Transport Layer

ccnp-routing
Notes on transport layer services, the TCP and UDP headers, flow and congestion control, and real time media delivery with RTP and RTCP.
Published

Dec 5, 2017

Transport Services

  • Transport Services vs Data Link Services
    • Similarities:
      • Both provide point-to-point connection
        • point-to-point means one path to the destination. When a packet goes in one end, it must come out the other end!
      • Both have to deal with error control, sequencing, flow control, retransmission, etc
    • Differences:
      • Transport connection is indirect
      • Network has memory: packets may be stored and arrive with varying delays and out of order
        • In Data link, when we send the first and second frame to PHY, we receive them in order. But in transport layer, when sending segments maybe we receive the second segment prior to the first segment
      • Many connections have to be managed (instead of a fixed number of links)
  • Primitives for a simple transport service:
    • LISTEN: block until some process connects
    • CONNECT
    • SEND
    • RECEIVE: block until data TPDU arrives
    • DISCONNECT
  • TCP addressing:
    • IP protocol number, TCP/UDP port number

Transmission Control Protocol (TCP)

  • TCP Provides a logical full duplex connection between two application layer processes across an unreliable datagram network (IP network)
  • TCP Provides flow control using Selective Repeat
  • TCP Can support multiple connections at the same time
  • Each direction of the connection can be terminated independently
  • TCP Service is:
    • Connection oriented
    • Reliable
    • In-sequence
    • Byte-stream oriented
  • TCP Service Model
    • TCP connection is:
      • Full-duplex, point-to-point (No support for multicast or broadcast)
      • Byte stream : message boundaries are not preserved (Application layer should parse and detect messages)
  • Both sender and receiver have to create sockets
    • socket # = IP address + Port# (16-bit, = TSAP)
  • Connection is identified by: (socket 1, socket 2)
  • Port# < 1024 : well-known ports
  • FTP: 21, Telnet:23, SMTP: 25, HTTP: 80
  • TCP may buffer at both sides; consequently transmission may be delayed
  • TCP should handle lost, out of order, and delayed segments with non-aligned boundaries
  • TCP Windowing
    • Host cannot process packets as quickly as it is receiving them
    • Unprocessed packets are stored in a receive buffer
    • When the receive buffer is full, subsequent packets get tail-dropped
  • Former Maximum Window size supported by TCP was 65,535 bytes
    • The TCP window scale option is an option to increase the receive window size allowed in TCP (RFC 1323)
      • Multiples the Window Size
      • Maximum effective window size is \(2^{30}\) or about 1 gigabyte
      • R1(config)#ip tcp window-size [65536-1073741823]
      • TCP Window scaling must be enabled on both hosts
  • TCP Global synchronization
    • Occurs when multiple connections experienced packet loss
    • Network failure
    • Packet getting tail-dropped
  • Every byte has 32-bits sequence number
    • On 10 Mbps it takes about an hour to wrap around
    • Separate 32-bit sequence numbers are used for acknowledgement and for the window mechanism
  • Data exchange in segments
    • Segment contains 20-byte header, options, + data
    • Variable payload size (the size is decided by the TCP software and is usually 1500 bytes)
    • Must fit into MTU: maximum transfer unit size of a network
      • Ethernet MTU is the largest payload that can be carried in a single frame, not including Ethernet headers. A common default MTU is 1500 bytes
    • Path MTU Discovery (PMTUD):
      • Suppose (R1) ---MTU=1500Bytes--- (R2) ---MTU=1280Bytes--- (R3)
      • Suppose R1 wants to send a packet to R3. R2 realizes that the packet is too big to send it to R3.
      • Then R2 splits up the packet (to 220 B + 1280 B fragment) and sends it to R3?
        • More work for R2
        • IPv6 packets cannot be fragmented by transit routers as the minimum MTU for IPv6 is 1280 Bytes
          • R2 sends ICMPv6 type 2 message telling to R1 “Use MTU 1280”
  • Sliding window protocol with timeout
    • Receiver sends back ack# equal to next expected segment#
    • Receiver uses piggybacking. Meaning that I send you something, you want to send me something. Okay you can give me the ack of the segment I had sent you in the header of the TCP you want to send to me
    • Sender retransmits if timeout occurs

TCP Congestion Control

  • TCP controls the data flow between two endpoints of a connection
  • Selective repeat sliding window is used for flow control
  • Receiver specifies its reception capacity called “Advertised Window” in the header of the messages that it sends back to transmitter. This handles the end-to-end flow control
  • Network congestion can also be controlled by slowing down the flow of data that TCP generates into the network. This requires feedback from the network behavior
  • The maximum amount of bytes that a TCP sender can transmit without congesting the network is called “Congestion Window”
  • TCP works by adapting these two windows to control the traffic generation process
  • Sender can transmit up to the minimum of “Advertised Window” and “Congestion Window”. This is called “Current Window”
  • Timer management:
    • Round trip delay of each segment (Transmission to ack) is measured.
    • Appropriate time is chosen based on history and variation of this delay
    • If timeout happens, it assumes that congestion happened then it reduces the transmission rate
  • One method is slow start mode:
    • In slow start mode, the congestion window size increases by one Maximum Segment Size (MSS) per received ack which results in its doubling once every RTT. (MSS indicates the maximum number of data bytes any particular layer-2 technology allows per packet. TCP/IP headers are later added to the packet (normally 40 bytes) and the resulting total is called the Maximum Transmission Unit (MTU))
    • MSS = MTU - TCP/IP headers
      • Generally in Ethernet we have 1500 Bytes MTU. TCP/IP headers are 40 bytes
      • MSS = 1460 Bytes
  • Another method is congestion avoidance:
    • In the congestion avoidance mode the window size increases by one MSS every RTT. When a packet is lost, the window size is halved. TCP thus uses an additive increase, multiplicative decrease algorithm for congestion control

TCP Header Format


   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |          Source Port          |       Destination Port        |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                        Sequence Number                        |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                    Acknowledgment Number                      |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |  Data |           |U|A|P|R|S|F|                               |
   | Offset| Reserved  |R|C|S|S|Y|I|            Window             |
   |       |           |G|K|H|T|N|N|                               |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |           Checksum            |         Urgent Pointer        |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                    Options                    |    Padding    |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                             data                              |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
  • Source Port: 16 bits
    • The source port number
  • Destination Port: 16 bits
    • The destination port number
  • Sequence Number: 32 bits
    • The sequence number of the first data octet in this segment (except when SYN is present). If SYN is present the sequence number is the initial sequence number (ISN) and the first data octet is ISN+1
  • Acknowledgment Number: 32 bits
    • If the ACK control bit is set, this tells the sender I received until byte n you had sent to me
  • Data Offset: 4 bits
    • The length of the header
  • Reserved: 6 bits
    • Reserved for future use. Must be zero.
  • Control Bits: 6 bits (from left to right):
    • URG: The URG flag is used to inform a receiving station that certain data within a segment is urgent and should be prioritized. If the URG flag is set, the receiving station evaluates the urgent pointer, a 16-bit field in the TCP header. This pointer indicates how much of the data in the segment, counting from the first byte, is urgent. The URG flag is not employed much by modern protocols, but we can see an example of it in the Telnet packet capture
    • ACK: Acknowledges received data
    • PSH: The socket that TCP makes available at the session level can be written to by the application with the option of “pushing” data out immediately, rather than waiting for additional data to enter the buffer
    • RST: Aborts a connection in response to an error
    • SYN: Initiates 3-way handshake and Sequence Number initiation
    • FIN: No more data from sender. Closes a connection
  • Window: 16 bits
    • The number of data octets beginning with the one indicated in the acknowledgment field which the sender of this segment is willing to accept.
  • Checksum: 16 bits
    • Used for error detection

User Datagram Protocol

  • UDP is an unreliable, connectionless transport protocol
  • UDP Services:
    • Routes the received packet to the desired application on the host (Destination Port)
    • Checks the integrity of the datagram. This is optional! (UDP Checksum)
  • If a host does not wish to calculate the checksum, it sets it to all 0’s.
  • Applications that use UDP: Domain Name Services (DNS), Simple Network Management Protocol (SNMP), Real Time Protocol (RTP)
  • UDP checksum calculation: Similar to TCP (pad to 16, pseudoheader for IP verification)

UDP Header Format

      0      7 8     15 16    23 24    31
     +--------+--------+--------+--------+
     |     Source      |   Destination   |
     |      Port       |      Port       |
     +--------+--------+--------+--------+
     |                 |                 |
     |     Length      |    Checksum     |
     +--------+--------+--------+--------+
     |
     |          data octets ...
     +---------------- ...

TCP Starvation/UDP Dominance

  • UDP is unaware of bandwidth, delay, and packet loss. What if it consumes all bandwidth of the link?
  • Consequence for TCP:
    • Starvation
    • Higher latency
    • Lower throughput
  • Solution:
    • Place TCP and UDP into separate QoS classes. You can either limit the UDP flows or you could even give the TCP flows a minimum bandwidth guarantee

Real Time Transport Protocol (RTP)

  • A generic protocol for real time applications such as voice and video
    • Uses UDP and acts as an interface between user application and transport protocol (Mostly UDP)
  • Header specifies the profile and encoding format of the payload (single audio stream, mp3)
  • Packets are numbered to allow detection of missing packets
  • Time stamping is used to allow synchronization and jitter compensation
  • No flow control, no error control, no acknowledgement, and no repeat-requests
  • Services:
    • Payload type identification
    • Sequence numbering
    • Time stamping
    • Delivery monitoring

RTP Fixed Header Fields

    0                   1                   2                   3
    0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |V=2|P|X|  CC   |M|     PT      |       sequence number         |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                           timestamp                           |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |           synchronization source (SSRC) identifier            |
   +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
   |            contributing source (CSRC) identifiers             |
   |                             ....                              |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
  • V = version (V): 2 bits
  • P = padding (P):
    • 1 bit - the packet has been padded to a multiple of 4-bytes. The last padding byte tells how many bytes were padded
  • extension (X): 1 bit
    • If the extension bit is set, the fixed header MUST be followed by exactly one header extension
  • CSRC count (CC): 4 bits
    • The CSRC count contains the number of CSRC identifiers that follow the fixed header
  • marker (M): 1 bit - used by the application
  • payload type (PT): 7 bits
    • identifies the format of the RTP payload and determines its interpretation by the application.
    • A receiver MUST ignore packets with payload types that it does not understand
  • sequence number: 16 bits
    • The sequence number increments by one for each RTP data packet sent, and may be used by the receiver to detect packet loss and to restore packet sequence
  • timestamp: 32 bits
    • Appropriate time alignment of the samples in the receiver for smooth playback and jitter reduction
    • Synchronization between multiple streams such as video and audio in a video conference
  • SSRC: 32 bits: Tells which stream this packet belongs to
  • CSRC list: 0 to 15 items, 32 bits each

Real Time Transport Control Protocol

  • RTP is used in conjunction with the RTP Control Protocol (RTCP).
  • RTP carries the media streams (e.g., audio and video) and RTCP is used to monitor transmission statistics and quality of service (QoS) and aids synchronization of multiple streams.
  • When both protocols are used in conjunction, RTP is originated and received on even port numbers and the associated RTCP communication uses the next higher odd port number.
  • RTCP feedback can let the source know about delay, jitter, bandwidth, congestion, etc.
  • This data can be used by the source to adjust encoding or transport properties of the stream
  • Different streams can use different clocks with different granularities and different drift rates.
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