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

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

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 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:john.doe@company.com sip:4081234567@proxy1.company.com Default Port: SIPS URI 5061

SIP:

SIP:

SIP:

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

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

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