Network packet

A network packet is a formatted unit of data carried by a packet-switched network. Computer communications links that do not support packets, such as traditional point-to-point telecommunications links, simply transmit data as a bit stream. When data is formatted into packets, packet switching is possible and the bandwidth of the communication medium can be better shared among users than with circuit switching.

A packet consists of control information and user data,^[1] which is also known as the payload. Control information provides data for delivering the payload, for example: source and destination network addresses, error detection codes, and sequencing information. Typically, control information is found in packet headers and trailers.

Terminology

In the seven-layer OSI model of computer networking, packet strictly refers to a data unit at layer 3, the Network Layer. The correct term for a data unit at Layer 2, the Data Link Layer, is a frame, and at Layer 4, the Transport Layer, the correct term is a segment or datagram. For the case of TCP/IP communication over Ethernet, a TCP segment is carried in one or more IP packets, which are each carried in one or more Ethernet frames.

Packet framing

Different communications protocols use different conventions for distinguishing between the elements and for formatting the data. For example, in Point-to-Point Protocol, the packet is formatted in 8-bit bytes, and special characters are used to delimit the different elements. Other protocols like Ethernet, establish the start of the header and data elements by their location relative to the start of the packet. Some protocols format the information at a bit level instead of a byte level.

A good analogy is to consider a packet to be like a letter: the header is like the envelope, and the data area is whatever the person puts inside the envelope.^[2]

A network design can achieve two major results by using packets: error detection and multiple host addressing. A packet has the following components.

Addresses

The routing of network packets requires two network addresses, the source address of the sending host, and the destination address of the receiving host.

Error detection and correction

Error detection and correction is performed at various layers in the protocol stack. Network packets may contain a checksum, parity bits or cyclic redundancy checks to detect errors that occur during transmission.

At the transmitter, the calculation is performed before the packet is sent. When received at the destination, the checksum is recalculated, and compared with the one in the packet. If discrepancies are found, the packet may be corrected or discarded. Any packet loss is dealt with by the network protocol.

In some cases modifications of the network packet may be necessary while routing, in which cases checksums are recalculated.

Hop counts

Under fault conditions packets can end up traversing a closed circuit. If nothing was done, eventually the number of packets circulating would build up until the network was congested to the point of failure. A time to live is a field that is decreased by one each time a packet goes through a network node. If the field reaches zero, routing has failed, and the packet is discarded.

Ethernet packets have no time-to-live field and so are subject to broadcast radiation in the presence of a switch loop.

Length

There may be a field to identify the overall packet length. However, in some types of networks, the length is implied by the duration of transmission.

Priority

Some networks implement quality of service which can prioritize some types of packets above others. This field indicates which packet queue should be used; a high priority queue is emptied more quickly than lower priority queues at points in the network where congestion is occurring.

Payload

In general, payload is the data that is carried on behalf of an application. It is usually of variable length, up to a maximum that is set by the network protocol and sometimes the equipment on the route. Some networks can break a larger packet into smaller packets when necessary.

Example: IP packets

IP packets are composed of a header and payload. The IPv4 packet header consists of:^[3]

00	01	02	03	04	05	06	07	08	09	10	11	12	13	14	15	16	17	18	19	20	21	22	23	24	25	26	27	28	29	30	31	(bit position)
Version				IHL				QoS								Length
ID																0	DF	MF	Fragment Offset
TTL								Protocol								Checksum
Source IP
Destination IP

1. 4 bits that contain the version, that specifies if it's an IPv4 or IPv6 packet,
2. 4 bits that contain the Internet Header Length, which is the length of the header in multiples of 4 bytes (e.g., 5 means 20 bytes).
3. 8 bits that contain the Type of Service, also referred to as Quality of Service (QoS), which describes what priority the packet should have,
4. 16 bits that contain the length of the packet in bytes,
5. 16 bits that contain an identification tag to help reconstruct the packet from several fragments,
6. 3 bits. The first contains a zero, followed by a flag that says whether the packet is allowed to be fragmented or not (DF: Don't fragment), and a flag to state whether more fragments of a packet follow (MF: More Fragments)
7. 13 bits that contain the fragment offset, a field to identify position of fragment within original packet
8. 8 bits that contain the Time to live (TTL), which is the number of hops (router, computer or device along a network) the packet is allowed to pass before it dies (for example, a packet with a TTL of 16 will be allowed to go across 16 routers to get to its destination before it is discarded),
9. 8 bits that contain the protocol (TCP, UDP, ICMP, etc.)
10. 16 bits that contain the Header Checksum, a number used in error detection,
11. 32 bits that contain the source IP address,
12. 32 bits that contain the destination address.

After those 160 bits, optional flags can be added of varied length, which can change based on the protocol used, then the data that packet carries is added. An IP packet has no trailer. However, an IP packet is often carried as the payload inside an Ethernet frame, which has its own header and trailer.

Many networks do not provide guarantees of delivery, non-duplication of packets, or in-order delivery of packets, e.g., the UDP protocol of the Internet. However, it is possible to layer a transport protocol on top of the packet service that can provide such protection; TCP and UDP are the best examples of layer 4, the Transport Layer, of the seven layered OSI model.

Example: the NASA Deep Space Network

The Consultative Committee for Space Data Systems (CCSDS) packet telemetry standard defines the protocol used for the transmission of spacecraft instrument data over the deep-space channel. Under this standard, an image or other data sent from a spacecraft instrument is transmitted using one or more packets.

CCSDS packet definition

A packet is a block of data with length that can vary between successive packets, ranging from 7 to 65,542 bytes, including the packet header.

• Packetized data is transmitted via frames, which are fixed-length data blocks. The size of a frame, including frame header and control information, can range up to 2048 bytes.
• Packet sizes are fixed during the development phase.

Because packet lengths are variable but frame lengths are fixed, packet boundaries usually do not coincide with frame boundaries.

Telecom processing notes

Data in a frame is typically protected from channel errors by error-correcting codes.

• Even when the channel errors exceed the correction capability of the error-correcting code, the presence of errors is nearly always detected by the error-correcting code or by a separate error-detecting code.
• Frames for which uncorrectable errors are detected are marked as undecodable and typically are deleted.

Handling data loss

Deleted undecodable whole frames are the principal type of data loss that affects compressed data sets. In general, there would be little to gain from attempting to use compressed data from a frame marked as undecodable.

• When errors are present in a frame, the bits of the subband pixels are already decoded before the first bit error will remain intact, but all subsequent decoded bits in the segment usually will be completely corrupted; a single bit error is often just as disruptive as many bit errors.
• Furthermore, compressed data usually are protected by powerful, long-blocklength error-correcting codes, which are the types of codes most likely to yield substantial fractions of bit errors throughout those frames that are undecodable.

Thus, frames with detected errors would be essentially unusable even if they were not deleted by the frame processor.

This data loss can be compensated for with the following mechanisms.

• If an erroneous frame escapes detection, the decompressor will blindly use the frame data as if they were reliable, whereas in the case of detected erroneous frames, the decompressor can base its reconstruction on incomplete, but not misleading, data.
• However, it is extremely rare for an erroneous frame to go undetected.
• For frames coded by the CCSDS Reed–Solomon code, fewer than 1 in 40,000 erroneous frames can escape detection.
• All frames not employing the Reed–Solomon code use a cyclic redundancy check (CRC) error-detecting code, which has an undetected frame-error rate of less than 1 in 32,000.

Example: Radio and TV broadcasting

MPEG packetized stream

Packetized Elementary Stream (PES) is a specification defined by the MPEG communication protocol (see the MPEG-2 standard) that allows an elementary stream to be divided into packets. The elementary stream is packetized by encapsulating sequential data bytes from the elementary stream inside PES packet headers.

A typical method of transmitting elementary stream data from a video or audio encoder is to first create PES packets from the elementary stream data and then to encapsulate these PES packets inside an MPEG transport stream (TS) packets or an MPEG program stream (PS). The TS packets can then be multiplexed and transmitted using broadcasting techniques, such as those used in an ATSC and DVB.

PES packet header

Optional PES header

NICAM

In order to provide mono "compatibility", the NICAM signal is transmitted on a subcarrier alongside the sound carrier. This means that the FM or AM regular mono sound carrier is left alone for reception by monaural receivers.

A NICAM-based stereo-TV infrastructure can transmit a stereo TV programme as well as the mono "compatibility" sound at the same time, or can transmit two or three entirely different sound streams. This latter mode could be used to transmit audio in different languages, in a similar manner to that used for in-flight movies on international flights. In this mode, the user can select which soundtrack to listen to when watching the content by operating a "sound-select" control on the receiver.

NICAM offers the following possibilities. The mode is auto-selected by the inclusion of a 3-bit type field in the data-stream

• One digital stereo sound channel.
• Two completely different digital mono sound channels.
• One digital mono sound channel and a 352 kbit/s data channel.
• One 704 kbit/s data channel.

The four other options could be implemented at a later date. Only the first two of the ones listed are known to be in general use however.

NICAM packet transmission

The NICAM packet (except for the header) is scrambled with a nine-bit pseudo-random bit-generator before transmission.

• The topology of this pseudo-random generator yields a bitstream with a repetition period of 511 bits.
• The pseudo-random generator's polynomial is: x^9 + x^4 + 1.
• The pseudo-random generator is initialized with: 111111111.

Making the NICAM bitstream look more like white noise is important because this reduces signal patterning on adjacent TV channels.

• The NICAM header is not subject to scrambling. This is necessary so as to aid in locking on to the NICAM data stream and resynchronisation of the data stream at the receiver.
• At the start of each NICAM packet the pseudo-random bit generator's shift-register is reset to all-ones.

See also

References