Voice over IP Fundamentals

1. CHAPTER 11 + 12 • H.323 • SIP Voice over IP Fundamentals

2. Trunking Connections Between Systems: • Common language must be used or conversion between languages • Available languages are H.323, Session Initiation protocol (SIP), Media Gateway Control protocol (MGCP), and Skinny Client Control Protocol (SCCP) • SCCP is Cisco proprietary

3. H.323: • International Telecommunications Union (ITU) accepted in 1996. • Designed to carry multimedia over Integrated Services Digital Network (ISDN) • Based or modeled on the Q.931 protocol • Cryptic messages based in binary • Designed as a peer-to-peer protocol so each station functions independently • More configuration tasks • Each gateway needs a full knowledge of the system • Can configure a single H.323 Gatekeeper that has all system information • Each end system can contact the gatekeeper before making a connection • Gatekeeper can perform Call Admission Control (CAC) to determine if resources are available before a call is accepted • Gatekeeper and Gateway can be the same device

4. H.323:

5. H.323: • System Control Unit: Provides call control, capabilities exchange messaging and signaling • Media Transmission: Formats transmitted audio, video, data control streams and messages • Audio Codec: Encodes the signal • Network Interface: A packet based interface capable of end-to-end Transmission Control Protocol and User Datagram Protocol for both unicast and multicast

6. H.323: • Video Codec: Capable of encoding and decoding video to H.261/H.263 standards • Data Channel: Supports applications such as database access

7. H.323:

8. H.323: • Gateway reflects the characteristics of a Switched Circuit Network.

9. H.323 Gatekeeper: • Address Translation: Provides endpoint IP addresses from H.323 aliases or E.164 addresses • Admissions Control: Provides authorized access to H.323 • Bandwidth Control: Manages endpoint bandwidth requirements • Zone Management: Provided for registered terminals, gateways and Multipoint Control Unit (MCUs). • Call Control Signaling: Uses gatekeeper routed call signaling (GKRCS)

10. H.323 Gatekeeper: • Call Authorization: Enables the gatekeeper to restrict access to certain terminals and gateways based on time-of-day • Bandwidth Management: Enables the gatekeeper to reject admission if required bandwidth is unavailable (Call Admission Control (CAC)) • Call Management: Provides services including an active call list

11. H.323 Protocol Suite:

12. H.323 RAS Signaling: • Gatekeeper Request (GRQ) • Gatekeeper Confirm (GCF) • Gatekeeper Reject (GRJ) • Registration Request (RRQ) • Registration Confirm (RCF) • Registration Reject (RRJ) • Unregister Request (URQ)

13. H.323 RAS Signaling: • Unregister Confirm (UCF) • Unregister Reject (URJ)

14. H.323 RAS Signaling:

15. H.323 RAS Signaling:

16. H.323 RAS Signaling:

19. SIP: • SIP was designed by the IETF as an alternative to H.323 • SIP is a single protocol whereas H.323 is a suite of protocols as FTP is a single protocol within the TCP/IP protocol suite • SIP is designed to set up connections between multimedia endpoints • Uses other protocols (UDP, RTP, TCP….) to transfer voice or video data • Messaging is in clear ASCII text • Vendors can create their own “add-ons” to the SIP protocol • SIP is still evolving • SIP is destined to become the only voice and video protocol

20. SIP Functionality: • User Location: Can discover the location of the end user. Supporting user mobility • User Capabilities: Will determine the media capabilities if the devices • User Availability: Determines the willingness of the end user to participate in a conversation • Session Setup: Enable the establishment of session parameters • Session Handling: Enables the modification, transfer and termination of a session

21. SIP Network Elements: • User Agent: Initiates or Responds to SIP transactions • User Agent Client: Initiates requests and accepts responses • User Agent Server: Accepts requests and returns responses • Proxy: Responsible for forwarding requests to the target • Redirect Server: Will direct other devices to a Uniform Resource Identifier (URI) • Registrar Server: Accepts messages to update the location database • Back-to-Back User Agent: Intermediate entity that processes requests

22. SIP Protocols: • Real-time Transport Protocol • RSVP • TLS: Privacy and Integrity • STUN: Used with NAT

23. SIP Addressing: • E-Mail type: • sip:user@domain:port • sip:user@host:port • sip:john.doe@company.com • sip:4081234567@proxy1.company.com • Default Port: • SIPS URI 5061

25. SIP:

26. SIP:

27. SIP:

28. MGCP: • IETF standard with developmental aid from Cisco • All devices under a central control • Voice gateway becomes a dumb terminal • Allows minimal local configuration • Single point of failure • Uses UDP port 2427

30. SCCP: • Often called “skinny” protocol • Cisco proprietary • Similar to MGCP in that it is a stimulus/response protocol • Allows Cisco to deploy new features in their phones • Cisco phones must work with Cisco systems (CME, CUCM,CUCME…) • Cisco phones can also use other protocols such as SIP or MGCP with downloaded firmware

31. Internet Telephone Service Providers: • New service providers that provide phone services over the internet (Vonage) • They interface with the traditional phone service providers (PSTN) • Bundle voice and data together

32. End of Chapter 11 +12