1. By Stephen Tomko H.323 vs. SIP

2. Internal PBX Call Extension number is dialed PBX receives extension Routes extension Routes call to the phone Call begins

3. Internal VoIP Call Extension number is dialed Server picks up extension Tells phone how to reach the other phone via IP VoIP phone establishes a connection to the other phone Via TCP/IP protocol stack Uses network mediums Both confirm connection and call begins

4. Difference? Extension is still dialed Call is still placed Still routed Call quality is equal

5. Convergence Data, voice, and video are all combined under one medium Many reliable mechanisms to provide reliability, security, and manageability already exist TCP, RTP, QoS params SSL, SSH, S-HTTP

6. How we handle calls VoIP, Video, and data must have sessions Sessions need protocols Think SS7 for landline Two largest protocols SIP H.323

7. SIP Background Created by the Internet Engineering Task Force (IETF) as a method to control “sessions” between one or many points February 1996 Designed around the HTTP protocol Session Codes E.g. 400 – user error, 500 – server error, etc…

8. H.323 Background Created by ITU-T as a method to control voice, audio, and data November 1996 A Suite of protocols H.225 H.235 H.245 Derived from Q.931 standard Think PSTN and ISDN signaling

10. SIP Components User Agent Client Creates and sends requests A SIP compatible phone is a User Agent doing the work of a User Agent Client Registar/Location Server Registers User Agents Stores locations Address resolution (SIP:user2209@statefarmIT)SIP:user2209@statefarmIT User Agent Server Accepts, Forwards and routes calls Proxy Server Routes calls Redirect Server Multifunctioned Discussed Later

11. H.323 Components Terminal Standard VoIP phone or any device that starts or terminates H.323 sessions Gateway Translates data from one incompatible network to another Think PSTN to Ethernet Gatekeeper Controls calls and sessions Address resolution Zone control Bandwidth control Multipoint Control Unit Bridges many sessions into one

12. pulver.com Proxy server nortel.com user@nortel.com 1. INVITE sip:jeff.pulver@pulver.com SIP/2.0 From: sip:user@nortel.com Location Server jeff.pulver pulver@von1 2. INVITE sip:pulver@von1 SIP/2.0 From: sip:user@nortel.com 3. SIP/2.0 200 ok From: sip:pulver@von1 pulver@von1 4. SIP/2.0 100 OK From: sip:jeff.pulver@pulver.com 5. ACK sip:jeff.pulver@pulver.com SIP/2.0 From: sip:user@nortel.com 6. ACK sip:pulver@von1 SIP/2.0 From: sip:user@nortel.com

13. 1. INVITE sip:jeff.pulver@pulver.com From: sip:user@nortel.com 2. SIP/2.0 320 Moved temporarily Contact: sip:pulver@von1.pulver.com nortel.com user@nortel.com pulver.com Redirect Server Location Server Jeff.pulver pulver@von1 4. INVITE sip:pulver@von1.pulver.com From: user@nortel.com 3. ACK sip:jeff.pulver@pulverr.com From: sip:user@nortel.com 5. SIP/2.0 200 OK To: user@nortel.com 6. ACK sip:pulver@von1.pulver.com From: sip:user@nortel.com Pulver@von1

15. H.323 Process Extensive H.225 – Call signaling Provides call signaling, alerting, and connected statuses for the call in question RAS Signaling Communication between terminals and gatekeepers Communication between gatekeepers H.245 Capability negotiation Defines what codecs to be used for audio/video/data H.264, G.729, T.140 Master/Slave determinator Settles all disputes between two devices during negotiation H.235 Encryption using SSL

16. H.323 vs. SIP http://www.microtronix.ca/sip_vs_h323.htm

17. Business Value Converging data/voice/video under one medium - network Control how your business communicates Multipoint conferencing Unified communication allows all types of communication under one protocol All in one package

18. Why technology is important One word: Convergence umbrella Geographic locations are reduced H.323 All-in-one suite, managers many facets SIP Simplicity, small, modularity

19. Major Commercial products Nortel’s Application Server 5300 for SIP http://products.nortel.com/go/product_content.jsp?segId=0 &catId=null&parId=0&prod_id=66621&locale=en-US http://products.nortel.com/go/product_content.jsp?segId=0 &catId=null&parId=0&prod_id=66621&locale=en-US Avaya Aura for SIP http://www.avaya.com/usa/product/avaya- aura#%20Avaya%20Aura%E2%84%A2%20Communication% 20Manager http://www.avaya.com/usa/product/avaya- aura#%20Avaya%20Aura%E2%84%A2%20Communication% 20Manager Cisco UT Communications Manager (CallManager) for H.323/SIP http://www.cisco.com/en/US/products/sw/voicesw/ps556/