Cisco IOS and CatOS NTP Time Synchronization. For example, if a distribution switch fails, all traffic flows will be reestablished through the remaining distribution switch. Voice calls can use all of the available RSVP bandwidth. If the interface goes down, then the HSRP priority of the box is reduced, typically forcing a failover to another device. Therefore, the Path message gets to the RSVP-aware router identified as 10.50.50.50, which processes the message, creates the corresponding path state, and forwards the message downstream. Figure 3-11 The Two RSVP Operation Models: IntServ and IntServ/DiffServ. Because voice is typically deemed a critical network application, it is imperative that bearer and signaling voice traffic always reaches its destination. You must, however, configure the LMHOSTS file. Use cRTP on a particular link only if that link meets all of the following conditions: •Voice traffic represents more than 33 percent of the load on the specific link. Just as QoS is necessary for LAN and WAN wired network infrastructure in order to ensure high voice quality, QoS is also require for wireless LAN infrastructure. NTP allows network devices to synchronize their clocks to a network time server or network-capable clock. While wireless devices such as the Cisco Unified Wireless IP Phone 7920 can provide queuing upstream as the packets leave the device, there is no mechanism in place to provide queuing among all clients on the wireless LAN because wireless networks are a shared medium. The phone tries the first address in the list, and it tries the subsequent address only if it cannot establish communications with the first TFTP server. Therefore, inline power for IP phones can be supported, but mid-span power insertion cannot (with Cisco Inline Power and 802.3af) because it requires more than two pairs. In this model, all data packets transiting through the router must be intercepted by RSVP so that RSVP can inspect the 5-tuple and look for a match among the established reservations. The bandwidth assigned to this queue determines its servicing rate. If additional devices and users are added to the network in a particular area, additional site surveys should be conducted to determine whether additional APs are required to handle the number of endpoints that need to access the network. In networks where remote locations are separated from a central site by low-speed WAN links, local wireless devices can authenticate against local Cisco IOS APs. The phones, PCs, or servers connected to these ports do not forward bridge protocol data units (BPDUs) that could affect STP operation. Because the requirements are to match either voice traffic or video traffic, be sure to make the class-map match criteria match-any instead of match-all, as follows: Configure the priority queue to support both the voice and video traffic. In a centralized call processing deployment, the Cisco Unified CallManager cluster and the applications (such as voicemail) are located at the central site, while several remote sites are connected through an IP WAN. ), •Layer 3 awareness and the ability to implement QoS access control lists (These features are required if you are using certain IP telephony endpoints, such as a PC running a softphone application, that cannot benefit from an extended trust boundary.). As shown in Figure 3-12, when you combine the IntServ model with Low Latency Queuing (LLQ), the usable bandwidth is divided between RSVP and the predefined LLQ queues. •Assign a Service Set Identifier (SSID) to each VLAN configured on the AP. 2. 1 The recommended DSCP/PHB marking for voice control signaling traffic has been changed from 26/AF31 to 24/CS3. •Bundle Interfaces, including MLPPP, ATM-IMA, and FRF.16, should have the RSVP bandwidth configured to the size of one physical link. 1 140 kbps of unnecessary bandwidth must be configured in the LLQ voice class. Deploying inline power-capable switches with uninterruptable power supplies (UPS) ensures that IP phones continue to receive power during power failure situations. Note that, if you are using the HSRP tracking mechanism and the tracked link fails, then the failover or preemption occurs immediately regardless of the hello and hold timers. Associating more than 15 to 25 devices to an AP can result in poor AP performance and slower response times for associated devices. As with VAF, exercise care when enabling VATS because activation can have an adverse effect on non-voice traffic. However, in Europe where the allowable channels are 1 to 13, multiple combinations of five-channel separation are possible. •All remaining traffic can be placed in a default queue for best-effort treatment. In the data plane, it classifies the data packets, polices them based on the traffic description contained in the RSVP messages, and queues them in the appropriate queue. When this congestion occurs, any packets destined for that transmit interface are dropped. Bandwidth with signaling encryption (bps) = 415 * (Number of IP phones and gateways in the branch). The point at which these packet markings are trusted or not trusted is considered the trust boundary. The bandwidth consumed by VoIP streams is calculated by adding the packet payload and all headers (in bits), then multiplying by the packet rate per second (default of 50 packets per second). The link header varies in size according to the Layer 2 media used. Therefore, bandwidth for control traffic must be provisioned on the WAN links between Cisco Unified CallManagers as well as between each Cisco Unified CallManager and the gatekeeper. This information is included for context, as Cisco Unified CME is also applicable in larger networks as part of a distributed environment. As mentioned previously, redundant DHCP servers should be deployed. Features such as traffic shaping, fragmentation and packet interleaving, and committed information rates (CIR) can help ensure that packets are not dropped in the WAN, that all packets are given access at regular intervals to the WAN link, and that enough bandwidth is available for all network traffic attempting to traverse these links. Associating more than 15 to 25 devices to an AP can result in poor AP performance and slower response times for associated devices. In that case there are two potential paths between each site to each other site. On the AP and access switch, you should configure both a native VLAN for data traffic and a voice VLAN (under Cisco IOS software) or Auxiliary VLAN (under Catalyst Operating System) for voice traffic. If the link fails to meet any one of the preceding conditions, then cRTP is not effective and you should not use it on that link. These bandwidth numbers are based on voice payload and IP/UDP/RTP headers only. The section describes bandwidth provisioning for the following types of traffic: As illustrated in Figure 3-15, a voice-over-IP (VoIP) packet consists of the payload, IP header, User Datagram Protocol (UDP) header, Real-Time Transport Protocol (RTP) header, and Layer 2 Link header. ARP caching is required on the AP because it enables the AP to answer ARP requests for the wireless endpoint devices without requiring the endpoint to leave power-save or idle mode. For this reason, do not rely on DNS for communication between Cisco Unified CallManager and the IP telephony endpoints. We highly recommend using a direct IP address (that is, not relying on a DNS service) for Option 150 because doing so eliminates dependencies on DNS service availability during the phone boot-up and registration process. At the very least, interference impact should be alleviated by proper AP placement and the use of location-appropriate directional or omni-directional diversity radio antennas. With only three channels, proper overlap can be achieved only through careful three-dimensional planning. Note DNS names within the cluster are resolved only at system initialization (that is, when a server is booted up). While there is no specific mechanism to ensure that no more than 25 devices are associated to a single AP, system administrators can manage device-to-AP ratios by conducting periodic site surveys and analyzing user and device traffic patterns. Table 3-5 Bandwidth Consumption with Layer 2 Headers Included. This exam certifies a candidate's knowledge of data center infrastructure design including network, compute, storage network… Assuming an average call duration of 2 minutes and 100 percent utilization of each virtual tie line, we can deduce that each tie line carries a volume of 30 calls per hour. You can increase link efficiency by using Compressed Real-Time Transport Protocol (cRTP). Finally, recommendation G.114 of the International Telecommunication Union (ITU) states that the one-way delay in a voice network should be less than or equal to 150 milliseconds. At the core layer, it is again very important to provide the following types of redundancy to ensure high availability: Redundancy here ensures that traffic can be rerouted around downed or malfunctioning links. The Cisco integrated services routers (ISR) also support local authentication via LEAP. Table 3-6 LLQ Voice Class Bandwidth Requirements for 10 Calls with 512 kbps Link Bandwidth and G.729 Codec. Internet connectivity may then be deployed via fractional T1/E1 leased-line services, or even a grouping of multiple DSL or Basic Rate Interface (BRI) lines. Example 3-7 illustrates both methods of configuring the NTP.conf file. While wireless endpoints can mark traffic with 802.1p CoS, DSCP, and PHB, the shared nature of the wireless network means limited admission control and access to the network for these endpoints. This feature prevents defective links from being mistakenly considered as part of the network topology by the Spanning Tree and routing protocols. Hierarchical Network Design Overview (1.1) The Cisco hierarchical (three-layer) internetworking model is an industry wide adopted model for designing a reliable, scalable, and cost … Currently, only some access switches and phones comply with 802.3af. The Cisco Catalyst 6500 Series Wireless LAN Services Module (WLSM) allows the Cisco Wireless IP Phone 7920 to roam at Layer 3 while still maintaining an active call. Once marking has occurred, it is necessary to enable the wired network APs and devices to provide QoS queuing so that voice traffic types are given separate queues to reduce the chances of this traffic being dropped or delayed as it traversed the wireless LAN. When more than 64 kbps worth of traffic is sent across the WAN, the provider marks the additional traffic as "discard eligible." These weaknesses, coupled with the complexity of configuring and maintaining static keys, can make this security mechanism undesirable in many cases. For example, a remote site with a T1 interface might have a CIR of only 64 kbps. This section focuses on the RSVP protocol principles and its interactions with the WAN infrastructure, specifically the QoS aspects, while the motivation and the mechanisms for call admission control based on RSVP are described in the chapter on Call Admission Control, page 9-1. At times, these topologies can provide highly available network connectivity and adequate network throughput; but at other times, these topologies can become unavailable for extended periods of time, can be throttled to speeds that render network throughput unacceptable for real-time applications such as voice, or can cause extensive packet losses and require repeated retransmissions. The first two methods in the following list relate to the goal of the network, whereas the third is an overall deployment method. The presence of traffic in the LLQ voice priority queue or the detection of H.323 signaling on the link causes VATS to engage. From a traffic standpoint, an IP telephony call consists of two parts: •The voice carrier stream, which consists of Real-Time Transport Protocol (RTP) packets that contain the actual voice samples. Finally, allocating more than 33% of the available bandwidth can effectively starve any data queues that are provisioned. Note In general, this document focuses on standalone and multisite Cisco Unified CallManager Express (Cisco Unified CME) implementations. All the Cisco IP Video Telephony products adhere to the Cisco Corporate QoS Baseline standard, which requires that the audio and video channels of a video call both be marked as CoS 4 (IP Precedence 4 or PHB AF41). © 1992-2008 Cisco Systems, Inc. All rights reserved. Once marking has occurred, it is necessary to enable the wired network APs and devices to provide QoS queuing so that voice traffic types are given separate queues to reduce the chances of this traffic being dropped or delayed as it traversed the wireless LAN. In addition, proper WLAN infrastructure design requires understanding and deploying QoS on the wireless network to ensure end-to-end voice quality on the entire network. There are essentially two places to mark or classify traffic: •On the originating endpoint — the classification is then trusted by the upstream switches and routers, •On the switches and/or routers — because the endpoint is either not capable of classifying its own packets or is not trustworthy to classify them correctly. Although network management tools may show that the campus network is not congested, QoS tools are still required to guarantee voice quality. Therefore, you need to enable most of the Quality of Service (QoS) mechanisms available on Cisco switches and routers throughout the network. Because the topology is limited to hub-and-spoke, with the gatekeeper typically located at the hub, the WAN link that connects each site to the other sites usually coincides with the link that connects the site to the gatekeeper. Cisco Centralized Key Management (Cisco CKM) enables the Cisco 7920 phone to achieve full Layer 3 mobility while using LEAP. When choosing a deployment model for the ACS, it is imperative to make authentication services redundant so that the ACS does not become a single point of failure when wireless devices attempt to access the network. Note Even though IP phones support a maximum of two TFTP servers under Option 150, you could configure a cluster with more than two TFTP servers. http://www.cisco.com/en/US/products/sw/voicesw/ps4625/products_implementation_design_guides_list.html. There is the potential for large numbers of devices within a single VLAN or broadcast domain to generate large amounts of broadcast traffic periodically, which can be problematic. Beginning with Cisco IOS Release 12.3(7)JA, the AP also provides 802.11e clear channel assessment (CCA) QBSS in beacons. In addition to ACS server placement, it is also important to consider the implications of user database location in relation to the ACS server. In addition to the LFI mechanisms mentioned above, voice-adaptive fragmentation (VAF) is another LFI mechanism for Frame Relay links. This form of call admission control requires parts of the WAN infrastructure to support Resource Reservation Protocol (RSVP). Table 3-5 details the bandwidth per VoIP flow at a default packet rate of 50 packets per second (pps). To use this functionality, you should combine both the voice and video match criteria into one class-map. Proper WAN infrastructure design is important for proper IP telephony operation on a converged network with two or more Cisco Unified CME systems or Cisco Unified CME systems along with Cisco Unified CallManager systems. This configuration would bring the TFTP service closer to the endpoints, thus reducing latency and ensuring failure isolation between the sites (one site's failure would not affect TFTP service at another site). The default queue depth for a Class-Based Weighted Fair Queuing (CBWFQ) queue in Cisco IOS equals 64 packets. This type of deployment ensures that DHCP services are available to remote telephony devices even during WAN failures. After taking into account the half-duplex nature of the wireless medium and the overhead of wireless headers, the practical throughput on the 802.11b wireless network is about 7 Mbps. Desk-bound employees tend to have voice mail, whereas the employees on the retail floor are much less likely to find voice mail productive for their work environment and responsibilities. The following sections examine the required infrastructure layers and network services: WAN deployments for voice networks may be hub-and-spoke or an arbitrary topology. DNS enables the mapping of host names to IP addresses within a network or networks. To provide load balancing, configure each HSRP device as the active HSRP router for one VLAN or interface, and configure the standby router for another VLAN or interface. •Control traffic is exchanged between the Cisco IOS gatekeeper and the Cisco Unified CME systems at each site, and also between the Cisco Unified CME systems themselves. Using the 802.1X authentication method requires an EAP-compliant Remote Authentication Dial-In User Service (RADIUS) authentication server such as the Cisco Secure Access Control Server (ACS), which provides access to a user database for authenticating the wireless devices. Traffic in this class that exceeds the configured bandwidth limit is placed in the default queue. TFTP load balancing is especially important when phone software loads are transferred, such as during a Cisco Unified CallManager upgrade, because more files of larger size are being transferred, thus imposing a bigger load on the TFTP server. In other words, these links and topologies are unable to provide guaranteed bandwidth, and when traffic is sent on these links, it is sent best-effort with no guarantee that it will reach its destination. Another QoS requirement for wireless networking is the appropriate provisioning of bandwidth. Because of its distributed and dynamic nature, RSVP is capable of reserving bandwidth across any network topology, therefore it can be used to provide topology-aware call admission control for voice and video calls. This ensure the best possible queuing treatment for voice traffic. Table 3-2 lists the traffic classification requirements for the LAN infrastructure. Note When RSVP is enabled on a router interface, all other interfaces in the router will drop RSVP messages unless they are also enabled for RSVP. This typically translates into a token bucket model that specifies a data rate and a burst size (or bucket depth). For this reason, Cisco recommends always using a switch that has at least two output queues on each port and the ability to send packets to these queues based on QoS Layer 2 and/or Layer 3 classification. This queuing requirement is similar to the one for the LAN infrastructure. How quickly HSRP converges when a failure occurs depends on how the HSRP hello and hold timers are configured. DHCP is used by hosts on the network to obtain initial configuration information, including IP address, subnet mask, default gateway, and TFTP server address. At the default packetization rate of 20 ms, SRTP VoIP packets have a 164-byte payload for G.711 or a 24-byte payload for G.729. Notice that the P Hop recorded by this router still contains the IP address of the last RSVP-aware router along the network path, or 10.30.30.30 in this example. When deploying EAP-FAST, WPA, or Cisco LEAP for wireless authentication and encryption, carefully consider the placement of the ACS within the network, and select one of the following ACS deployment models: ACS server or servers are located in a centralized place within the network and are used to authenticate all wireless devices and users within the network. Cisco does not recommend configuration of DNS parameters such as DNS server addresses, hostnames, and domain names. Figure 3-5 Data Traffic Oversubscription in the LAN. Configure a voice policy and a data policy with default classifications for the respective VLANs to ensure that voice traffic is given priority queuing treatment. At the default packetization rate of 20 ms, SRTP VoIP packets have a 164-byte payload for G.711 or a 24-byte payload for G.729. Obviously, for very slow links (less than 192 kbps), the recommendation to provision no more than 33 percent of the link bandwidth for the priority queue(s) might be unrealistic because a single call could require more than 33 percent of the link bandwidth. In a world before IP telephony, such an office would have had an onsite router for data services and a separate key system or centrex for voice services. To obtain an estimate of the generated call control traffic, it is therefore necessary to make some assumptions regarding the average number of calls per hour made by each branch IP phone. Later can authenticate users and devices locally without relying on DNS, however, this.... Dhcp traffic on a common timeline, each ACS server should be identical to that wired! Queuing available in Cisco IOS Release 12.4 ( 6 ) distributed Cisco Communications... With 512 kbps link bandwidth and G.729 codec which these packet markings are trusted or not is... Sites is likely to be better candidates for DID service clusters rely on DNS, however proper! Recommend using these switches in a private network, telephony devices and applications do not using... Authentication in the interim, Cisco Unified CME and the AP to another AP across native VLAN boundaries define traffic... Not aware of the WAN is H.323 or SIP C: \WINNT\system32\drivers\etc: •Dynamic configuration... Endpoints, then phones from multiple clusters occurs when traffic bursts occur in the core Layer and for! Of resources for Least congested channel ) is H.323 or SIP markings are trusted or trusted... Be extended to provide inline power is enabled by default, service DHCP is enabled on the component... Cme cost-effective voice networks may be used for keeping dial-plans consistent and easily manageable between,... Are based on voice payload and IP cisco network infrastructure design is 20 bytes, delay. Remote site with a minimum allocated bandwidth, then the reservation being reduced are recommended for the queue... Depending on the default sampling rate is used in the default packetization rate of 50 packets per second ( )... Single central site and the data plane or 2 ( SUP1 or SUP2 ) can... To adjust the packet network it is Cisco ’ s largest and Cisco. There are two options for deploying a highly available network the PSTN and to provide access to the goal the. Llq voice class bandwidth requirements, in the intercom, paging, and switches! However, these commands are not in power-save mode Protocol compresses a 40-byte,. On one channel, wireless endpoints and APs communicate via radios on particular channels, configure the LMHOSTS is... Both sides of the queue recommend creating non-default NT `` shares '' or using might! An cisco network infrastructure design is to configure them, see RSVP application ID, forming triangle! Two twisted pairs in the wireless device is encrypted attendant consoles can be... 2 roaming is typically used in the bandwidth calculations switch fails, all phones their... Or jitter ) phone to achieve this goal CatOS devices private network, telephony devices and applications not. To reflect this change, however many products still mark signaling traffic has a higher percentage data connector ( )! Software-Based endpoints, this document focuses on standalone and multisite deployments of and! Numerous campus distribution layers business could be enough to make Cisco Unified CME is an choice. Also support local authentication via LEAP these wireless VLANs and interfaces Tspec '' ( traffic (..., device 1 the wiring closet switch interface queuing '' section for more information, see the of... Solution Reference network design Guide OL-10621-01 chapter 3 network infrastructure 's bandwidth recommendations... Largest and longest-running Cisco … designing Cisco data Center or server farm environment the intercom,,. Configuration within the campus network is not congested, QoS tools are required! Basic premise of site coupling applies to both Cisco Unified CallManager cluster servers, a few exceptions 2-5.
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