Proper AP selection, deployment, and configuration are essential to ensure that the wireless network handles voice traffic in a way that provides high-quality voice to the end users. Call signaling for voice and videoconferencing is now classified as CoS 3 (IP Precedence 3, PHB CS3, or DSCP 24) but was previously classified as PHB AF31 or DSCP 26. Example 3-7 illustrates both methods of configuring the NTP.conf file. A retail organization has comparatively few desk-bound employees, whereas a bank or insurance company has a higher percentage. Cisco Unified IP Phones adhere to the conditions of the DHCP lease duration as specified in the DHCP server's scope configuration. This motto implies careful planning of the LAN infrastructure so that the available bandwidth is always considerably higher than the load and there is no steady-state congestion over the LAN links. Cisco does not recommend configuration of DNS parameters such as DNS server addresses, hostnames, and domain names. SSIDs enable endpoints to select the wireless VLAN they will use for sending and receiving traffic. When troubleshooting or managing a telephony network, it is crucial to synchronize the time stamps within all error and security logs, traces, and system reports on devices throughout the network. When troubleshooting or managing a telephony network, it is crucial to synchronize the time stamps within all error and security logs, traces, and system reports on devices throughout the network. Mobile devices typically use IP addresses for short increments of time and then might not request a DHCP renewal or new address for a long period of time. Note that, if the 64-packet limit is reached, any subsequent packets are either assigned to the best-effort queue or are dropped. All other traffic on the wireless network should be marked as best-effort or with some intermediary classification as outlined in wired network marking guidelines. Finally, each link between the core and distribution devices should belong to its own VLAN or subnet and be configured using a 30-bit subnet mask. All primary TFTP servers would be configured to write their configuration files to TFTP1_P, while all the secondary TFTP servers would write theirs to TFTP1_S, as follows: •TFTP2_P's alternate file location is set to: \\TFTP1_P\Program Files\Cisco\TFTPpath\TFTP2. Currently, only some access switches and phones comply with 802.3af. Proper LAN infrastructure design requires following basic configuration and design best practices for deploying a highly available network. •Match the transmit power on the AP to that on the wireless voice endpoints. This method ensures that, when voice traffic is being sent on the WAN interface, large packets are fragmented and interleaved. Figure 3-2 Standalone Office Network Topology. The entrance criterion for this queue is a DSCP value of 24 or a PHB value of CS3. Therefore, it is important to ensure that the RSVP-enabled traffic is marked with the appropriate DSCP value and that the bandwidth of the corresponding PQ or CBWFQ queues is provisioned to accommodate both RSVP-enabled traffic and all other traffic. This condition can be extremely problematic for real-time multicast applications such as music on hold and streaming video. All that is required is re-authentication, if Cisco LEAP or Extensible Authentication Protocol (EAP) is used, and the passing of Inter-Access Point Protocol (IAPP) messages between the last AP and the new AP to indicate that the endpoint has roamed. Note Beginning with Cisco IOS Release 12.3(7)JA, the AP sends 802.11e CCA-based QBSS. It should also be noted that, beginning in Cisco IOS Release 12.2(13)T, cRTP can be configured as part of the voice class with the Class-Based cRTP feature. Medium Branch Office Design The distribution layer of the Campus LAN includes the portion of the network from the wiring closet switches to the next-hop switch, and it is the first Layer-2-to-Layer-3 traversal in the LAN. The classification that RSVP performs is based on the 5-tuple consisting of the source IP address, source port, destination IP address, destination port, and protocol number. When deploying voice in a WAN environment, Cisco recommends that you use the lower-bandwidth G.729 codec for any voice calls that will traverse WAN links because this practice will provide bandwidth savings on these lower-speed links. Figure 3-5 Data Traffic Oversubscription in the LAN. Note With the introduction of RSTP 802.1w, features such as PortFast and UplinkFast are not required because these mechanisms are built in to this standard. Another common configuration is to allow traffic bursts above the CIR, which represents the rate that the service provider has guaranteed to transport across its network with no loss and low delay. •Centralized DHCP Server and Remote Site Cisco IOS DHCP Server. In that case there are two potential paths between each site to each other site. To resolve this inability to calculate the exact overhead ratio for video, Cisco recommends that you add 20% to the call speed regardless of which type of Layer-2 medium the packets are traversing. This 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. (See Application ID Call Admission Control Model.) In addition, objects and obstructions can cause signal reflection and multipath distortion. If you specify the keyword fair, the queuing algorithm will be weighted fair queuing (WFQ). In this scenario, each remote or spoke site is one WAN link hop away from the central or hub site and two WAN link hops away from all other spoke sites. Layer 2 roaming is typically unnoticeable to the end user. –All the non-RSVP traffic destined for the PQ can be deterministically limited to a certain amount by an out-of-band call admission control mechanism (such as Cisco Unified CallManager locations or a Cisco IOS gatekeeper). In this situation, the TFTP server whose address is provided to all phones in the subnet or VLAN must answer the file transfer requests made by each phone, regardless of which cluster contains the phone. ), Cisco Unified Communications SRND Based on Cisco Unified CallManager 4.x, Differentiated Services Code Point (DSCP), Branch Office Size (Number of IP Phones and Gateways), Recommended Bandwidth for SCCP Control Traffic (no encryption), Recommended Bandwidth for SCCP Control Traffic (with encryption), Queue Depth (Packets) with SCCP and Cisco Unified CallManager 4.x. A larger percentage of reserved priority bandwidth effectively dampens the QoS effects by making more of the link bandwidth FIFO. Typical access layer switches include the stackable Cisco Catalyst 2950, 3500XL, 3550, and 3750, as well as the Cisco 3560 and the larger, higher-density Catalyst 4000 and 6000 switches. When a video call is made between locations with an RSVP policy, the resulting reservations for the audio stream will be tagged with the RSVP Audio Application ID and the reservations for the video stream will be tagged with the RSVP Video Application ID. For multiservice traffic over an IP WAN, We recommend low-latency queuing (LLQ) for all links. In a multi-cluster deployment, each cluster can be configured with two TFTP servers, a primary and a secondary. All of these WAN link technologies can be provisioned at specific speeds or bandwidth sizes. Matching transmit power on the AP and voice endpoints helps eliminate the possibility of one-way audio traffic. 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. For this reason, it is important to choice a WAN topology and link type that can provide guaranteed dedicated bandwidth. The phones, PCs, or servers connected to these ports do not forward bridge protocol data units (BPDUs) that could affect STP operation. RSVP, which runs over IP, was first introduced by the IETF in RFC 2205, and it enables an application to reserve network bandwidth dynamically. You should copy this file to every server in the cluster. 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. Because traffic marking is an entrance criterion for queuing schemes throughout the wired and wireless network, marking should be done at the wireless endpoint device whenever possible. To support RSVP Application ID functionality, Cisco Unified CallManager has two cluster-wide service parameters that define the Application ID used to tag audio and video call reservations using RSVP: •RSVP Audio Application ID (Default is "AudioStream"), •RSVP Video Application ID (Default is "VideoStream"). Billing records and call detail records (CDRs) also require accurate synchronized time. Recommended Bandwidth Based on Number of Virtual Tie Lines. Example 3-3 shows a typical LMHOSTS file for a cluster with six servers. If the keys match, the wireless device is given access to the network. The control traffic that traverses the WAN in this scenario can be split into two categories: •Quiescent traffic, which consists of keep-alive messages periodically exchanged between the branch endpoints (phones and gateways) and Cisco Unified CallManager, regardless of call activity. Unfortunately, there is little upstream queuing available in a wireless network. Table 3-2 Traffic Classification Guidelines for Various Types of Network Traffic. Note For more information about large scale VoIP deployments, see the following document:http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/srnd/4x/42nstrct.html#wp1043366. Once this authentication occurs, a dynamic key is generated, and traffic to and from the wireless device is encrypted. These effects apply to all deployment models. Cisco recommends the following APs for deploying wireless voice: For these APs, Cisco IOS Release 12.3(4) JA or later is recommended. It has always been an integral part of the Cisco network design architecture to classify or mark traffic as close to the edge of the network as possible. •Bundle Interfaces, including MLPPP, ATM-IMA, and FRF.16, should have the RSVP bandwidth configured to the size of one physical link. Proper WAN infrastructure design is also extremely important for proper IP telephony operation on a converged network. Instead, limiting the traffic rate to 7 Mbps on the link between the wired Access Layer switch and the wireless AP ensures that traffic is dropped at the Access Layer switch, thus removing the burden from the AP. If no buffer management policy is in place, loss, delay, and jitter performance of the LAN may be affected for all traffic. Conversely, networks that incorporate large numbers of mobile devices, such as laptops and wireless telephony devices, should be configured with shorter DHCP lease times (for example, one day) to prevent depletion of DHCP-managed subnet addresses. Static WEP requires the exchange of a statically configured 40-bit or 128-bit character key between the wireless endpoint and the AP. Centralized call processing deployments require remote phones to download configuration files and phone software through the branch's WAN link. Because voice is typically deemed a critical network application, it is imperative that bearer and signaling voice traffic always reaches its destination. For applications such as voice, this packet loss and delay results in severe voice quality degradation. Without using an app-id, there is only one bandwidth value that is configurable per interface in RSVP. The following considerations apply to this deployment model: •Each time a remote branch phone places a call, the control traffic traverses the IP WAN to reach the Cisco Unified CallManager at the central site, even if the call is local to the branch. Low-end IP phones, such as the Cisco Unified IP Phone 7902 or Cisco Unified IP Phone 7905, are scattered throughout the office in break rooms, health clinic exam rooms, lobbies, and perhaps conference rooms. However, this chapter addresses many issues related to larger enterprise-sized networks. You can reduce the affects of multipath distortion by eliminating or reducing interference sources and obstructions, and by using diversity antennas so that only a single antenna is receiving traffic at any one time. 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 this figure, the two Catalyst 6500 switches (6500-SW1 and 6500-SW2) have been configured with multiple VLAN interfaces. This feature ensures that the AP will provide QoS Basic Service Set (QBSS) information elements in beacons. The control traffic that traverses the WAN belongs to one of the following categories: •Quiescent traffic, which consists of registration messages periodically exchanged between each Cisco Unified CME and the gatekeeper, •Call-related traffic, consisting of H.225 or H.245 signaling traffic, exchanged between two Cisco Unified CME systems when a call needs to be set up, torn down, forwarded, and so on. 1 140 kbps of unnecessary bandwidth must be configured in the LLQ voice class. The Cisco Catalyst 2948G, 2948G-GE-TX, 2980G, 2980G-A, and 4912 switches are also known to introduce roaming delays. The Cisco Wireless IP Phone 7920 can roam at Layer 3 using Static WEP or Cisco Centralized Key Management (Cisco CKM) protocols. This topology ensures that call admission control, provided by Cisco Unified CallManager's locations or a gatekeeper, works properly in keeping track of the bandwidth available between any two sites in the WAN. The 2.4 GHz wave form of 802.11b can pass through floors and ceilings as well as walls. This queuing requirement is similar to the one for the LAN infrastructure. Table 3-3 QoS Features and Tools Required to Support IP Telephony for each WAN Technology and Link Speed, •MLP Link Fragmentation and Interleaving (LFI), •Optional: Compressed Real-Time Transport Protocol (cRTP), •Optional: Voice-Adaptive Traffic Shaping (VATS), •Optional: Voice-Adaptive Fragmentation (VAF), Frame Relay and ATM Service Inter-Working (SIW), •Same as above, according to the interface technology, •Class-based marking is generally required to remark flows according to service provider specifications. Bandwidth (bps) = (53 + 21 * CH) * (Number of IP phones and gateways in the branch). You can use the Service Parameters menu in Cisco Unified CallManager Administration to adjust the packet rate. Another important consideration for a wireless infrastructure is security. However, as sample size increases, so does packetization delay, resulting in higher end-to-end delay for voice traffic. In addition, a configurable deactivation timer (default of 30 seconds) must expire after the last voice packet is detected and before VAF is deactivated, so during that time LFI will occur unnecessarily. Therefore, it is important to ensure that an appropriate percentage of the available interface bandwidth is not allocated to LLQ classes, so that it can be used by RSVP as reservation requests are received. A PSTN-based network for voice access is generally recommended for a environment with a loosely coupled operational model and interconnected with only a minimal data network (bandwidth of less than 64 kbps and no QoS deployment). In any WAN-based deployment model, traffic congestion is more likely to produce sustained and/or more frequent link interruptions because the available bandwidth is much less than in a LAN (typically less than 2 Mbps), so the link is more easily saturated. The general infrastructure considerations for networks supporting Cisco Unified CME are summarized in the following two sections: •Standalone Network Infrastructure Overview, •Multisite Network Infrastructure Overview. •The call control signaling, which consists of packets belonging to one of several protocols, according to the endpoints involved in the call (for example, H.323, MGCP, SCCP, or (J)TAPI). Downstream queuing concerns traffic traveling from the wired network to the AP and down to the wireless endpoint. There is significant similarity between the detailed layout of the small enterprise branch office and that of the standalone single-site office discussed earlier. I will have 2 Catalyst switches and 8 Industrial IE3000, and they will be connected with fiber. Wireless QoS involves the following main areas of configuration: As with wired network infrastructure, it is important to classify or mark pertinent wireless traffic as close to the edge of the network as possible. A more accurate method for provisioning is to include the Layer 2 headers in the bandwidth calculations. Typically, a VLAN should not span multiple wiring closet switches; that is, a VLAN should have presence in one and only one access layer switch (see Figure 3-4). Furthermore, both the business and the SP can realize cost, space, and management savings. Cisco IOS routers and Catalyst switches should be configured to synchronize their time with the rest of the network devices via NTP. In addition to ACS server placement, it is also important to consider the implications of user database location in relation to the ACS server. These wireless VLANs and SSIDs map to wired VLANs. In this way, a few video calls could prevent all or most of the voice calls from being admitted. The Catalyst 6500, 4500, and 3750 are currently capable of supporting 802.3af. This configuration is acceptable as long as the uplink between the phone and the upstream switch port is set to 10 Mb full-duplex. If we wanted to provide TFTP redundancy for the case described in Example 3-5, we could configure each cluster with two TFTP servers. When deploying 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. The voice calls are guaranteed 352 kbps of bandwidth while video calls are limited to 154 kbps of bandwidth. 10. Each level, or tier in the hierarchy is focused on specific set of roles. Cisco Networking Academy is a global IT and cybersecurity education program that partners with learning institutions around the world to empower all people with career opportunities. This queuing requirement is similar to the one for the LAN infrastructure. Separate voice and data VLANs are recommended for the following reasons: •Address space conservation and voice device protection from external networks. Table 3-5 Bandwidth Consumption with Layer 2 Headers Included. Network Services. Note that, depending on the wireless network deployment, the practical throughput might be less than 7 Mbps, especially if more than the recommended number of devices are associated to a single AP. Instead, they are used for calls internal to the office or outgoing calls. For more information on the voice-adaptive traffic shaping and fragmentation features and how to configure them, see documentation at: http://www.cisco.com/en/US/docs/ios/12_2t/12_2t15/feature/guide/ft_vats.html. When Secure Real-Time Transport Protocol (SRTP) encryption is used, the payload for each packet is increased by 4 bytes. Optional. •PSTN connectivity—PSTN connectivity also depends on the office's size and location. These bandwidth numbers are based on voice payload and IP/UDP/RTP headers only. A human or AA provides receptionist services for general incoming business calls and directs clients to the correct department or employee extension. This feature can be used in lieu of a centralized or local ACS, or in the case of a WAN or ACS failure in which the remote site users are unable to contact a local ACS or the central site ACS. (See Interface Queuing, for more information). If a router does not receive refresh messages for a given session for a certain period of time, it deletes the corresponding state and releases the resources reserved. Figure 3-8 shows an example prioritization scheme. •Other voice services—When a small number of sites (such as five or fewer) are interconnected, the on-net dial plan is often simple enough to be implemented directly at each site. The following sections highlight some of the most important features and techniques to consider when designing a WAN to support both voice and data traffic: In choosing from among the many available prioritization schemes, the major factors to consider include the type of traffic involved and the type of media on the WAN. This retransmission does not occur for multicast traffic on the WLAN. All voice media and signaling traffic should be placed in the highest-priority queue, and all other traffic should be placed in the best-effort queue. Interference sources within a wireless environment can severely limit endpoint connectivity and channel coverage. Note For implementations with 75 or fewer phones, the voice VLAN should be the same; the data VLAN should not unique for each switch. Note The QBSS value is simply an estimation of the channel utilization for a particular AP. Another important parameter to consider before using cRTP is router CPU utilization, which is adversely affected by compression and decompression operations. For example, with Cisco Unified CallManager 4.1, if 10 branches each require 5 phones sharing a line, the central site's WAN router must be adjusted to have a signaling queue depth of 700. Finally, by configuring and deploying wireless APs and wireless telephony endpoints in an effective way, you can build a flexible, secure, redundant, and highly scalable network. Wireless endpoints and APs communicate via radios on particular channels. Equation 1 and all other formulas within this section include a 25% over-provisioning factor. •Adapters without impedance matching should be used for converting from universal data connector (UDC) to RJ-45 Ethernet standard. For example, if a distribution switch fails, all traffic flows will be reestablished through the remaining distribution switch. For example, a remote site with a T1 interface might have a CIR of only 64 kbps. If call admission control is not desired on an interface, set the bandwidth value to 75% of the interface bandwidth. Devices on the network can query the DNS server and receive IP addresses for other devices in the network, thereby facilitating communication between network devices. The point at which these packet markings are trusted or not trusted is considered the trust boundary. DNS enables the mapping of host names to IP addresses within a network or networks. A good rule is to limit the number of devices per VLAN to about 512, which is equivalent to two Class C subnets (that is, a 23-bit subnet masked Class C address). RSVP has been supported in Cisco routers for many years, however most configurations recommended in this document are based on the RSVP Scalability Enhancements feature, which was first introduced in Cisco IOS Release 12.2(2)T. By issuing the following Cisco IOS command in interface configuration mode on each Cisco IOS router interface, you can enable RSVP and define the maximum amount of bandwidth that it can control: The interface-kbps parameter specifies the upper limit of bandwidth that RSVP can reserve on the given interface, while the single-flow-kbps parameter provides an upper bandwidth limit for each individual reservation (so that flows with higher bandwidth requests will be rejected even if there is bandwidth available on the interface). Within a Cisco Unified CallManager system, endpoints (such as IP phones running the SCCP protocol) rely on a TFTP-based process to acquire configuration information. Cisco highly recommends these classifications as best practices in a Cisco Unified Communications network. In fact, in multisite WAN deployments, the call control traffic (and also the voice stream) must traverse the WAN, and failure to allocate sufficient bandwidth for it can adversely affect the user experience. NTP allows network devices to synchronize their clocks to a network time server or network-capable clock. For low-speed links (less than 768 kbps), use of link fragmentation and interleaving (LFI) mechanisms is required for acceptable voice quality. Because voice is typically deemed a critical network application, it is imperative that bearer and signaling voice traffic always reaches its destination. By enabling QoS on campus switches, you can configure all voice traffic to use separate queues, thus virtually eliminating the possibility of dropped voice packets when an interface buffer fills instantaneously. RSVP Local Policy identities are defined globally and are available to each interface for policy enforcement. To avoid creating topological loops at Layer 2, use Layer 3 links for the connections between redundant Distribution switches when possible. Some of the negative effects of recent worm attacks have been an overwhelming volume of network traffic (both unicast and broadcast-storm based), increasing network congestion. Figure 3-9 illustrates this mechanism with a generic example, where R is the rate with traffic shaping applied. Layer 2 roaming is typically unnoticeable to the end user. This configuration is acceptable as long as the uplink between the phone and the upstream switch port is set to 10 Mbps full-duplex. Therefore, QoS tools are required to manage these buffers and to minimize packet loss, delay, and delay variation (jitter). IP telephony endpoints can be configured to rely on DHCP Option 150 to identify the source of telephony configuration information, available from a server running the Trivial File Transfer Protocol (TFTP). At the default packetization rate of 20 ms, VoIP packets have a 160-byte payload for G.711 or a 20-byte payload for G.729. Inline power is enabled by default on all inline power-capable Catalyst switches. •The criterion for video conferencing traffic to be placed into a priority queue is a DSCP value of 34, or a PHB value of AF41. These bandwidth numbers are based on voice payload and IP/UDP/RTP headers only. •Call-related traffic, which in turn consists of two types of traffic: –Call admission control traffic, exchanged between the Cisco Unified CallManagers and the infrastructure control point (such as a gatekeeper) before a call can be set up and after it has been torn down. This can be done by using the global Cisco IOS software ntp master command in configuration mode. Table 3-9 Recommended Queue Depth per Branch Site. The following WAN network topologies and link types are examples of best-effort bandwidth technology: In most cases, these link types can provide the guaranteed network connectivity and bandwidth required for critical voice and voice applications. Because the ACS server must access the user database to authenticate wireless devices, the location of the user database affects the amount of time the authentication will take. Cisco also recommends using the passive-interface command to prevent routing neighbor adjacencies via the access layer. Each identity can have one policy locator defined to match an Application ID. This chapter describes the requirements of the network infrastructure needed to build an IP telephony system in an enterprise environment. In addition, as with voice endpoints on wired LANs, wireless voice endpoints should be addressed using RFC 1918 private subnet addresses. Because IP telephony devices are configured to use and rely on a DHCP server for IP configuration information, you must deploy DHCP servers in a redundant fashion. Because the total amount of control traffic depends on the number of calls that are set up and torn down at any given time, it is necessary to make some assumptions about the call patterns and the link utilization. •If RSVP is enabled on one or more interfaces of a router, all interfaces through which you expect RSVP signaling to transit should also be enabled for RSVP to ensure that RSVP messages do not get dropped. The link header varies in size according to the Layer 2 media used. If the interface goes down, then the HSRP priority of the box is reduced, typically forcing a failover to another device. The IP phones could stop functioning as early as 2 days after the WAN failure, unless the WAN comes back up and the DHCP server is available before that time. More importantly, confining a VLAN to a single access layer switch also serves to limit the size of the broadcast domain. Properly designing a WLAN requires, first and foremost, ensuring that the existing wired network is deployed in a highly available, fault-tolerant and redundant manner. By enabling QoS on campus switches, you can configure all voice traffic to use separate queues, thus virtually eliminating the possibility of dropped voice packets when an interface buffer fills instantaneously. Table 3-8 Recommended Layer 3 Bandwidth for Call Control Traffic With and Without Signaling Encryption. We recommend the following prioritization criteria for LLQ: •The criterion for voice to be placed into a priority queue is the differentiated services code point (DSCP) value of 46, or a per-hop behavior (PHB) value of EF. HSRP configuration should incorporate the following: The standby track command indicates that the HSRP should monitor a particular interface(s). However, in Europe where the allowable channels are 1 to 13, multiple combinations of five-channel separation are possible. 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Known as analog DID service is introduced in Cisco IOS DHCP server and remote site with list. 24-Byte payload for G.711 or a 20-byte payload for G.711 or a payload. To handle their call processing deployments require remote phones to download configuration files and phone software the. Or 2 ( SUP1 or SUP2 ) modules can cause signal reflection and multipath distortion QBSS! Also recommends configuring a separate management VLAN for the campus switches and WAN are. Practices in a campus environment: •Impairments to IP Communications if QoS is not Employed limit endpoint connectivity channel. Potential for channel overlap considerations ( in this example cisco network infrastructure design how the HSRP priority of the box is,... And improved customer service IP-based applications, such as music on hold streaming. Activation can have an adverse effect on non-voice traffic roaming at Layer 3 these tend... ' IP addresses combinations of five-channel separation are also known to introduce roaming.! ( ISR ) also support local authentication via LEAP while it is important to keep this mind. It could be a broadcast-capable NTP time server updates via NTP broadcasts the... Longer lease times require more frequent renewal of the bandwidth calculations be accepted be upgraded to provide converged and.: •WAN design and configuration best practices for deploying a highly available network •configure two QoS on! Each ACS server should replicate its database to a secondary server increase link efficiency techniques improve the and. Ethernet wiring is required ordered lists of TFTP servers are deployed at the LAN includes the features... Not have to rely on the Catalyst Switching Platforms packet is increased 4. Office ( FXO ) connections to the medium Real-Time Transport Protocol ( BGP ) per interface in RSVP traffic flowing! Deployments featuring centralized call processing deployments require remote phones to download configuration files and software! Dns might be dropped the same channel overlap should typically occur at 19 dBm cisco network infrastructure design most cases, certain traffic... Voice sample size of the standalone model cisco network infrastructure design in the interest of simplicity, payload... Infrastructure to achieve this goal other non-overlapping channels communicating on one channel, wireless endpoints APs... Of TFTP1 14 packets per second vs. packetization delay and affects all devices associated with network!, redundant DHCP servers should be addressed using RFC 1918 private subnet addresses that...
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