The State of SIP and How it Enables Unified Communications
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The document discusses Session Initiation Protocol (SIP) and provides examples of how SIP is used to set up voice and video calls between two endpoints. It explains the different SIP messages used - INVITE, OK, ACK, and shows how SIP establishes a voice or voice/video "stream". It also discusses SIP trunking, SBCs, codecs, QoS, NAT and security considerations. The last sections provide an overview of SIP certification through The SIP School and some example questions to ask an ITSP regarding their SIP offerings and interoperability.
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The State of SIP and How it Enables Unified Communications
- 1. The ‘State’ of SIP Graham Francis, CEO The SIP School
- 2. What SIP is SIP and Interoperability SIP Trunking craziness SBCs and why? ITSP what’s on offer? Advice!
- 4. SIP SIP SIP SIP SIP SIP
- 5. Sessions Initiates Protocol
- 6. Session Initiation Protocol - INVITE 1002 1002 1003 1003 Call 1003 Want to talk? SIP SIP From: ‘Bob’ <sip:1002@sipbiz.com> To: <sip:1003@sipbiz.com> Hold Video Hold Video Hold Video SDP Hold Video 11 22 33 11 22 33 Audio Codec – G711 or … Audio Codec – G729 44 55 66 44 55 66 77 88 99 77 88 99 ** 00 ## ** 00 ##
- 7. Session Initiation Protocol – 200 OK 1002 1002 1003 1003 Call 1003 Want to talk? SIP OK SIP From: ‘Bob’ <sip:1002@sipbiz.com> To: <sip:1003@sipbiz.com> Hold Video Hold Video Hold Video SDP Hold Video 11 22 33 11 22 33 Audio Codec – G711 or … Audio Codec – G729 44 55 66 44 55 66 77 88 99 77 88 99 ** 00 ## SIP 200 OK ** 00 ## SIP From: ‘Bob’ <sip:1002@sipbiz.com> To: <sip:1003@sipbiz.com> SDP Let’s use G711…!
- 8. Session Initiation Protocol – ACK 1002 1002 1003 1003 CallCall On 1003 ACK SIP On Call with 1003 with 1002 SIP From: ‘Bob’ <sip:1002@sipbiz.com> To: <sip:1003@sipbiz.com> Hold Video Hold Video Hold Video SDP Hold Video 11 22 33 11 22 33 OK, G711 it is….! 44 55 66 44 55 66 77 88 99 77 88 99 ** 00 ## ** 00 ## Voice ‘Stream’
- 9. Session Initiation Protocol – Video? 1002 1002 1003 1003 On Call Support Video? SIP On Call with 1003 SIP with 1002 From: ‘Bob’ <sip:1002@sipbiz.com> To: <sip:1003@sipbiz.com> Hold Video SDP Hold Video Hold Video Hold Video Audio Codec – G711, G729 11 22 33 11 22 33 Video Codec – H.264,H263,H261 44 55 66 44 55 66 77 88 99 77 88 99 ** 00 ## SIP OK ** 00 ## SIP From: ‘Bob’ <sip:1002@sipbiz.com> To: <sip:1003@sipbiz.com> SDP Let’s use G711 and H.264 Voice and Video ‘Streams’
- 10. Enable UC
- 12. Request for Comments ~ RFC They’ll love this! Awesome! Individual Working Group
- 13. SDP SIMPLE INFO Load Balancing P2P SIP 3261 REFER SIP Routing With ENUM Overload RTP for Control SIPREC
- 14. The road to compatibility 344 344 Should 475 475 Can May 381 381 Option 144 144 3261
- 15. Cookie RFC Cookie RFC Flour Flour Bicarbonate of Bicarbonate of soda soda Salt Salt Butter Butter Caster sugar Caster sugar Soft brown Soft brown sugar sugar Vanilla extract Vanilla extract Eggs Eggs Dark chocolate Dark chocolate Raspberry Jam Raspberry Jam Coconut Coconut
- 19. 40% – 80% savings but not the only benefits!
- 20. Lines – You choose DDIs – How many? Virtual Presences Disaster Recovery Flexible Provisioning Branch 2 Branch $0?
- 21. 2018 PSTN
- 23. SESSION BORDER CONTROLLER
- 28. Measure = 10 Trunks please Voice = G.711 @ 90Kbps Pipe = 10 x 90Kbps = 900Kbps If using a 2Meg link That’s ~ 44% for VoIP and the rest for Data – Fully loaded…..!
- 29. Measure = 10 Trunks please Voice G.711 @ 90Kbps Video H.264 approx. 250Kbps (min) Pipe = 10 x 340Kbps = 3.4Mbps If using a 2Meg link, oops!
- 30. ITSP Supply this line? Degradation Multiple lines needed? Cable, Ethernet?
- 35. Let’s leave the video this Let’s leave the video this time time L7 SIP ~ 192.168.1.1 L7 SIP ~ 192.168.1.1 77.128.13.88 L3 IP ~ 192.168.1.1 L3 IP ~ 77.128.13.88 ?? ? L7 SIP ~ 192.168.1.1 77.128.13.88 L3 IP ~ 77.128.13.88
- 37. What? ITSP Network Your PBX REGISTER Client X SIP Registrar 10 Trunks Authentication G.711 Authorization Call Rates Encryption Voice – YES • Signalling Video -Traffic • Voice NO
- 38. SESSION BORDER CONTROLLER
- 39. ITSP ted Network Your PBX Encryp REGISTER SIP Registrar Secured G.711 G.711 G.711 to G.729 G.711 to G.729 G.729 G.729 SIP Signaling Media
- 40. So, do you have this device that will do it all for you?
- 41. Does the Provider tell you that you don’t need one?
- 43. PSTN ITSP Offerings We need have our ‘old’ numbers please? Can we some more DDI’s PBX Router That’s us No course! Of problem Can I have a ‘London’ number? PSTN
- 44. PSTN ITSP Offerings We need more lines more our marketing push! Thwarted PBX SHA //TLS //SRTP SHA TLS SRTP Router I’ll do that! I can do that via our Web Interface PSTN
- 45. Ask ITSP Interop Qs Ask PBX vendors RFC Compliance?
- 46. Rental Price per Trunk/Channel/Session Unlimited Trunks for a fee / month Call charges
- 48. Interesting but we’re not done % yet! 18% had no problems at all
- 51. NEW ONE PLEASE
- 52. SIP SSCA® SIP School Certified Associate SIP SSMP™ SIP Sales and Marketing Professional
- 54. The Telecommunications Industry Association (TIA), the leader in advocacy, standards development, business development and intelligence for the information and communications technology industry, has officially endorsed ‘The SIP School’ as the provider of choice for training and certification for Session Initiation Protocol (SIP).
- 56. SRTP JS JS Internet Web server
- 57. SRTP Signaling Real Time JS Media engine SIP Voice / Video Gateway Data Internet Internet Best Buy Web server
- 60. Thank you! discount Code ~ IT2013 graham@thesipschool.com http://ingate.thesipschool.com
Editor's Notes
- Hurricane Sandy and Verizon, not replacing Copper lines but just using as a chance to cut the cord.!!!!!! Comment on this Companies that rely on PSTN Gary Audin and Stuff that needs Analogue Fax!!! AT&T ask the Federal Communications Commission to create a timetable to shutdown the analog PSTN phone system in the United States. AT&T explains that “maintaining two networks - IP and PSTN is retarding the deployment of the newer broadband IP network”. Teliasonera – 2 years SO…. It’s happening then…. No going back…………………………… AT&T has asked the FCC to create a timetable that would allow the company to shut down its analog public switched telephone network (PSTN). AT&T said doing so is the only way to meet Congress' goal of covering all Americans with broadband services as more investment would flow to its IP-based initiatives. "That transition is under way already," AT&T wrote to the FCC. "With each passing day, more and more communications services migrate to broadband and IP-based services, leaving the public switched telephone network (PSTN) and plain-old telephone service (POTS) as relics of a bygone era.“ AT&T argues that having to maintain and invest in two networks--broadband and the PSTN--means Congress' goal won't be met efficiently or timely. The company said that while 90 percent of Americans have access to broadband services, reaching that last 10 percent would require an investment of about $350 billion. Meanwhile, AT&T continues to struggle with its own 3G network because of data hungry smartphone users. "Due to technological advances, changes in consumer preference and market forces, the question is when , not if , POTS service and the PSTN over which it is provided will become obsolete," AT&T wrote to the FCC. AT&T also said that less than 20 percent of Americans rely exclusively on POTS for voice service, while 25 percent of households have abandoned POTS. It noted that some 700,000 lines are being turned off each month. AT&T outlined steps for shutting down the PSTN and wants the FCC to swiftly follow them. AT&T's letter was in response to a public query from the FCC seeking comment on how the nation should migrate from circuit-switched networks to all-IP voice networks.
- Let’s do the maths….
- So if you are supplying a connection for your client, you need to ask these questions…