Download Chapter XI & XII Slides

VOICE OVER IP FUNDAMENTALS
•
•
•
CHAPTER 11 + 12
H.323
SIP
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
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
H.323:
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
H.323:
• Video Codec: Capable of encoding and decoding
video to H.261/H.263 standards
•Data Channel: Supports applications such as database
access
H.323:
H.323:
• Gateway reflects the characteristics of a Switched
Circuit Network.
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)
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
H.323 Protocol Suite:
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)
H.323 RAS Signaling:
• Unregister Confirm (UCF)
• Unregister Reject (URJ)
H.323 RAS Signaling:
H.323 RAS Signaling:
H.323 RAS Signaling:
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
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
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
SIP Protocols:
• Real-time Transport Protocol
• RSVP
• TLS: Privacy and Integrity
• STUN: Used with NAT
SIP Addressing:
• E-Mail type:
• sip:user@domain:port
• sip:user@host:port
• sip:[email protected]
• sip:[email protected]
• Default Port:
• SIPS URI 5061
SIP:
SIP:
SIP:
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
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
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
End of Chapter 11 +12