IP-Network DKT Branch Office (BIPU-M2A) Switch LAN (10/100Base-T) IPT IPT Personal Computer

  12. 1) Create Unidencom file-à LIPU IP address SIP phone configuration

  13. SIP phone configuration 2) Create SIP phone MAC

  14. Configure TFTP server SIP= 192.168.254.252

  15. How to configure LIPU/LIPS? Node ID 10 Node ID 11 LIPS LIPS IP-QSIG LIPU LIPU DKT 204 DKT 201 IP Phone DN 200 IP Phone DN 202

  16. Program 100

  17. New- Program 260 IPT LIPU = LAN1

  18. Program 260 LIPU is LAN1 LIPS is LAN2

  19. Program 304 -ILG

  20. Program 306 -OLG

  21. Program 325- New - Configure IP-Qsig as LAN 2

  22. Program 161 -New • LAN 1 ‘IP’ • Program 260 • For IPT • LAN 2 ‘IP’ • Program 325 • For IP-Qsig TOTAL = 32 CHANNELS Program 102, 653,654 ,655 ,656,671 & 672

  23. Strata Net IP connections LIPU-to-BIPU Enable

More Related