1. Voice over IP Fundamentals
CHAPTER
H.323
SIP

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. International Telecommunications Union (ITU) accepted in 1996.
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 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
Default Port:
SIPS URI 5061

25. SIP:

26. SIP:

27. SIP:

28. IETF standard with developmental aid from Cisco
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. Often called “skinny” protocol
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