Documenti di Didattica
Documenti di Professioni
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aUthOrs
sYLVain COrnaY, Marketing Manager, eXFO haMMaDOUn DiCKO, Product specialist, eXFO thiernO DiaLLO, Product specialist, eXFO sOPhie LeGaULt, Product Line Manager, eXFO sUe JUDGe, Consultant eXFO inc. March 2011
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Carrier ethernet applications: 1.4.1 1.4.2 1.4.3 Business services Mobile Backhaul services Key Performance indicators MPLs
1.5
Key technologies Overview 1.5.1 1.5.2 MPLs-tP 1.5.3 PBB-te 1.5.4 Ptn 1.5.5 PWe3 1.5.6 Circuit emulation services 1.5.7 ethernet OaM 1.5.8 synchronization
Also coming soon to the Carrier Ethernet Basic Educational Series, modules that will focus on the following aspects of Carrier Ethernet, including service turn-up, service monitoring and troubleshooting.
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1.1
As technologically sophisticated businesses and residential consumers continue to drive the demand for premium, high-bandwidth data services such as voice and video, service providers worldwide must evolve their transport infrastructures to support these bandwidth and quality-intensive services. No longer is an all-IP core sufficientproviders must now expand their IP convergence to the edge/metro network, in a cost-effective, quality-assured manner.
1.2
transport Media
The diagram below outlines the different media used within a Carrier Ethernet network and indicates where they are commonly deployed: ETHERNET ACCESS
1.2.1 Copper
To this day, copper cabling (i.e., insulated twisted copper wires) is still one of the most widely used media in Carrier Ethernet due to its existing vast deployment and its relatively low cost. It is almost everywhere as it was the media of choice to deliver plain old telephony service (POTS) to homes and businesses. Leveraging this infrastructure, service providers can avoid building out new and costly networks, as they address markets with lower-rate traffic of up to 1 Gigabit per second (Gbit/s) and begin to carry higher-speed traffic (in some cases up to 10 Gbit/s). Ethernets inherent scalability gives carriers a highly flexible platform for delivering incremental services to smaller enterprises, branch offices, cellular towers and other sites. However, copper is subject to both electromagnetic interference and cross-talk, which can negatively affect the reliable transfer of digital dataand at high speeds, the problem is even worse.
1.2.2 Microwave
Ethernet is also used for mobile backhaul, the distance from a cell tower to a switching office or between switching offices. The medium used is actually microwave-over-the-air. Microwave radio is a popular infrastructure choice for wireless operators. Ethernet-enabled microwave is becoming an increasingly important component of a wireless infrastructure. The increasing interest in microwave is driven by the higher bandwidth demands at the base station sites and the requirement to provide a substantial reduction in operational costs of backhauling the data traffic. The growth of the wireless industry combined with the proliferation of the mobile backhaul will only contribute to increase the use of microwave radio as a transport medium.
1.2.3 Fiber
Since fiber can carry much more information than copper, carrier Ethernet service providers typically use fiber to transport high-speed traffic (usually 1 Gbit/s or more) over long distances or within the network core. Fiber is used with SONET/SDH, dense wavelength-division multiplexing (DWDM) or optical transport networks (OTNs). Fiber cabling may have an initial higher cost, but even at the fastest speeds, it is entirely resistant to both cross-talk and electromagnetic interference, therefore it can provide much more reliable data transmission. As the demand for bandwidth and speed increases, the need to implement fiber on networks, even at the business site, is growing. However, the main issue with fiber is the high cost of deployment and maintenance.
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1.3
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1.4
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Bandwidth refers to the maximum amount of data that can be forwarded. This measurement is a ratio of the total amount of traffic forwarded during a measurement window of one second. Bandwidth can either be committed or excess with different performance guarantees. Frame delay variation, or packet jitter, refers to the variability in arrival time between packet deliveries. As packets travel through a network, they are often queued and sent in bursts to the next hop. Random prioritization may occur, resulting in packet transmission at random rates. Packets are therefore received at irregular intervals. This jitter translates into stress on the receiving buffers of the end nodes, where buffers can be overused or underused when there are large swings of jitter. Real-time applications are especially sensitive to packet jitter. Buffers are designed to store a certain quantity of video or voice packets, which are then processed at regular intervals to provide a smooth and error-free transmission to the end user. Too much jitter will affect the quality of experience (QoE)where packets arriving at a fast rate will cause buffers to overfill, leading to packet loss; while packets arriving at a slow rate will cause buffers to empty, leading to still images or sound.
1.5
1.5.2 MPLs-tP
With the movement toward packet-based services, transport networks have to encompass the provision of packet-aware capabilities while enabling carriers to leverage their installed transport infrastructure investments. MPLS transport profile (MPSL-TP) is a derivative of MPLS designed for transport networks. It supports the capabilities and functionalities needed for packet-transport network services and operations through combining the packet experience of MPLS with the operational experience and practices of existing transport networks. MPLS-TP enables the deployment of packet-based transport networks that efficiently scales to support packet services in a simple and cost-effective way.
1.5.3 PBB-te
Provider backbone bridge traffic engineering or PBB-TE (also referred to as PBT) is an alternative Ethernet-based implementation that enables carrier-grade provisioning and management of connection-oriented transport services across an
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all-IP MAN and core network by disabling the flooding/broadcasting and spanning tree protocol features. It is an evolution of MAC-in-MAC by making it connectionoriented. PBB-TE separates the Ethernet service layer from the network layer; its flexibility also allows service providers to deliver native Ethernet initially and MPLSbased servicesi.e., virtual private wire service (VPWS) or virtual private LAN service (VPLS)if and when they are required.
1.5.4 Ptn
The packet transport network (PTN) is the next generation of networks designed around the best elements of traditional TDM technologies and the emergent packet technologies. It is typically deployed at two layers. At the access layer, PTN provides convergence of multiple services by converging TDM and packets into the PTN cloud. TDM packets are encapsulated and forwarded as packets in the PTN cloud while native Ethernet/IP packets are encapsulated and forwarded in the same PTN cloud. PTN networks overcome many of the challenges of carriers by providing the efficient data transport of packetized technologies with the fault detection and resiliency of TDM-based networks. Service providers can now leverage the inherent advantages of Ethernet and TDM technologies, such as cost effectiveness, flexibility, multiservice applications as well as quality of service.
1.5.5 PWe3
Pseudo wire emulation edge-to-edge (PWE3) is a mechanism that emulates the essential attributes of a service such as ATM, frame relay or Ethernet over a packet switched network (PSN). PWE3 only provides the minimum required functionality to emulate the wire. From the customer perspective, it is perceived as an unshared link or circuit of the chosen service. PW3 specifies the encapsulation, transport, control, management, interworking and security of services emulated over PSNs. To maximize the return on their assets and minimize their operational costs, many service providers are looking to consolidate the delivery of multiple service offerings and traffic types onto a single IP-optimized network. PWE3 is a possible solution since it emulates Ethernet frame formats over IP networks.
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1.5.8 synchronization
As the network moves toward Ethernet as the transport technology of choice, synchronization remains a major issue. As Ethernet and TDM technologies continue to coexist, technologies like circuit-emulation services (CES) provide capabilities to map TDM traffic on Ethernet infrastructure and vice versa, enabling a smooth changeover for network operators transitioning to an all-packet network. To interconnect these two technologies, frequency synchronization is key since the TDM technologies have frequency-offset tolerances that are much more restrictive than the asynchronous Ethernet technologies. Ethernet relies on inexpensive holdover oscillators and can stop transmitting traffic or buffer data, while TDM technologies rely on the continuous transmission and presence of synchronization reference. Synchronous Ethernet solves these issues by ensuring frequency synchronization at the physical level. However, since SyncE is a synchronization technology based on layer 1, it requires that all ports on the synchronized path be enabled for SyncE. Any node that is non SyncE-enabled on the path will automatically break the synchronization from this node. This is an issue for network providers that have a multitude of Ethernet ports between the primary synchronization unit and the edge device that needs synchronization as all the ports must be SyncEenabled to synchronize to the edge. Such requirements can increase the cost of deployments as hardware and software upgrades can dramatically increase the total cost of ownership. SyncE also only focuses on frequency synchronization and does not guarantee phase synchronizationalthough the phase requirements can be somewhat assessed via SyncE.
Many services need synchronization, but wireless base stations today have the largest stake in frequency and time distribution. The frequency stability of the air interface between the cell tower and the handset supports handing off a call between adjacent base stations without interruption. Synchronization for base stations is therefore central to the QoS that an operator provides.
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The next packet synchronization technology, the Precise Time Protocol (PTP) also referred to as the IEEE 1588v2, is specifically designed to provide high clock accuracy through a packet network via a continuous exchange of packets with appropriate timestamps. In this protocol, a highly precise clock source, referred to as the grand-master clock generates timestamp announcements and responds to timestamp requests from boundary clocks, thus ensuring that the boundary clocks and the slave clocks are precisely aligned to the grand-master clocks. By relying on the handover capability and the precision of the integrated clocks in combination with the continuous exchange of timestamps between PTP-enabled devices, frequency and phase accuracy can be maintained at a sub-microsecond range, thus ensuring synchronization within the network. In addition to frequency and phase synchronization, ToD synchronization can also ensure that all PTPenabled devices are synchronized with the proper time, based on coordinated universal time (UTC). The great advantages of PTP is that as a packed-based technology, only boundary and slave clock needs to be aware of the nature of the packets and therefore synchronization packets are forwarded as any other data packets within the network. This flexibility reduces the cost of ownership as the main upgrade to the networks are limited to synchronization equipment contrarily to the SyncE approach that requires both synchronization equipment and upgrade of all Ethernet ports on the link to SyncE specifications.
This diagram shows an application of IEEE 1588v2 PTP in a mobile backhaul to establish synchronization.
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autHOrS
Sylvain cOrnay, marketing manager, exfO HammadOun dicKO, product Specialist, exfO tHiernO diallO, product Specialist, exfO SOpHie legault, product line manager, exfO Sue Judge, consultant exfO inc. march 2011
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Also coming soon to the Carrier Ethernet Basic Educational Series, modules that will focus on the following aspects of Carrier Ethernet, including service turn-up, service monitoring and troubleshooting.
The increasing demand for data-centric services driven by smartphone technologies, along with the growth of social networking (e.g., Facebook, Twitter, etc.) and multimedia applications (e.g., gaming, YouTube, etc.), has prompted operators to shift toward packetbased Ethernet/IP technologies across their access and core networks in order to costeffectively support the rapidly escalating bandwidth requirements. In fact, according to the latest report from Ciscos Visual Networking Index Global Mobile Data Traffic Forecast, it is predicted that global mobile data traffic will grow by 26 times between 2010 and 2015, to 6.3 exabytesone billion gigabytesper month. Additionally, the report foresees that by 2015, two-thirds of all mobile data traffic will be video, underscoring the challenges operators face as they try to manage the tidal wave of mobile data set to flood their networks. While there is little doubt about the cost efficiencies and scalability of Ethernet, the time- and delay-sensitive nature of established voice services, in addition to the growing popularity of new mobile video services, cannot be ignored and necessitate an advanced approach to Ethernet networking and testing to maintain the customer expectations for quality of service (QoS) and quality of experience (QoE). As Carrier Ethernet technology matures, networks will eventually become entirely packetbased; this will greatly simplify the network architecture, reduce costs and provide the necessary scalability for expected growth with data-centric applications. But as the network infrastructure evolves to support packet-based transmission, operators must also evolve from only managing network performance to also managing service performance. This means that testing the network with a simple ping is no longer an option as operators must now constantly validate and measure the key performance indicators (KPIs) on a per-service basis.
2.1
rfc-2544
The Internet Engineering Task Forces (IETFs) RFC 2544 is a benchmarking methodology for network interconnect devices. This request for comment (RFC) was created in 1999 as a methodology to benchmark network devices, such as hubs, switches and routers, as well as to provide accurate and comparable values for comparison and benchmarking. RFC 2544 provides engineers and network technicians with a common language and results format. RFC 2544 describes the following six subtests: Throughput: This test measures the maximum rate at which none of the offered frames is dropped by the device/system under test (DUT/SUT). This measurement translates into the available bandwidth of the Ethernet virtual connection.
Back-to-back or burstability: This test measures the longest burst of frames at maximum throughput or minimum legal separation between frames that the device or network under test will handle without any loss of frames. This measurement is a good indication of the buffering capacity of a DUT. Frame loss: This test defines the percentage of frames that should have been forwarded by a network device under steady state (constant) loads that were not forwarded due to lack of resources. This measurement can be used for reporting the performance of a network device in an overloaded state, as it can be a useful indication of how a device would perform under pathological network conditions, such as broadcast storms. Latency: This test measures the round-trip time of a test frame to travel through a network device or across the network and back to the test port. Latency is the time interval that begins when the last bit of the input frame reaches the input port and ends when the first bit of the output frame is seen on the output port. It is the time taken by a bit to go through the network and back. Latency variability can be a problem. With protocols like voice over Internet protocol (VoIP), a variable or long latency can cause degradation in voice quality. System reset: This test measures the speed at which a DUT recovers from a hardware or software reset. This subtest is performed by measuring the interruption of a continuous stream of frames during the reset process. System recovery: This test measures the speed at which a DUT recovers from an overload or oversubscription condition. This subtest is performed by temporarily oversubscribing the device under test and then reducing the throughput at normal or low load while measuring frame delay in these two conditions. The difference between delay at overloaded conditions and the delay and low-load conditions represent the recovery time. From a laboratory and benchmarking perspective, the RFC 2544 methodology is an ideal tool for automated measurement and reporting. From a service turn-up and troubleshooting perspective, RFC 2544, although acceptable and valid, does have some drawbacks:
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Service providers are shifting from only providing Ethernet pipes to enabling services. Networks must support multiple services from multiple customers, while each service has its own performance requirements that must be met even under full load conditions and with all services being processed simultaneously. RFC 2544 was designed as a performance tool with a focus on a single stream to measure maximum performance of a DUT or network under test and was never intended for multiservice testing. With RFC 2544s focus on identifying the maximum performance of a device or network under test, the overall test time is variable and depends heavily on the quality of the link and subtest settings. RFC 2544 test cycles can easily require a few hours of testing. This is not an issue for lab testing or benchmarking, but becomes a serious issue for network operators with short service maintenance windows. Packet delay variation is a KPI for real-time services such as VoIP and Internet protocol television (IPTV) and is not measured by the RFC 2544 methodology. Network operators that performed service testing with RFC 2544 must typically execute external packet jitter testing outside of RFC 2544as this KPI is not defined or measured by the RFC 2544. Testing is performed sequentially on one KPI after another. In todays multiservice environments, traffic is going to experience all KPIs at the same time, although throughput might be good, it can also be accompanied by very high latency due to buffering. Designed as a performance assessment tool, RFC 2544 measures each KPI individually through its subtest and therefore cannot immediately associate a very high latency with a good throughput, which should be a cause for concern.
2.2.3 methodology
The ITU-T Y.1564 is built around two key subtests, the service-configuration test and the service-performance test, which are performed in order:
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Bidirectional test
EtherSAM can perform round-trip measurements with a loopback device. In this case, the results reflect the average of both test directions, from the test set to the loopback point and back to the test set. In this scenario, the loopback functionality can be performed by another test instrument in Loopback mode or by a network interface device in Loopback mode. The same test can also be run in simultaneous bidirectional mode (dual test set). In this case, two test sets, one designated as local and the other as remote, are used to communicate and independently and run tests per direction. The tests are performed simultaneously as well. This provides much more precise test results such as independent assessment per direction and the ability to quickly determine which direction of the link is experiencing failure. This allows service providers to test asymmetrical links. This test uncovers more configuration errors than the EtherSAM test with one loopback device on the other end especially when testing multiple services with different EIRs and CIRs.
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Burst test methodologies assume a token bucket algorithm to police and shape the traffic. The token bucket is a control mechanism that dictates when traffic can be transmitted, based on the presence of tokens in the bucketan abstract container that holds aggregate network traffic to be transmitted.
The burst test is provided for non-color aware and color aware applications. In advanced business Ethernet services, it is possible to have traffic that is tagged with different colors (green and yellow) within the same service. The colors are a method allowing the end customer to tell the network that specific traffic has higher priority in case of congestion. Color Aware mode is also offered only in more complex/advanced Ethernet Services. Color mode testing consists in verifying that the traffic policers and shapers properly respects the Color mode. Testing Color mode is a complex test and will not be tested frequently in the field. On the other hand Non-Color Aware mode requires only one color per service. It is the most common mode.
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2.2.5 metrics
Y.1564 focuses on the following KPIs for service quality: Bandwidth: This is a bit rate measure of the available or consumed data communication resources expressed in bits/second or multiples of it (kilobits/s, megabits/s, etc.). Frame transfer delay (FTD): Also known as latency, this is a measurement of the time delay between the transmission and the reception of a frame. Typically this is a round-trip measurement, meaning that the calculation measures both the near-end to far-end and far-end to near-end directions simultaneously. Frame delay variations: Also known as packet jitter, this is a measurement of the variations in the time delay between packet deliveries. As packets travel through a network to their destination, they are often queued and sent in bursts to the next hop. There may be prioritization at random moments, also resulting in packets being sent at random rates. Packets are therefore received at irregular intervals. The direct consequence of this jitter is stress on the receiving buffers of the end nodes where buffers can be overused or underused when there are large swings of jitter. Frame loss: Typically expressed as a ratio, this is a measurement of the number of packets lost over the total number of packets sent. Frame loss can be due to a number of issues such as network congestion or errors during transmissions.
2.2.6 Benefits:
ITU-T Y.1564 provides numerous benefits to service providers, offering mobile backhaul, commercial and wholesale Ethernet services. Measurement of all KPIs in a single test: While existing methodologies like RFC 2544 only provide the capability to measure the maximum performances of a link, EtherSAM uses a validation approach where KPIs are measured and compared to the expected values for each service. This approach focuses on proving that KPIs are met while in guaranteed traffic conditions. Significantly faster: The RFC 2544 methodology uses a sequential approach where each subtest is executed one after the other until they have all been completed, making it a timeconsuming procedure. Additionally, the completion of a subtest heavily relies on the quality of the link. In opposition, ITU-T Y.1564 uses a defined ramp-up approach where each step takes an exact amount of time. Link quality issues are quickly identified without necessarily increasing test time because a pass/fail condition is based on the KPI assessment during the step. Multiservice testing capabilities: As described earlier, the majority of Ethernet services deployed today include multiple classes of service within the same connection. A major drawback of RFC 2544 is the fact that it can only test a single service at a time; Y.1564 on the other hand can test multiple services simultaneously. 11
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More representative of real-life conditions: The worst-case scenario for a network is handling multiple traffic types during a congestion period. Since RFC 2544 only tests a single stream at a time, it cannot simulate worst-case scenarios. The ITU-T Y.1564 service subtest can generate all configured services at the same time, providing the ability to stress network elements and data paths in worst-case conditions. The service test provides powerful test results since all KPIs are measured simultaneously for all services with clear pass/fail indication, as well as identification of failed KPIs. This ensures that any failure or inconsistency is quickly pinpointed and reported, again contributing to an efficient and more meaningful test cycle.
2.4
Synchronization
Synchronization can be defined as the coordinated and simultaneous relationship between time-keeping among multiple devices. For people outside of the telecom world, synchronization typically refers to time synchronization where one or more devices have the same time as a reference clock, typically the universal time clock (UTC); when synchronized, two devices will have the proper time of day (ToD) in reference to the universal time reference, regardless of their geographical location. However for network engineers, synchronization has a very precise and critical use. Telecom networks, such as SONET and SDH networks, are based on a synchronous architecture, meaning that all data signals are synchronized and clocked using virtually the same clock throughout. This ensures that all of the ports that carry data do so at the same frequency or with very little offset, and therefore, network throughput is deterministic and fixed for a specific transport rate. Ethernet on the other hand is an asynchronous technology where each Ethernet port has its own independent clock circuit and oscillator. Because each port is clock independent, frequency offsets between interconnected ports can be relatively high. To solve this issue, Ethernet devices typically implement buffers that can store traffic and then mitigate the effect of offsets between two ports. Therefore, telecom networks require two other types of synchronization in addition to time synchronization, that is, frequency synchronization and phase synchronization.
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Frequency synchronization is typically a physical synchronization where the output clocks between devices is synchronized. When two devices are frequency synchronized, they basically generate the same number of bits over an integration period (typically 1 second). When they are not frequency synchronized, one device will generate more bits per second than the other, which can cause overflow and eventually bit errors or traffic loss. Phase synchronization refers to the simultaneous variation of clocks between devices. When phase-synchronized, the two devices will shift at exactly the same time from one clock pulse to the other. A real-world example would be to compare two watches side-by-side. When synchronized, these two watches will increment at exactly the same time; when unsynchronized, one device will count faster than the other and in the network world, these variations are the equivalent of phase offset.
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Precise Time Protocol (PTP) The 1588v2 standard, defined by the ITU and also known as Precise Time Protocol, (PTP) is specifically designed to provide high clock accuracy through a packet network via a continuous exchange of packets with appropriate timestamps. In this protocol, a highly precise clock source, referred to as the grand-master clock generates timestamp announcements and responds to timestamp requests from boundary clocks, thus ensuring that the boundary clocks and the slave clocks are precisely aligned to the grand-master clocks. By relying on the holdover capability and the precision of the integrated clocks in combination with the continuous exchange of timestamps between PTP-enabled devices, frequency and phase accuracy can be maintained at a sub-microsecond range, thus ensuring synchronization within the network. In addition to frequency and phase synchronization, ToD synchronization can also ensure that all PTP-enabled devices are synchronized with the proper time, based on the coordinated universal time clock (UTC). The advantage of PTP is that as a packed-based technology, only the boundary and slave clock need to be aware of the nature of the packets and therefore, synchronization packets are forwarded as any other data packets within the network. This flexibility reduces the cost of ownership as the main upgrade to the networks are limited to synchronization equipment, contrarily to the SyncE approach which requires both synchronization equipment and upgrade of all Ethernet ports on the link to SyncE specifications. However, the major weakness of PTP is due to its packet nature. As the synchronization packets used by PTP are forwarded in the network between grand master and hosts, they are subject to all the network events, such as frame delay (latency), frame-delay variation (packet jitter) and frame loss. Even with the best practice of applying high priority to synchronization flows, these synchronization packets still experience congestion and possible routing and forwarding issues, such as out-of-sequence and route flaps. This means that the host clocks holdover circuit must be stable enough to maintain synchronization where the synchronization packets experienced network events.
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