VOIP ch 12

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  1. signaling protocol that controls the initiation, modification, and termination of interactive multimedia sessions
    text-based protocol
    peer-to-peer protocol,
    Session Initiation Protocol (SIP)
  2. call routing and session management functions are distributed across all the nodes (including endpoints
    and network servers)
    Session Initiation Protocol (SIP)
  3. In June 2002, the IETF published a new SIP RFC
    RFC 3261
  4. provides the capability to discover the location of the end user for the purpose of establishing a session or delivering a SIP request
    user location
  5. enables the determination of the media capabilities of the devices that are involved in the session.
    User Capabilities
  6. willingness of the end user to engage in communication.
    User Availability
  7. enables the establishment of session parameters for the parties who are involved in the session.
    session setup
  8. enables the modification, transfer, and termination of an active session.
    session handeling
  9. is responsible for forwarding SIP requests to the target UAS
    provides routing infromation
    enforce security policy
  10. for reserving network resources
    Resource Reservation Protocol (RSVP)
  11. for gateway selection and load balancing
    Telephony Routing over IP (TRIP)
  12. for hostname-to-IP address resolution
    Domain Name System (DNS)
  13. for preventing eavesdropping, tampering, or message forgery.
    Transport Layer Security (TLS)
  14. to resolve host or domain names into routable IP addresses
    used to load-share across multiple servers in a cluster identified by a hostname
  15. to describe the parameters of the multimedia session. This information includes session type such as audio, video, or both and parameters such as codecs or ports needed to establish a media stream.
    Session Description Protocol (SDP)
  16. transports real-time data such as audio or video packets to the endpoints that are involved in a session
    Realtime Transport Protocol (RTP)
  17. encryption technology can cause what
  18. to reserve network resources such as bandwidth prior to establishment of the media session. This ensures that the network resources are in place prior to the called party
  19. to provide privacy and integrity of SIP signaling information over the network
  20. protocol to discover the presence and type of Network Address Translation (NAT) between them and the
    public Internet.
  21. signaling is independent of the type of session being established
  22. can send calls to the pstn
    sip gateway
  23. in sip network the telephone picking up the call is the
  24. 5060
    sip default port
  25. is typically an e-mail-type address with a format such as one of the following: sip:user@domain:port sip:user@host:port
  26. identify a user or a resource within a network domain
    SIP address
  27. are messages that are sent from client to server to invoke a SIP operation
    SIP Request
  28. indicates that the recipient user or service is invited to participate in a session
    "placing a call"
  29. that the UAC has received the final response to an INVITE request.
  30. to query a UAS about its capabilities
  31. **request the termination of a previously established session**
  32. *hang up a call be befor the user picked up *
  33. email address be corsponed to an ip address
  34. sent from the server to client
    1xx provisional status
    numbered from 100 to 699
    SIP Response Messages
  35. establishment of the session
    defined as a peer-to-peer
    SIP relationship between two or more UAs that persists for the duration of the session
  36. uses TCP, Stream Control Transmission Protocol (SCTP) and UDP
    SIP transport layer
  37. the SIP RFC 3261 encrypt the signaling information what 2
    TLS IPsec
  38. my drop out or stay in the call while it take place
    SIP proxy server
  39. proxy sends a message back to the user say that the user has moved
  40. starts at the server not the phone
    only involved in call set up
    can initiate new SIP calls and modify and terminate existing calls.
    creation of two distinct dialogs, which enable it to modify one SIP session without affecting the other
    act as a third-party call controller (3PCC) and can establish calls between two user agents.
    B2BUA(Back-to-back user agent)
  41. To determine the address of SIP servers that support UDP transport in company.com domain, you need to
    query a What ? with the DNS server
    query string
  42. to determine what services are supported in a domain
    Naming Authority Pointer (NAPTR)
  43. the services and protocols that are supported within the domain.
    Service Record (SRV)
  44. registrar and proxy are functions implemented with the
    same server
  45. sip proxies router SIP request to the server and sip resoponces to the
  46. client insists that the server must understnd the SIP extension to process a request
    Require header
  47. indication to the UAS(server) that the UAC(client) understands a certain extension
    supported header
  48. SIP extension that enables a caller to express his preferences about request handling at the intermediate
    Caller and Callee Preferences
  49. which SIP nodes can request notification from remote peers when monitored events take place
    (voice mail messafe changing online status)
  50. is waiting for the end-user input to determine whether to accept a subscription from this subscriber.
    Subscription-State value is pending in the NOTIFY request
  51. near real time mesafes but the person must be willing to engage in the session
    Instant Message (IM)
  52. enables users to publish their availability status and display messages or icons as a form of self-expression
  53. MESSAGE requests carry the message content in the form of
  54. an IETF signaling protocol for multimedia applications involving one or more participants
  55. a flexible protocol that supports extensions for new applications and services
  56. SIP dialog state is maintained at the
  57. SIP network servers are either stateless or maintain transaction state information for at least
    32 sec
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VOIP ch 12
2011-07-27 02:43:50

VOIP ch 12
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