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8-10. Application Layer (Domain Name System (DNS) (Name Resolution (Uses…

- - - - Single point of failure: internet won't work when there is an attack
      - Traffic volume: not scalable, hard to maintain
      - Distant centralized database: affects latency for those far away
      - Difficult to maintain
  - - - Busy sites replicated over multiple servers
      - Set of IP addresses is associated with one canonical name
      - DNS server rotates the order of the addresses to distribute the load
  - - - Case insensitive
      - Max 63 characters per constituent
      - Max 255 chars per path
      - Internationalised
  - - - Iterated query: contacted server replies with name of server to contact
      - Recursive query: server obtains mapping on client's behalf
    - - IP addresses of TLD servers typically cashed in local name servers
      - Root servers not often visited
      - Cached entries timeout after some time
      - ipconfig / display dns : cached records on local computer
      - nslookup <<addresss>: IP v6 address of a hostname
  - - - BIND
      - djbdns
      - Microsoft Domain Name Server
    - - nslookup
      - dig
- - - - Compose
      - Report
      - Display
      - Dispose
    - - Destination address
      - Priority
      - Security level
      - Used by mail serves for routing
  - - - Languages with accents (French, Spanish)
      - Non-Latin alphabets (Cyrillic)
      - Non-alphabetic language (Chinese, Japanese)
      - Messages with content other than text (audio, images)
    - - Leverage existing infrastructure for RFC 822 services
      - Leave MIME functionality changes to the user agent
- - - - Steps
        
        1a. Initiate TCP connection to HTTP server at port 80
        
        1b. HTTP server at host (main URL) waiting for TCP connection at port 80; accepts connection and notifies client
        
        HTTP client sends a HTTP request message containing URL path into TCP connection socket (indicate it wants object)
        
        HTTP server receives request message, forms response message containing request object -> sends msg into its socket
        
        HTTP server closes TCP connection
        
        Client parses HTML file, finds other referenced objects, repeat steps 1-6 for each object
      - Response time (diagram)
        
        Round trip time (RTT): each object needs two RTT (initiate TCP connection + receive file)
        
        Total Response time = 2 * RTT + file transmission time
      - Issues
        
        Requires new connection per requested object
        
        OS overhead for each TCP connection
        
        Delivery delay of 2 RTTs per requested object
    - - Server leaves connection open after sending response
      - Pipelining: client sends request as soon as it encounters a referenced object -> only 1 RTT for all referenced objects -> slightly more than 3 RTT in total for response time
      - Server closes connection if it hasn't been used for some time
      - Now the default connection type (> HTTP 1.0)
    - - Request and Response message example
      - HTTP Request methods
      - HTTP error codes
  - - - Domain
      - Path
      - Content
      - Expiry
      - Security
    - - Initial HTTP request made from a cookie-enabled computer -> site creates unique ID + entry in backend DB
      - Includes cookie ID in the HTTP response
      - Browser appends the cookie ID to the cookie file that it manages
      - Each subsequent request, cookie header line will be added
      - Server will perform a cookie-specific action
    - - Authorization (e.g. stay logged in)
      - Shopping carts
      - Recommendations
      - User session state (e.g. keeps your drafted web email)
  - - - Reduce response time for client request
      - Reduce traffic on an institution's access link
- - - - Higher bandwidth
      - Higher QoS (delay sensitive)
      - Need separate multimedia servers from web servers
      - Streaming multimedia service providers often separated and highly specialised compared to traditional web hosts
    - - Entire media transmitted first before playback starts -> delay
      - Assumes point-to-point media distribution rather than point-to-multipoint (broadcast) distribution model
      - Does not scale
      - Assume browser/plugin/helper integration and traditional service types
    - - Transport Protocols
        
        TCP: on-demand video download partically before playing
        
        UDP: live streaming
      - Open protocols
        
        HTTP (on top of TCP)
        
        Real Time Transport Protocol RTP (on top of UDP)
        
        Real time Streaming Protocol RTSP (on top of TCP) for control i.e. fast-forward/pause
        
        RTSP commands are sent from player to server to adjust streaming (diagram)
        
        Data stream
        
        RTP over UDP
        
        RTP over HTTP over TCP
      - Closed protocols
        
        Real Networks' RealAudio
        
        Microsoft's Windows Media
        
        Apple's QuickTime
    - - Manage user interface (volume, playback, next etv.) (diagram) : metafile response from the web server contains how media player can contact the media server
      - Handle transmission errors in conjunction with transport protocols (if RTP/UDP used, errors more likely, playback software must manage them gracefully)
      - Decompress multimedia files
      - Eliminate jitter
        
        Small buffer -> quick playback but more susceptible to jitter/delay
        
        Large buffer -> delay at start of playback while buffer fills -> less jitter/delay
    - - Use reliable transport (TCP)
        
        Repairs all errors
        
        Increase jitter significantly
      - Add Forward Error Correction (FEC) - add a parity packet for every 4 packets (diagram)
        
        Repairs most errors - can repair one lost packet in a group of N
        
        Increases overhead, decoding complexity and jitter; decoding delayed for N packets
      - Interleave media (diagram) - puts all odd frames in one packet, all even frames in one packet
        
        Masks most errors
        
        Slightly increases decoding complexity and jitter
    - - Ideal: stream buffer continue to be filled at the same rate the stream is played back to the user
      - Two extreme cases
        
        Data sent by media server is faster than media player -> buffer overflow; lost data
        
        Media server sending data slower than the media player -> users have to wait
      - High water mark: media player tell media server to stop sending
      - Low water mark: media player tells media server to start sending again
      - Bandwidth delay product = RTT * transmission rate
    - - Frequency masking: some sounds can mask others; e.g. loud sounds mask soft sounds
      - Temporal masking: human ears miss soft sounds immediately after loud sounds; decrease amount of data to encode
    - - Compression with JPEG
        
        Lossy compression sequence: input and output have the same resolution
        
        Compression ratios of 20:1
        
        Symmetric; decoding takes as long as encoding
      - Motion Picture Experts Group (MPEG)
        
        Compress both audio and video together using synchronised streams
        
        MPEG-1: VCR quality (40:1)
        
        MPEG-2: Broadcast quality (200:1)
        
        MPEG-4: DVD quality (1200:1)
  - - - Data has overtaken voice as the primary traffic on many networks originally built for voice
      - PSTN infrastructure is not flexible enough for the rapid deployment of new features
      - Public Switch Telephone Network (PSTN) technologies are largely incompatible with the convergence of data/voice/video
      - Network providers are increasingly looking to leverage investment in network infrastructure by bringing new services to data networks
    - - Financial savings
      - Consolidated infrastrcuture
      - Flexible infrastructure
      - Standards based voice and data
    - - 3 models
        
        Infrastructural: PSTN/PABX (Private automatic branch exchange) integration
        
        Virtual: Media gateways, virtual directories
        
        Value-added: voice mail
      - 3 standards
        
        H.323 STANDARD (diagram)
        
        Supports calls between Internet computers and PSTN phones
        
        Gateway exists to ensure call limit is not exceeded
        
        Standard that specifies how multimedia traffic is carried over packet networks
        
        Integrates existing standards to provide a layered protocol stack (diagram)
        
        Developed to enable multimedia applications to run over unreliable data networks
        
        SGCP (simple Gateway Control Protocol) + MGCP
        
        Session Initiation Protocol (SIP)
        
        Used for setting up the connection
        
        Single network module -> need other protocols like RTP/RTCP for data transport
        
        An application layer protocol; runs over UDP/TCP
        
        Functionality
        
        Two party / multiparty / multicase
        
        Callee location, callee capabilities , call setup, call termination
        
        Addressing
        
        Based on a URL type schema
        
        Can contain either IPv4 / IPv5 or actual telephone addresses
        
        Methods (diagram)
    - - UnPABX: server based call routing, with direct connections to data network , trunk telephony and analogue telephony networks -> cost less than the systems they replace
      - LAN-PABX: telephony function is in software on the desktop -> greater flexibility for users
      - Both leverage existing infrastructure: UnPABX convergence of communications infrastructure; LAN-PABX: wired/wireless connections to the workstation