The basic services provided by the network are referred to as “AV” and “IT”.
[Note: previously they were referred to as “synchronous” and “asynchronous” respectively, but that is inconsistent with the definition of an asynchronous service used by IETF's Deterministic Networking (detnet) group.]
Other services built upon them, including IP, are also described below.
Both the basic services are connection-oriented (also see here), so the mechanism is similar to making a telephone call or “tuning in” to a radio or TV broadcast. This allows the necessary resources to be reserved to provide a guaranteed service, and eliminates the overhead of providing addressing information in every packet. It also allows a clean separation between the choosing the route (a complex process done once per call, by software) and directing the packets along it (a simple process done for every packet by hardware).
The AV service is appropriate to live media such as audio and video (hence the name), also for telephony and other latency-critical applications such as virtual reality and tactile feedback. Data flow is one-to-many: one unit is the source, and the network copies the packets as required to reach all the destinations. This is similar to the way analogue audio and video routers work, also to the ATM point-to-multipoint configuration, but different from IP multicast.
A call is initiated from the destination. This is similar to tuning into a broadcast or “taking” a source from a cross-point router.
The call carries packets which are sent at defined intervals. The interval, and the maximum size of a packet, are specified by the source.
The overhead per packet is very low, so frequent short packets can be used, minimising the delays.
AV packets (like ISDN channels) are identified by their position in the data stream, so the system does not need to examine the packet in order to know where to send it. This simplifies switching equipment and allows the delay to be kept as small as possible and to be calculated exactly.
By contrast, RTP, the current Internet standard for “streaming” media, needs up to about 100 bytes of overhead per packet. Routing is “store and forward”, so each packet must be read into memory and then join the queue for the next link after the system has read the header and decided where it needs to go next. Most broadcasts over the Internet send a separate copy of the data to each listener.
The foreground service is also better than an ATM CBR service, which requires the packets to contain 48 bytes each and has higher latency because it does more buffering of the packets and is less tightly synchronised.
All of the capacity not required for AV packets, including capacity reserved but not used, is available for IT traffic. IT packets are label-routed and can be of any size from zero to about 2000 bytes; they experience a similar “best effort” service, with store-and-forward routing, to other packet networks.
There are two ways in which this service is used:
These calls are bidirectional. Session-oriented protocols such as TCP can be used directly over this service, eliminating the additional headers that are otherwise required to carry addressing information in every packet.
This is used to carry IP and similar traffic, and works in a similar way to route-caching in IP networks. The first packet to be sent to a new address is encapsulated in a signalling message, and as it is conveyed towards its destination a route is set up which subsequent packets will use. As soon as it arrives at a switch that already has a route to the destination, the new route is joined onto the existing one; thus these routes are many-to-one.
An address may serve to identify a service or endpoint, or to locate it within the topology of the network, or both. Where it does both, there should be a clear partition between the two parts. This is true of the NSAP addresses used in ATM (the prefix is a locator and the ESI is an identifier) but not of IPv4 addresses.
Flexilink supports a wide range of addressing schemes, including IPv4 and IPv6 as well as higher-level forms such as URLs. Call connection uses the protocols specified in IEC 62379-5-2. This includes the option of hierarchical addressing whereby the first part of the address specifies a location, such as a gateway into a subnetwork, and the remainder is to be interpreted at that location. Where there is no locator, the address is interpreted as being on the local subnetwork.
Various checks are made when a connection is requested, and if any of them fails the call is refused, with a message giving the reason. Among the things that may be checked are the caller's rights to access the equipment or service, the destination's support for the format to be transmitted, and of course for AV flows that the necessary bandwidth can be reserved. If appropriate, billing information can be collected.
The network can also be requested to set up more than one route for a call, avoiding, as far as possible, letting both routes pass through the same piece of equipment. An application that needs extra reliability can thus set up two different routes, and transmit the data on both of them. Then any single failure can only interrupt one of the calls, and the destination will continue to receive at least one copy.
The addressing mechanism allows connection to units on other kinds of network, such as IP networks that use SIP. It also allows connection between subnetworks of different kinds.
Management protocols are based on IEC 62379, which is in turn based on SNMP.
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