- Email: [email protected]
voice integration for fibre optic LANs with arbitrary topology
Data/voice integration for fibre optic LANs with arbitrary
topology
The design and performance of a point-to-point protocol are evaluated by H K Pung and P A Davies
A high-performance protocol for arbitrary point-topoint fibre optic local area networks with combined voice and data traffic is described. Simulation on a 3 Mbit/s nine-node mesh network has shown that it is capable of supporting a substantial number of data users and more than 100 active voice calls (200 active telephone users, coding rate 64 kbit/s without silence detection) with a packet delay of less than 10 ms and without any loss of information. The network performance is superior to the previous reported voice protocol which used the modified anarchy flood routing (MAFR) technique. Performance limitation due to growing network size for flood-routed networks is also discussed. A method of extending the flood-routing protocols over subnetworks which are connected via bridges is also presented. This method enables a large network to be divided into smaller subnetworks which reduces the number of links required.
for mixed voice and data transmission which uses the controlled flood routing (CFR) 3 protocol previously proposed for data transmission. A protocol for voice/ data integration will be described and simulation results for the nine-node network 4 will be presented and discussed. Performance limitations due to the growing network size for flood-routed networks is also considered. A method of extending the flood-routing protocols to subnetworks which are connected by bridges is proposed. Finally, conclusions and some general comments are given. To gateway
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Keywords: computer networks, voice/data integration, protocols, fibre optics
The authors have recently been examining the feasibility of transmitting data on local area networks (LANs) which use optical fibres as the transmission medium 1-3 and which have a mesh topology based on an arbitrary interconnection of point-to-point links (Figure 1). The use of point-to-point links considerably simplifies the problems associated with using optical fibres in LANs. The authors have also examined the problem of transmitting pure voice traffic on such networks 4. In this paper they investigate the performance of a network E l e c t r o n i c s L a b o r a t o r i e s , U n i v e r s i t y o f K e n t at C a n t e r b u r y , CT2 7NT, UK
Canterbury
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Figure I. General structure of flood-routed LAN with arbitrary topology
0140-3664/84/050236-07503.00 © 1984 Butterworth & Co (Publishers) Ltd 236
computer communications
PROTOCOL FOR VOICE/DATA INTEGRATION The most distinctive difference between long-distance networks and LANs is the datarate. LANs provide a high data transmission rate with low packet transfer delay, and hence smaller variability in the delay. Obviously, design issues for voice protocols in a local area environment are somewhat different from the long-haul network, and they have been well discussed in the literature. In this section, the authors will describe a protocol for integrating voice and data transmission in a network with arbitrary topology. The protocol consists of a switch protocol and a station protocol. The station protocol, a modification of the real-time flood-routed voice (RTFRV) protocol 4, defines an interface between a user and a switch. The interface is called the terminal interface module (TIM) (Figure 1). The switch protocol, quite similar to the controlled flood-routing (CFR) protocol 3, is responsible for packet routing in the network. Voice and data terminals are connected to the network switches via TIMs.
Switch protocol A switch consists of a routing pointer array and a port connection array. It can exist in any one of the four states; the idle state, the transmission state, the routing state and the transient clearing state. A typical packet format includes an estimated hop (EH) field whose entry indicates the required hops from a source to a sink. A switch operates as follows: • If the switch is not in the routing state and receives an incoming packet from the source TIM, the switch sets routing pointers to point to all free outward links and then floods the packet to those links on the fly; otherwise a clear link signal (CLS) is generated to the TIM. The switch is then said to enter a routing state. A switch which exits from the routing state will flag 'switch ready', so that retransmission of the packet from the TIM can proceed if the previous attempt has failed. Subsequent packets arriving from the TIM will be flooded to the network in the same manner as described above. • A switch repeats an incoming flood packet to all its free outward links if the packet is not destined for itself. At the same time routing pointers are set to point to all the repeating ports. The switch is said to be in the routing state. • Upon detecting a packet which is destined for itself, a sink switch repeats the packet to the TIM and sets a port connection pointer to point to the input port while awaiting for an acknowledgment (ACK) from the TIM. The ACK will be repeated to the outward link of the input port on its arrival. The switch is then said to be in the transmission state. Any switch which is in the transmission state is allowed to repeat a newly arrived flood packet or to receive any packet destined for itself from different ports
vol 7 no 5 october 1984
simultaneously, provided that they are not from the same source. • Any switch which receives an ACK while it is in the routing state will change into the transmission state. A port connection pointer pair is then set to point to the most recent input port and the port where the ACK has just arrived. At the same time the switch clears the flood packet forward by ceasing broadcasting to all other outward links, as indicated by the routing pointers. These pointers are then reset. The ACK is also repeated to the outward link of the input port (at switch 2 in Figure 2). If it happens that the ACK reaches a source switch, it will be repeated to the source TIM only. • A flood packet which arrives at an intermediate switch with its EH decremented by one will not be repeated if: o the switch is already at routing state, o no free outward link is available, o the EH is smaller than its hop table entry whose index is equal to the packet's sink address. A clear backward procedure is then initiated. A clear link signal (CLS) is generated by the blocked switch to the outward link of the port (provided it is free) where the flood packet first arrived (at switch 7 in Figure 2). A switch upstream which receives the CLS while in the routing state will cease repeating the flood packet to the blocked switch, reset the corresponding routing pointer and then exit from the routing state (at switch 6 in Figure 2). The link is thus freed for other transmissions. This procedure is repeated towards the source, until a switch where none or more than one of the routing pointers is busy (at switches 4 in Figure 2). If it happens that the CLS reaches the source, it will cause a retransmission procedure to be activated at the source TIM. An input flood packet being terminated prematurely (i.e. before the end of the packet is reached) will cause a switch in the routing state to cease broadcasting. It is
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Figure 2. Snap shot of flood clearing in network with controlled flood routing (CFR) protocol
237
then said to be in a transient clearing state, and it may change into either an idle state or a transmission state.
Station protocol A station consists of user term inals, a TI M (Figure 1) and a hop table with each entry indicating the minimum number of hops to a sink. A voice terminal consists of a telephone set and a coder which generates voice samples at a constant rate. Output from user terminals is accumulated in their respective input buffers at the TIM. Packetization and delivery of packets to and from the switch are carried out by the TIM. The station protocol is very similar to the real-time voice protocol described in Reference 4 with the following refinements and exceptions. • A copy of the packet (voice or data) at a buffer which has first reached a predetermined length is transmitted to the switch by the TIM. Transmission will be aborted if a CLS (generated by the switch which has been busy) is subsequently received. Transmission, however, will be resumed once the switch become free again. • Unlike voice samples, a data stream is packetized into fixed size packets, each of a predetermined length. Since data integrity is more important than delay in data transmission, an unsuccessful transmission attempt will be repeated until the maximum number of retransmissions (typically 16) is reached. The packet will then be discarded and the ambiguity must be resolved by the higher level protocol. • The timeout clock and the retansmission backoff algorithm are implemented in the TIM instead of at the switch, as in the RTFRV protocol. • The EH field in the packet is incremented by one for each retransmission. EH should not exceed the maximum number of permissible hops (chosen to be six in the simulation). • The TIM at a sink will load the incoming packet to its appropriate output buffer. At the same time it generates an ACK signal with a bit pattern unique to the switch. A request for retransmission will be forwarded as part of the ACK if an error is detected. However, no error checking is required for voice packets. • To smooth out variability in voice packet delay, a simple fixed packet reassembling delay can be employed before playing out a packet to the user. It should be pointed out that the packet reassembling process is not included in this simulation. To provide a better understanding of the behaviour of the protocol under the mixed voice/data environment, a simulation model has been developed in SIMULA. Details of the assumptions of the model can be found in Reference 4. Definition of network parameters, simulation results and discussions are presented in the following section.
238
SIMULATION MODEL Definition of network parameters A simulation run is initiated by the creation of packet generators, and statistics are printed at the end of the run. The network parameters* to be determined and their definitions are as follows: • overall throughput - - the ratio of the total number of bits (voice and data) being received per second to the line rate, • overall offered load - - the ratio of the total number of bits being generated per second (voice and data) to the line rate, • packet transfer d e l a y - - the time between when a packet is ready for transmission and the successful transmission of the entire packet; it can be normalized to the data packet transmission time (Id/C), or in the case of a voice packet to the transmission time for the minimum voice packet length (Imin/C), where Id and Imin a r e the predetermined packet lengths for data and voice, respectively, • offered data load (rd) - - the ratio of the total offered data traffic from all data users to the line rate C, • offered voice load ( r r ) - - t h e ratio of the total offered voice traffic from all voice users to the line rate, • offered data/voice load ratio (r) -- simply given by rd/rv. • fairness--expressed in terms of the standard deviation of station throughput/station offered load; a routing protocol is fair if throughput for all stations is narrowly spread, i.e. with a small variance. The aim of the simulation is to establish the relationship of the network throughput, average packet delay, loss and fairness with respect to the overall offered load and the number of active calls for a particular traffic composition. It is also used to estimate the maximum number of simultaneous two-way voice conversations (active calls) that the network can support for a given set of performance requirements. Four simulation experiments, A1, A2, B1, B2, were carried out on the nine node point-to-point mesh network in Reference 4 as follows: • keeping r d constant while varying rv; rd was held at 0.4 for A1 and 0.8 for A2, • keeping r constant by varying r d and rv; r was kept at 1/3 for B1 and 1/2 for B2. Network load was increased by incrementing the number of users. Coder rate was chosen to be 64 kbit/s; voice sample size was 2 byte; Irnin = 128 byte; /max = 512 byte and C = 3 Mbit/s. The interarrival time of data packets is exponentially distributed, data packet length following a bimodal distribution whose mode values are 128 byte and 512 byte, respectively. A packet length will assume the first value with 0.8 probability and the second value with 0.2 probability. *parameters that have been defined in Reference 4 are not repeated here
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Simulation results and discussion Throughput The overall network throughput is not sensitive to traffic composition at low to moderate load. However, better throughput is obtained when the voice traffic proportion is larger (Figure 3, curve A1 and B1) for the following reasons; collisions are completely avoided once a transmission path has been established, and the path can be held for as long as is neccessary as if it is in a circuit-switched mode, which favours long packet transmissions. It has been observed that an increase in voice traffic proportion results in higher link blocking. Under these circumstances, more voice packets are transmitted, with a longer packet length as the load increases, and hence better throughput as a result of the circuit switched property. Surprisingly, data throughput does not seem to be affected by integration. In fact, throughput for data
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Packet transfer delay for both data and voice are approximately equal to the packet transmission time under light loading. However, it increases slowly until network saturation begins (Figure 5), thereafter delay for the data packet increases rapidly. This is not the case for the voice packets. In fact, delay for voice packets converges to the maximum dmax* when the offered load increases further. This is because packet queueing delay due to increasing link blocking probability becomes so large that most of the queued voice packets are close to or even exceed the maximum permissible length,/max- However, voice packet delay is bounded by allowing the source buffer to overflow, which means higher loss. The results in Figure 5 also show that the packet delay for voice and data tends to be higher in the network with greater voice traffic proportion.
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Figure 4. Voice a n d data throughput against overall offered load for the mesh network
vol 7 no 5 october 1984
*Ureax is a function of the maximum permissible voice packet length, the coder rate and the line rate. An expression for dma~ has been derived in Reference 4.
239
Loss
Figure 6 shows that there is no loss of voice information for an overall offered load of below 400%, which is twice that of the RTFRV under the pure voice environment. Beyond that the loss rate increases almost linearly with load, and larger r or rv results in greater loss (curves B1 and At). However, losses converge if the offered load increases further. Loss rate also increases with increasing number of active voice users. Fairness
The routing protocol is fair even when the network load is as high as 400% of the line rate (Figure 7). Fairness deteriorates slightly as the network begins to saturate, and levels off under heavy load due to the linear backoff. Fairness can obviously be improved if a nonpersistant network accessing scheme is used. The authors' previous study also indicated that fairness for flood routing techniques in general is topology dependant. Better fairness could be achieved in a mesh 30
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Further discussion Simulation results for fixed offered data load (rd) of 40% and 80% of the channel capacity (3 Mbitls) have shown that the network can support approximately 140 and 120 active calls (two-way conversations), respectively, with 64 kbit/s coder rate (Figure 8). This results in an average 10 ms voice packet end-to-end delay with no loss of information while maintaining almost 100% throughput for data transmission. Under constant data-to-voice offered load ration (r) of I / 3 and I / 2 , the network can support 100 two-way simultaneous voice conversations (200 active telephone users) with the same performance, which is twice as many as in the RTFRV protocol under pure voice load. The switch protocol for voice/data integration is equally well suited for use in any other flood-routing protocol. Flood-routing protocols in principle can be applied to any interconnected network of any size. However, it is very difficult to assess quantitatively the effects of increasing network size on the network performance via simulation. The overall network throughput improvement due to increasing numbers of simultaneous transmissions may not be substantial as network size grows, because a longer timeout is needed in a larger network. The extra link connections and the growing hop table size as required in the CFR protocol may not be justified in terms of cost/ performance. It is nevertheless possible to reduce the link connections and the timeout in a huge network by
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Figure 9. Network configuration for very large LAN with flood routing dividing it into smaller subareas (Figure 9). Bridges can be used to interconnect these subareas. A packet is still flooded in the same way as defined by a flood-routing protocol in a subarea. However, the packet will not be repeated bya bridge if it is destined within the subarea, otherwise a 'bridge found' signal is generated which will trace back to the source, as has been done by the ACK described earlier, so that the transmission path to the bridge can be established as rapidly as possible. If a bridge receives a flood packet, it either broadcasts the packet to its own subarea if the area code matches its own, or to others' bridges. Effectively, bridge connection forms a network at a higher level. However, a bridge should be treated as part of a subarea network as far as routing of packets within that area is concerned. If a bridge which is repeating a packet to its own area does not receive an ACK or a clear link signal (CLS) within the subarea's timeout, then a CLS is generated which will trace back to the source so that retransmission can take place. On the other hand, if the source does not receive an ACK, a CLS or a 'bridge found' signal within its local timeout, then retransmission will again take place. In this manner, packets can be flooded from area to another until the sink is found. The idea of dividing a network into subnetworks to allow network expansion was first mentioned by Ludwig and Roy in their study of the saturation routing protocol 5 and more recently by Petitpierre in his floodnet 6'7 which is similar to the Anarchy 8.
structure. Fortunately, the switch is transparent to the packet types and treats both types of packets in an identical manner. This should facilitate a simpler switch design. Furthermore, switches in this type of network may be easily duplicated, and hence, mass production is possible with VLSI technology. The novel backward clearing process as described in the switch protocol or in the CFR protocol can also be adapted by the Anarchy or the MAFR to relax the timeout constraint mentioned earlier. Flood routing protocols seem less effective for large networks. Longer transmission paths and an increasing number of transmission hops mean longer timeout, more packet duplications and, hence, lower channel utilization (the overall throughput may increase due to the increasing number of parallel packet transmissions). The hop count flood control in the CFR protocol also becomes less attractive. However, it is possible to divide a very large network into interconnected subareas in which packets for other subareas are repeated by bridges. Only packets with the correct area code will be broadcast by a bridge to its own subarea. The advantages of this scheme are threefold: first, the number of connection links can be reduced as network size grows; second, the total network size can be increased significantly and, third, flooding is confined to the related subareas, and the timeout is dependent on the size of the subarea only. The protocol for data/voice integration has retained the merit of network robustness and easy network reconfiguration, as in the Anarchy and the MAFR. Node and link failures are unlikely to halt the network operation, and nodes and links can be added regardless of the existing network topology and without disrupting the network operation. This kind of network has obvious applications in a hostile environment or an environment where network survivability and easy network reconfiguration are desired. It can either be a self-contained system within a ship or aircraft or even as part of the command control network on a battlefield. It can also be employed in an airport, a nuclear power station or a factory.
ACKNOWLEDGEMENT CONCLUSIONS The authors have proposed a protocol for combined data and voice transmission over arbitrary point-topoint connected fibre optic LANs. A simulation study conducted on a 3 Mbit/s nine-node mesh network using this protocol has shown that the network is capable of supporting a substantial number of data and voice users (without silence detection) with good performance. The superiority of the controlled flood-routing protocol over the Anarchy and the MAFR in a pure data or voice environment is reaffirmed in the mixed traffic environment. However, the additional intelligence that is required will no doubt complicate the node
vol 7 no 5 october 1984
The research into optical fibres in local area networks at the University of Kent is funded by British Telecom, whose support the authors gratefully acknowledge. H Pung acknowledges the support of a University of Kent Studentship and an ORS award.
REFERENCES Davies, P A, AbduI-Ghani, F and Pung, H K 'Optical fibres in local area networks' Internepcon 82, UK (12-14 October 1982)
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2
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4
Pung, H K and Davies, P A 'Fibre optic local area networks with arbitrary topology' lEE Proc. H:Microwaves, optics and antenna Part H No 2 Vol 131 (April 1984) Pung, H K and Davies, P A 'A new flood routing technique for fibre optic local area network with arbitrary topology'/CC 84 Amsterdam, The Netherlands, Vol 3 (14-17 May 1984) p 11 54 Pung,H K and Davies, PA'A real-time flood-routed voice protocol for fibre optic LANs Computer Commun. Vol 7 No 3 (June 1984) p 119
5 6 7 8
Ludwig,G and Roy, R 'Saturation routing networks limits' Proc. IEEE Vol 65 No 9 (September 1977) Petipierre, C 'Contention-free local area network technique with high-channel utilization' Melecon 83, Greece (May 1983) pp A2.08 Petipierre, C 'Influence of the network topology upon the throughput limitations' ICC 84 Amsterdam, The Netherlands, Voll (14-17 May) p428 Neff, R and Senzig, D 'A local area network design using fibre optics' IEEEProc. Compcon Spring 1987, USA (1981)
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242
computer communications
topology
The design and performance of a point-to-point protocol are evaluated by H K Pung and P A Davies
A high-performance protocol for arbitrary point-topoint fibre optic local area networks with combined voice and data traffic is described. Simulation on a 3 Mbit/s nine-node mesh network has shown that it is capable of supporting a substantial number of data users and more than 100 active voice calls (200 active telephone users, coding rate 64 kbit/s without silence detection) with a packet delay of less than 10 ms and without any loss of information. The network performance is superior to the previous reported voice protocol which used the modified anarchy flood routing (MAFR) technique. Performance limitation due to growing network size for flood-routed networks is also discussed. A method of extending the flood-routing protocols over subnetworks which are connected via bridges is also presented. This method enables a large network to be divided into smaller subnetworks which reduces the number of links required.
for mixed voice and data transmission which uses the controlled flood routing (CFR) 3 protocol previously proposed for data transmission. A protocol for voice/ data integration will be described and simulation results for the nine-node network 4 will be presented and discussed. Performance limitations due to the growing network size for flood-routed networks is also considered. A method of extending the flood-routing protocols to subnetworks which are connected by bridges is proposed. Finally, conclusions and some general comments are given. To gateway
//
Switch
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Keywords: computer networks, voice/data integration, protocols, fibre optics
The authors have recently been examining the feasibility of transmitting data on local area networks (LANs) which use optical fibres as the transmission medium 1-3 and which have a mesh topology based on an arbitrary interconnection of point-to-point links (Figure 1). The use of point-to-point links considerably simplifies the problems associated with using optical fibres in LANs. The authors have also examined the problem of transmitting pure voice traffic on such networks 4. In this paper they investigate the performance of a network E l e c t r o n i c s L a b o r a t o r i e s , U n i v e r s i t y o f K e n t at C a n t e r b u r y , CT2 7NT, UK
Canterbury
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Station
TIM
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Figure I. General structure of flood-routed LAN with arbitrary topology
0140-3664/84/050236-07503.00 © 1984 Butterworth & Co (Publishers) Ltd 236
computer communications
PROTOCOL FOR VOICE/DATA INTEGRATION The most distinctive difference between long-distance networks and LANs is the datarate. LANs provide a high data transmission rate with low packet transfer delay, and hence smaller variability in the delay. Obviously, design issues for voice protocols in a local area environment are somewhat different from the long-haul network, and they have been well discussed in the literature. In this section, the authors will describe a protocol for integrating voice and data transmission in a network with arbitrary topology. The protocol consists of a switch protocol and a station protocol. The station protocol, a modification of the real-time flood-routed voice (RTFRV) protocol 4, defines an interface between a user and a switch. The interface is called the terminal interface module (TIM) (Figure 1). The switch protocol, quite similar to the controlled flood-routing (CFR) protocol 3, is responsible for packet routing in the network. Voice and data terminals are connected to the network switches via TIMs.
Switch protocol A switch consists of a routing pointer array and a port connection array. It can exist in any one of the four states; the idle state, the transmission state, the routing state and the transient clearing state. A typical packet format includes an estimated hop (EH) field whose entry indicates the required hops from a source to a sink. A switch operates as follows: • If the switch is not in the routing state and receives an incoming packet from the source TIM, the switch sets routing pointers to point to all free outward links and then floods the packet to those links on the fly; otherwise a clear link signal (CLS) is generated to the TIM. The switch is then said to enter a routing state. A switch which exits from the routing state will flag 'switch ready', so that retransmission of the packet from the TIM can proceed if the previous attempt has failed. Subsequent packets arriving from the TIM will be flooded to the network in the same manner as described above. • A switch repeats an incoming flood packet to all its free outward links if the packet is not destined for itself. At the same time routing pointers are set to point to all the repeating ports. The switch is said to be in the routing state. • Upon detecting a packet which is destined for itself, a sink switch repeats the packet to the TIM and sets a port connection pointer to point to the input port while awaiting for an acknowledgment (ACK) from the TIM. The ACK will be repeated to the outward link of the input port on its arrival. The switch is then said to be in the transmission state. Any switch which is in the transmission state is allowed to repeat a newly arrived flood packet or to receive any packet destined for itself from different ports
vol 7 no 5 october 1984
simultaneously, provided that they are not from the same source. • Any switch which receives an ACK while it is in the routing state will change into the transmission state. A port connection pointer pair is then set to point to the most recent input port and the port where the ACK has just arrived. At the same time the switch clears the flood packet forward by ceasing broadcasting to all other outward links, as indicated by the routing pointers. These pointers are then reset. The ACK is also repeated to the outward link of the input port (at switch 2 in Figure 2). If it happens that the ACK reaches a source switch, it will be repeated to the source TIM only. • A flood packet which arrives at an intermediate switch with its EH decremented by one will not be repeated if: o the switch is already at routing state, o no free outward link is available, o the EH is smaller than its hop table entry whose index is equal to the packet's sink address. A clear backward procedure is then initiated. A clear link signal (CLS) is generated by the blocked switch to the outward link of the port (provided it is free) where the flood packet first arrived (at switch 7 in Figure 2). A switch upstream which receives the CLS while in the routing state will cease repeating the flood packet to the blocked switch, reset the corresponding routing pointer and then exit from the routing state (at switch 6 in Figure 2). The link is thus freed for other transmissions. This procedure is repeated towards the source, until a switch where none or more than one of the routing pointers is busy (at switches 4 in Figure 2). If it happens that the CLS reaches the source, it will cause a retransmission procedure to be activated at the source TIM. An input flood packet being terminated prematurely (i.e. before the end of the packet is reached) will cause a switch in the routing state to cease broadcasting. It is
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Figure 2. Snap shot of flood clearing in network with controlled flood routing (CFR) protocol
237
then said to be in a transient clearing state, and it may change into either an idle state or a transmission state.
Station protocol A station consists of user term inals, a TI M (Figure 1) and a hop table with each entry indicating the minimum number of hops to a sink. A voice terminal consists of a telephone set and a coder which generates voice samples at a constant rate. Output from user terminals is accumulated in their respective input buffers at the TIM. Packetization and delivery of packets to and from the switch are carried out by the TIM. The station protocol is very similar to the real-time voice protocol described in Reference 4 with the following refinements and exceptions. • A copy of the packet (voice or data) at a buffer which has first reached a predetermined length is transmitted to the switch by the TIM. Transmission will be aborted if a CLS (generated by the switch which has been busy) is subsequently received. Transmission, however, will be resumed once the switch become free again. • Unlike voice samples, a data stream is packetized into fixed size packets, each of a predetermined length. Since data integrity is more important than delay in data transmission, an unsuccessful transmission attempt will be repeated until the maximum number of retransmissions (typically 16) is reached. The packet will then be discarded and the ambiguity must be resolved by the higher level protocol. • The timeout clock and the retansmission backoff algorithm are implemented in the TIM instead of at the switch, as in the RTFRV protocol. • The EH field in the packet is incremented by one for each retransmission. EH should not exceed the maximum number of permissible hops (chosen to be six in the simulation). • The TIM at a sink will load the incoming packet to its appropriate output buffer. At the same time it generates an ACK signal with a bit pattern unique to the switch. A request for retransmission will be forwarded as part of the ACK if an error is detected. However, no error checking is required for voice packets. • To smooth out variability in voice packet delay, a simple fixed packet reassembling delay can be employed before playing out a packet to the user. It should be pointed out that the packet reassembling process is not included in this simulation. To provide a better understanding of the behaviour of the protocol under the mixed voice/data environment, a simulation model has been developed in SIMULA. Details of the assumptions of the model can be found in Reference 4. Definition of network parameters, simulation results and discussions are presented in the following section.
238
SIMULATION MODEL Definition of network parameters A simulation run is initiated by the creation of packet generators, and statistics are printed at the end of the run. The network parameters* to be determined and their definitions are as follows: • overall throughput - - the ratio of the total number of bits (voice and data) being received per second to the line rate, • overall offered load - - the ratio of the total number of bits being generated per second (voice and data) to the line rate, • packet transfer d e l a y - - the time between when a packet is ready for transmission and the successful transmission of the entire packet; it can be normalized to the data packet transmission time (Id/C), or in the case of a voice packet to the transmission time for the minimum voice packet length (Imin/C), where Id and Imin a r e the predetermined packet lengths for data and voice, respectively, • offered data load (rd) - - the ratio of the total offered data traffic from all data users to the line rate C, • offered voice load ( r r ) - - t h e ratio of the total offered voice traffic from all voice users to the line rate, • offered data/voice load ratio (r) -- simply given by rd/rv. • fairness--expressed in terms of the standard deviation of station throughput/station offered load; a routing protocol is fair if throughput for all stations is narrowly spread, i.e. with a small variance. The aim of the simulation is to establish the relationship of the network throughput, average packet delay, loss and fairness with respect to the overall offered load and the number of active calls for a particular traffic composition. It is also used to estimate the maximum number of simultaneous two-way voice conversations (active calls) that the network can support for a given set of performance requirements. Four simulation experiments, A1, A2, B1, B2, were carried out on the nine node point-to-point mesh network in Reference 4 as follows: • keeping r d constant while varying rv; rd was held at 0.4 for A1 and 0.8 for A2, • keeping r constant by varying r d and rv; r was kept at 1/3 for B1 and 1/2 for B2. Network load was increased by incrementing the number of users. Coder rate was chosen to be 64 kbit/s; voice sample size was 2 byte; Irnin = 128 byte; /max = 512 byte and C = 3 Mbit/s. The interarrival time of data packets is exponentially distributed, data packet length following a bimodal distribution whose mode values are 128 byte and 512 byte, respectively. A packet length will assume the first value with 0.8 probability and the second value with 0.2 probability. *parameters that have been defined in Reference 4 are not repeated here
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Simulation results and discussion Throughput The overall network throughput is not sensitive to traffic composition at low to moderate load. However, better throughput is obtained when the voice traffic proportion is larger (Figure 3, curve A1 and B1) for the following reasons; collisions are completely avoided once a transmission path has been established, and the path can be held for as long as is neccessary as if it is in a circuit-switched mode, which favours long packet transmissions. It has been observed that an increase in voice traffic proportion results in higher link blocking. Under these circumstances, more voice packets are transmitted, with a longer packet length as the load increases, and hence better throughput as a result of the circuit switched property. Surprisingly, data throughput does not seem to be affected by integration. In fact, throughput for data
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vol 7 no 5 october 1984
*Ureax is a function of the maximum permissible voice packet length, the coder rate and the line rate. An expression for dma~ has been derived in Reference 4.
239
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Figure 6 shows that there is no loss of voice information for an overall offered load of below 400%, which is twice that of the RTFRV under the pure voice environment. Beyond that the loss rate increases almost linearly with load, and larger r or rv results in greater loss (curves B1 and At). However, losses converge if the offered load increases further. Loss rate also increases with increasing number of active voice users. Fairness
The routing protocol is fair even when the network load is as high as 400% of the line rate (Figure 7). Fairness deteriorates slightly as the network begins to saturate, and levels off under heavy load due to the linear backoff. Fairness can obviously be improved if a nonpersistant network accessing scheme is used. The authors' previous study also indicated that fairness for flood routing techniques in general is topology dependant. Better fairness could be achieved in a mesh 30
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Further discussion Simulation results for fixed offered data load (rd) of 40% and 80% of the channel capacity (3 Mbitls) have shown that the network can support approximately 140 and 120 active calls (two-way conversations), respectively, with 64 kbit/s coder rate (Figure 8). This results in an average 10 ms voice packet end-to-end delay with no loss of information while maintaining almost 100% throughput for data transmission. Under constant data-to-voice offered load ration (r) of I / 3 and I / 2 , the network can support 100 two-way simultaneous voice conversations (200 active telephone users) with the same performance, which is twice as many as in the RTFRV protocol under pure voice load. The switch protocol for voice/data integration is equally well suited for use in any other flood-routing protocol. Flood-routing protocols in principle can be applied to any interconnected network of any size. However, it is very difficult to assess quantitatively the effects of increasing network size on the network performance via simulation. The overall network throughput improvement due to increasing numbers of simultaneous transmissions may not be substantial as network size grows, because a longer timeout is needed in a larger network. The extra link connections and the growing hop table size as required in the CFR protocol may not be justified in terms of cost/ performance. It is nevertheless possible to reduce the link connections and the timeout in a huge network by
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Figure 9. Network configuration for very large LAN with flood routing dividing it into smaller subareas (Figure 9). Bridges can be used to interconnect these subareas. A packet is still flooded in the same way as defined by a flood-routing protocol in a subarea. However, the packet will not be repeated bya bridge if it is destined within the subarea, otherwise a 'bridge found' signal is generated which will trace back to the source, as has been done by the ACK described earlier, so that the transmission path to the bridge can be established as rapidly as possible. If a bridge receives a flood packet, it either broadcasts the packet to its own subarea if the area code matches its own, or to others' bridges. Effectively, bridge connection forms a network at a higher level. However, a bridge should be treated as part of a subarea network as far as routing of packets within that area is concerned. If a bridge which is repeating a packet to its own area does not receive an ACK or a clear link signal (CLS) within the subarea's timeout, then a CLS is generated which will trace back to the source so that retransmission can take place. On the other hand, if the source does not receive an ACK, a CLS or a 'bridge found' signal within its local timeout, then retransmission will again take place. In this manner, packets can be flooded from area to another until the sink is found. The idea of dividing a network into subnetworks to allow network expansion was first mentioned by Ludwig and Roy in their study of the saturation routing protocol 5 and more recently by Petitpierre in his floodnet 6'7 which is similar to the Anarchy 8.
structure. Fortunately, the switch is transparent to the packet types and treats both types of packets in an identical manner. This should facilitate a simpler switch design. Furthermore, switches in this type of network may be easily duplicated, and hence, mass production is possible with VLSI technology. The novel backward clearing process as described in the switch protocol or in the CFR protocol can also be adapted by the Anarchy or the MAFR to relax the timeout constraint mentioned earlier. Flood routing protocols seem less effective for large networks. Longer transmission paths and an increasing number of transmission hops mean longer timeout, more packet duplications and, hence, lower channel utilization (the overall throughput may increase due to the increasing number of parallel packet transmissions). The hop count flood control in the CFR protocol also becomes less attractive. However, it is possible to divide a very large network into interconnected subareas in which packets for other subareas are repeated by bridges. Only packets with the correct area code will be broadcast by a bridge to its own subarea. The advantages of this scheme are threefold: first, the number of connection links can be reduced as network size grows; second, the total network size can be increased significantly and, third, flooding is confined to the related subareas, and the timeout is dependent on the size of the subarea only. The protocol for data/voice integration has retained the merit of network robustness and easy network reconfiguration, as in the Anarchy and the MAFR. Node and link failures are unlikely to halt the network operation, and nodes and links can be added regardless of the existing network topology and without disrupting the network operation. This kind of network has obvious applications in a hostile environment or an environment where network survivability and easy network reconfiguration are desired. It can either be a self-contained system within a ship or aircraft or even as part of the command control network on a battlefield. It can also be employed in an airport, a nuclear power station or a factory.
ACKNOWLEDGEMENT CONCLUSIONS The authors have proposed a protocol for combined data and voice transmission over arbitrary point-topoint connected fibre optic LANs. A simulation study conducted on a 3 Mbit/s nine-node mesh network using this protocol has shown that the network is capable of supporting a substantial number of data and voice users (without silence detection) with good performance. The superiority of the controlled flood-routing protocol over the Anarchy and the MAFR in a pure data or voice environment is reaffirmed in the mixed traffic environment. However, the additional intelligence that is required will no doubt complicate the node
vol 7 no 5 october 1984
The research into optical fibres in local area networks at the University of Kent is funded by British Telecom, whose support the authors gratefully acknowledge. H Pung acknowledges the support of a University of Kent Studentship and an ORS award.
REFERENCES Davies, P A, AbduI-Ghani, F and Pung, H K 'Optical fibres in local area networks' Internepcon 82, UK (12-14 October 1982)
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Pung, H K and Davies, P A 'Fibre optic local area networks with arbitrary topology' lEE Proc. H:Microwaves, optics and antenna Part H No 2 Vol 131 (April 1984) Pung, H K and Davies, P A 'A new flood routing technique for fibre optic local area network with arbitrary topology'/CC 84 Amsterdam, The Netherlands, Vol 3 (14-17 May 1984) p 11 54 Pung,H K and Davies, PA'A real-time flood-routed voice protocol for fibre optic LANs Computer Commun. Vol 7 No 3 (June 1984) p 119
5 6 7 8
Ludwig,G and Roy, R 'Saturation routing networks limits' Proc. IEEE Vol 65 No 9 (September 1977) Petipierre, C 'Contention-free local area network technique with high-channel utilization' Melecon 83, Greece (May 1983) pp A2.08 Petipierre, C 'Influence of the network topology upon the throughput limitations' ICC 84 Amsterdam, The Netherlands, Voll (14-17 May) p428 Neff, R and Senzig, D 'A local area network design using fibre optics' IEEEProc. Compcon Spring 1987, USA (1981)
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