Cisco Systems 3600の取扱説明書

デバイスCisco Systems 3600の取扱説明書

デバイス: Cisco Systems 3600
カテゴリ: IPフォン
メーカー: Cisco Systems
サイズ: 0.31 MB
追加した日付: 9/6/2014
ページ数: 36
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内容要旨
ページ1に含まれる内容の要旨

Configuring Voice over IP for the
Cisco 3600 Series
This chapter shows you how to configure Voice over IP (VoIP) on the Cisco 3600 series. For a
description of the commands used to configure Voice over IP, refer to the “Voice-Related
Commands” chapter in the Voice, Video, and Home Applications Command Reference.
VoIP enables a Cisco 3600 series router to carry voice traffic (for example, telephone calls and faxes)
over an IP network. Voice over IP is primarily a software feature; however, to

ページ2に含まれる内容の要旨

List of Terms 5 The session application then runs the H.323 session protocol to establish a transmission and a reception channel for each direction over the IP network. If the call is being handled by a PBX, the PBX forwards the call to the destination telephone. If RSVP has been configured, the RSVP reservations are put into effect to achieve the desired quality of service over the IP network. 6 The CODECs are enabled for both ends of the connection and the conversation proceeds using RTP/U

ページ3に含まれる内容の要旨

Prerequisite Tasks PBX—Private Branch Exchange. Privately-owned central switching office. PLAR—Private Line Auto Ringdown. This type of service results in a call attempt to some particular remote endpoint when the local extension is taken off-key. POTS—Plain Old Telephone Service. Basic telephone service supplying standard single line telephones, telephone lines, and access to the public switched telephone network. POTS dial peer—Dial peer connected via a traditional telephony network. POTS pe

ページ4に含まれる内容の要旨

Voice over IP Configuration Task List Voice over IP Configuration Task List To configure Voice over IP on the Cisco 3600 series, you need to complete the following tasks: 1 Configure IP Networks for Real-Time Voice Traffic Configure your IP network to support real-time voice traffic. Fine-tuning your network to adequately support VoIP involves a series of protocols and features geared toward Quality of Service (QoS). To configure your IP network for real-time voice traffic, you need to take in

ページ5に含まれる内容の要旨

Configure IP Networks for Real-Time Voice Traffic (b) VoIP—Dial peer describing the characteristics of a packet network connection; in the case of Voice over IP, this is an IP network. VoIP peers point to specific VoIP devices. To minimally configure a VoIP peer, you need to configure the following two characteristics: associated destination telephone number and a destination IP address. Use the destination-pattern command to define the destination telephone number associated with a VoIP pe

ページ6に含まれる内容の要旨

Configure IP Networks for Real-Time Voice Traffic The important thing to remember is that QoS must be configured throughout your network—not just on the Cisco 3600 series devices running VoIP—to improve voice network performance. Not all QoS techniques are appropriate for all network routers. Edge routers and backbone routers in your network do not necessarily perform the same operations; the QoS tasks they perform might differ as well. To configure your IP network for real-time voice traffi

ページ7に含まれる内容の要旨

Configure Multilink PPP with Interleaving Configure Multilink PPP with Interleaving Multi-class Multilink PPP Interleaving allows large packets to be multilink-encapsulated and fragmented into smaller packets to satisfy the delay requirements of real-time voice traffic; small real-time packets, which are not multilink-encapsulated, are transmitted between fragments of the large packets. The interleaving feature also provides a special transmit queue for the smaller, delay-sensitive packets,

ページ8に含まれる内容の要旨

Configure IP Networks for Real-Time Voice Traffic For more information about Multilink PPP, refer to the “Configuring Media-Independent PPP and Multilink PPP” chapter in the Dial Solutions Configuration Guide. Multilink PPP Configuration Example The following example defines a virtual interface template that enables Multilink PPP with interleaving and a maximum real-time traffic delay of 20 milliseconds, and then applies that virtual template to the Multilink PPP bundle: interface virtual-tem

ページ9に含まれる内容の要旨

Configure RTP Header Compression You should configure RTP header compression if the following conditions exist in your network: � Slow links � Need to save bandwidth Note RTP header compression should not be used on links greater than 2 Mbps. Perform the following tasks to configure RTP header compression for Voice over IP. The first task is required; the second task is optional. � Enable RTP Header Compression on a Serial Interface � Change the Number of Header Compression Connections Enable R

ページ10に含まれる内容の要旨

Configure Frame Relay for Voice over IP Configure Custom Queuing Some QoS features, such as IP RTP reserve and custom queuing, are based on the transport protocol and the associated port number. Real-time voice traffic is carried on UDP ports ranging from 16384 to 16624. This number is derived from the following formula: 16384 = 4(number of voice ports in the Cisco 3600 series router) Custom Queuing and other methods for identifying high priority streams should be configured for these port ra

ページ11に含まれる内容の要旨

Frame Relay for Voice over IP Configuration Example Note Lowering the MTU size affects data throughput speed. � CIR equal to line rate—Make sure that the data rate does not exceed the CIR. This is accomplished through generic traffic shaping. — Use IP Precedence to prioritize voice traffic. — Use compressed RTP to minimize voice packet header size. � Traffic shaping—Use adaptive traffic shaping to throttle back the output rate based on the BECN. If the feedback from the switch is ignored, pac

ページ12に含まれる内容の要旨

Configure Number Expansion � Fair-queuing is enabled. � IP RTP header compression is enabled. The subinterface has been configured as follows: � MTU size is inherited from the main interface. � IP address for the subinterface is specified. � Bandwidth is set to 64 kbps. � Generic traffic shaping is enabled with 32 kbps CIR where Bc=4000 bits and Be=4000 bits. � Frame Relay DLCI number is specified. � IP RTP header compression is enabled. Note When traffic bursts over the CIR, output rate is held

ページ13に含まれる内容の要旨

Configure Number Expansion Figure 5 Sample Voice over IP Network 729 555-1001 729 555-1002 408 115-1001 729 555-1000 729 555-1003 T1 ISDN PRI Cisco 3600 Router 1 Voice port 408 116-1002 Voice port 0:D 0:D IP WAN cloud WAN 10.1.1.1 10.1.1.2 1:D T1 ISDN PRI Cisco 3600 Router 2 408 117-1003 Table 5 shows the number expansion table for this scenario. Table 5 Sample Number Expansion Table Extension Destination Pattern Num-Exp Command Entry 5.... 40811..... num-exp 5.... 408115.... 6.... 40811.....

ページ14に含まれる内容の要旨

Configure Dial Peers Configure Dial Peers The key point to understanding how Voice over IP functions is to understand dial peers. Each dial peer defines the characteristics associated with a call leg, as shown in Figure 6 and Figure 7. A call leg is a discrete segment of a call connection that lies between two points in the connection. All the call legs for a particular connection have the same connection ID. There are two different kinds of dial peers: � POTS—Dial peer describing the charac

ページ15に含まれる内容の要旨

Inbound versus Outbound Dial Peers POTS peers associate a telephone number with a particular voice port so that incoming calls for that telephone number can be received and outgoing calls can be placed. VoIP peers point to specific devices (by associating destination telephone numbers with a specific IP address) so that incoming calls can be received and outgoing calls can be placed. Both POTS and VoIP peers are needed to establish Voice over IP connections. Establishing communication using

ページ16に含まれる内容の要旨

Configure Dial Peers Figure 9 Outgoing Calls from the Perspective of POTS Dial Peer 2 Destination Source IP cloud Dial peer 2 Dial peer 3 Dial peer 4 Dial peer 1 Voice port Voice port 1/0/0 10.1.2.2 1/0/0 10.1.1.2 (408) 555-4000 (310) 555-1000 POTS call leg VoIP call leg To complete the end-to-end call between dial peer 1 and dial peer 4 as illustrated in Figure 9, enter the following commands on router 10.1.1.2: dial-peer voice 4 pots destination-pattern 1310555.... port 1/0/0 dial-peer voice

ページ17に含まれる内容の要旨

Configure POTS Peers Configure POTS Peers Once again, POTS peers enable incoming calls to be received by a particular telephony device. To configure a POTS peer, you need to uniquely identify the peer (by assigning it a unique tag number), define its telephone number(s), and associate it with a voice port through which calls will be established. Under most circumstances, the default values for the remaining dial-peer configuration commands will be sufficient to establish connections. To ente

ページ18に含まれる内容の要旨

Configure Dial Peers Figure 10 Incoming and Outgoing POTS Call Legs Cisco 3600 Cisco 3600 PBX PBX IP cloud Incoming Outgoing call leg call leg Unless otherwise configured, when a call arrives on the access server, the server presents a dial tone to the caller and collects digits until it can identify the destination dial peer. After the dial peer has been identified, the call is forwarded through the next call leg to the destination. There are cases where it might be necessary for the server

ページ19に含まれる内容の要旨

Configure VoIP Peers To configure DID for a particular POTS dial peer, use the following commands beginning in global configuration mode: Step Command Purpose 1 dial-peer voice number pots Enter the dial-peer configuration mode to configure a POTS peer. 2 direct-inward-dial Specify direct inward dial for this POTS peer. Note Direct inward dial is configured for the calling POTS dial peer. For additional POTS dial-peer configuration options, refer to the “Voice-Related Commands” section of th

ページ20に含まれる内容の要旨

Optimize Dial Peer and Network Interface Configurations Validation Tips You can check the validity of your dial-peer configuration by performing the following tasks: � If you have relatively few dial peers configured, you can use the show dial-peer voice command to verify that the data configured is correct. Use this command to display a specific dial peer or to display all configured dial peers. � Use the show dialplan number command to show the dial peer to which a particular number (destin


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