SIP A Technology Deep Dive Anshu Prasad Product Line Manager, Mitel June 2010 Laith Zalzalah Director, Mitel NetSolutions
What is SIP? Session Initiation Protocol (SIP) is a signaling protocol for establishing sessions We most commonly think of Voice over IP (VoIP), but applies to multimedia/video, instant messaging, faxing, web integration and more Devices SIP PBX Standardized by the Internet Engineering Task Force (IETF) in Request for Comment (RFC) documents The model is lightweight the standard focuses on initiating, modifying and terminating sessions The model is decentralized its up to the parties establishing the session to negotiate the attributes of the session The power of SIP is that it is Flexible, Extensible and Open SIP Service Providers Applications slide 3
SIP Versus Other VoIP Protocols Open / Flexible Feature Rich Interoperable SIP H.323 MGCP / Megaco Proprietary slide 4
Components of a SIP System Logically, there are several parts to a SIP system User Agent (UA) the phone or server Registrar keeps track of where the user is within a system Redirect Server used to inform devices when they need to contact different locations Proxy Server used to relay messages back and forth within the system In practice, several functions may actually be in the same server slide 5
SIP Walkthrough Registration I m Jane, and I ll be using a phone at 192.168.0.1 today! When I get a call for Jane, I ll know to contact her at 192.168.0.1! 192.168.0.1 SIP Server Lunch time! Use my mobile at 10.0.1.1 sip.telco.com Ah, now I will have to use 10.0.1.1 to reach Jane. 10.0.1.1 slide 6
SIP Walkthrough Proxy I need to call Jim! Jim is over at shopmart.com. I ll proxy the call over there! sip.telco.com Call here for Jim from Jane Call for you from Jane! sip.shopmart.com slide 7
SIP Walkthrough Redirection I need to call Jim! sip.telco.com Call here for Jim from Jane sip.shopmart.com Try him at newplace.com Call here for Jim from Jane sip.newplace.com Call for you from Jane! slide 8
SIP Protocol SIP is much like the HTTP request/response model Uses a text-based message containing a header and message body Invite 100 Trying 180 Ringing 200 OK ACK Invite 180 Ringing 200 OK ACK Media RTP Media BYE BYE 200 OK 200 OK Signaling Signaling slide 9
SIP Standards (RFCs) RFC 3261 SIP: Session Initiation Protocol SIP utilizes other protocols: TCP or UDP for connection, Real-time Transport Protocol (RTP) to carry media, Session Description Protocol (SDP) to define parameters of the RTP, etc. RFC 1321 The MD-5 Message Digest Algorithm RFC 2617 HTTP Authentication : Basic and Digest Authentication RFC 2782 A DNS RR for specifying the location of services (DNS SRV) RFC 3551 RTP Profile for Audio and Video Conferences with Minimal Control RFC 3665 Session Initiation Protocol Basic Call Flow Examples RFC 3725 Best Current Practices for 3rd Party Call Control RFC 2976 The SIP INFO Method RFC 3262 Reliability of Provisional Responses in SIP RFC 3263 Locating SIP Servers RFC 3264 An Offer/Answer Model with SDP RFC 3265 SIP-Specific Event Notification RFC 3311 The Session Initiation Protocol UPDATE Method RFC 3325 Asserted Identity within Trusted Networks RFC 3329 Security Mechanism Agreement RFC 3515 The Session Initiation Protocol (SIP) Refer Method RFC 3550 RTP: A Transport Protocol for Real-Time Applications RFC 3824 Using E.164 numbers with the Session Initiation Protocol (SIP) RFC 3842 A Message Summary and Message Waiting Indication Event Package for the Session Initiation Protocol (SIP) RFC 3891 The Session Initiation Protocol (SIP) 'Replaces' Header RFC 3892 The SIP Referred-By Mechanism RFC 3966 The Tel URI for Telephone Numbers RFC 4028 Session Timers in the Session Initiation RFC 4566 SDP: Session Description Protocol RFC 4733 RTP Payload for DTMF Digits, Telephony Tones and Telephony Signals slide 10
SIP Trunking, Stations and Applications SIP trunking to a service provider network By being virtual trunks, cost is significantly reduced compared to legacy physical digital or analog trunks Network simplification is achieved by combining voice and data network services on IP SIP line/station support Avoids vendor lock-in; customer can pick and choose the devices they want Functionality specific endpoints can be more easily integrated conference phones, video phones, mobile or desktop softphones, etc. SIP applications No need for proprietary API s or custom development to achieve business process integration with telephony systems Vendor benefits by developing only a single implementation Customer benefits by breadth of choice SIP Trunk enterprise VoIP worldwide growth forecast* 2009 2013 8.5M 24.3M trunks trunks 2009 2016 $717M $3.9B 2009 2016 3.8M 46M users users slide 11 * Data from Infonetics Research and Frost & Sullivan
SIP Trunking Deployment Example Enterprise Service Provider SIP Registrar SIP B2BUA PSTN SIP Signaling PSTN GW LAN RTP Media SIP Phones SIP-Aware Firewall Broadband Link LAN SIP Session Border Controller SoftSwitch slide 12
The Promise versus the Reality of SIP Promise Standards-based Vendor interoperability Application integration Reduces costs Reality Numerous SIP RFCs still evolving Most are open to interpretation Some contain ambiguous or optional provisions Successful interoperability can t be assumed just because its SIP Testing is required in every case as implementations continue to evolve Application developers are adopting SIP as their only telephony integration interface Proprietary integrations still exist in many cases Cost savings are significant as long as interop issues are avoided through careful testing prior to deployment in a production environment slide 13
Addressing the Challenges of SIP Mitel SIP Center of Excellence Validation of SIP interoperation with 3 rd party network devices, endpoints, applications and service provider networks Dedicated lab capable of executing on-site or remote SIP interop tests Standardized test plans for both trunk and line testing Test execution of 3 rd party test plans for 3 rd party certification Detailed test results and configuration documentation produced for every interop Aligned and integrated with the Mitel Solutions Alliance partner ecosystem program slide 14
Mitel SIP Qualified Interops slide 15
Addressing the Challenges of SIP Example - Managed Services Current CLEC Certifications Pending CLEC Certifications SIP Trunking Local Long Distance Wireless Data Fully integrated and optimized for Mitel s CPE portfolio Engineered to replicate PSTN feature set while providing the benefits of IP-based SIP trunking More efficiently allocate and utilize bandwidth Centralize call control across geographic boundaries Provides address-based E911 coverage for every registered site to their local Public Safety Responders Improves survivability with business continuity routing Enables cost containment by centralizing network support and infrastructure NetSolutions and the Mitel Communications Platforms Single Point of Accountability Managed Services Offering slide 16
Successfully Deploying SIP Top 10 Tips 1. Ensure your network infrastructure is capable of supporting VoIP Voice quality issues, connectivity failure, etc. are often due to the underlying network 2. Choose your solution provider carefully Need to be SIP knowledgeable in order to diagnose and debug issues efficiently 3. Confirm interop compatibility Version matching is important as SIP implementations change regularly 4. Understand your feature requirements SIP capabilities vary, so its critical to map your requirements against the features supported 5. Review the interop documentation Confirm known limitations won t affect your deployment 6. For SIP trunking, ensure you have a qualified SIP-aware firewall Most SIP trunking issues start and end here 7. For SIP devices, consider education and training for your users Keep in mind that SIP devices typically behave differently than legacy phones 8. Evaluate scalability and manageability of the solution Deploying a single device is very different from deploying 500 or 1000 9. Do a trial deployment prior to cutover Test important functionality is working prior to mass deployment 10. Don t assume too much Interop testing is limited; just because two things interop doesn t mean they ll work the way you need them to slide 17
Conclusion SIP is the present and future of IP-based communications Vendor implementations are maturing and normalizing SIP trunking is very common and customers are reaping significant benefit today SIP device and application integration continues to improve rapidly Mitel is fully committed to SIP Devices All Mitel PBX platforms offer SIP support and are extensively tested with 3 rd party implementations The Mitel Border Gateway product provides SIP-aware firewall functionality in addition to other advanced services SIP PBX Mitel NetSolutions offers fully managed communications services including SIP SIP Service Providers Applications slide 18 18
Thank you Anshu Prasad Mitel Product Line Manager anshu_prasad@mitel.com (613) 592-2122 ext 2263