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1 Optimizing Converged Cisco Networks (ONT) reserved.
2 Lesson 2.4: Calculating Bandwidth Requirements for VoIP reserved.
3 Objectives Describe factors influencing encapsulation overhead and bandwidth requirements for VoIP. Explain how the packetization period impacts VoIP packet size and rate. Explain how link encapsulation effects data-link overhead on a per link basis. Explain the bandwidth impact of adding a tunneling protocol header to voice packets. Use the bandwidth calculation process to calculate bandwidth needs for various VoIP call types. Describe how VAD is used in VoIP implementations. 3
4 Factors Influencing Encapsulation Overhead and dbandwidth Factor Packet rate Packetization size (payload size) IP overhead (including UDP and RTP) Data-link overhead Tunneling overhead (if used) Description Derived from packetization period (the period over which encoded voice bits are collected for encapsulation) Depends on packetization period Depends on codec bandwidth (bits per sample) Depends on the use of crtp Depends on protocol (different per link) Depends on protocol (IPsec, GRE, or MPLS) 4
5 Bandwidth Implications of Codecs Codec bandwidth is for voice information only. No packetization overhead is included. Codec Bandwidth G kbps G.726 r32 32 kbps G.726 r24 24 kbps G.726 r16 16 kbps G kbps G kbps 5
6 How the Packetization Period Impacts VoIP Packet Size and Rate High packetization period results in: Larger IP packet size (adding to the payload) Lower packet rate (reducing the IP overhead) Default packetization period on most Cisco VoIP devices is 20 ms. This default is the optimal value for most scenarios. 6
7 VoIP Packet Size and Packet Rate Examples Codec and Packetization Period G ms G ms G ms G ms Codec bandwidth (kbps) Packetization size (bytes) IP overhead (bytes) VoIP packet size (bytes) Packet rate (pps)
8 Data-Link Overhead Is Different per Link Data-Link Protocol Ethernet Frame Relay MLP Ethernet Trunk (802.1Q) Overhead [bytes]
9 Security and Tunneling Overhead IP packets can be secured by IPsec. Additionally, IP packets or data-link frames can be tunneled over a variety of protocols. GRE, transport L3 packets or L2 frames over IP packets Layer 2 Forwarding (L2F) and Layer 2 Tunneling Protocol (L2TP) PPPoE, which allows PPP to be used over Ethernet 802.1Q tunneling, transports 802.1Q frames inside another VLAN Characteristics of IPsec and tunneling protocols are: The original i lframe or packet ktis encapsulated dit into another protocol. The added headers result in larger packets and higher bandwidth. The extra bandwidth can be extremely critical for voice packets because of the transmission of small packets at a high rate. 9
10 Extra Headers in Security and Tunneling Protocols Protocol Header Size (bytes) IPsec transport mode IPsec tunnel mode L2TP/GRE 24 MPLS 4 PPPoE 8 IPsec and tunneling protocols add headers of different sizes. Depends on the use of the available headers (AH and ESP), the encryption or authentication algorithms that are used in these headers, and the IPsec mode (transport tor tunnel mode). ) 10
11 Example: VoIP over IPsec VPN G.729 codec (8 kbps) 20-ms packetization i period d(20b bytes) No crtp IPsec ESP with 3DES and SHA-1, tunnel mode 4 bytes of padding to reach a payload size of 64 bytes, and 30 bytes for the ESP header 11
12 Total Bandwidth Required for a VoIP Call Total bandwidth of a VoIP call, as seen on the link, is important for: Designingg gthe capacity of the physical link Deploying Call Admission Control (CAC) Limits the number of concurrent voice calls to avoid oversubscription of the link. Deploying QoS 12
13 Total Bandwidth Calculation Procedure To calculate the total bandwidth of a VoIP call, perform these steps: 1. Gather required packetization information: Packetization period (default is 20 ms) or size Codec bandwidth h( (approximately 8 64 kbps) 2. Gather required information about the link: crtp enabled Type of data-link protocol IPsec or any tunneling protocols used 3. Calculate the packetization size or period. 4. Sum up packetization size and all headers and trailers. 5. Calculate the packet rate (how many packets will be sent per second). 6. Calculate the total bandwidth. 13
14 Bandwidth Calculation Example Bandwidth [kbps] = (Total packet size [bytes per packet] * 8 / 1000) * packet rate [pps] 14
15 Quick Bandwidth Calculation l Total packet size Total bandwidth requirement = Payload size Nominal bandwidth requirement Total packet size = All headers + payload Parameter Layer 2 header IP + UDP + RTP headers Value 6 to 18 bytes 40 bytes Payload size (20-ms sample interval) 20 bytes for G.729, 160 bytes for G.711 Nominal bandwidth 8 kbps for G.729, 64 kbps for G.711 Example: G.729 with Frame Relay: Total bandwidth requirement = ( bytes) * 8 kbps = 26.4 kbps 20 bytes 15
16 Voice activity detection (VAD) Characteristics Voice activity detection (VAD) can take advantage of the fact that one-third of the average voice call consists of silence. Detects silence (speech pauses) Suppresses transmission of silence patterns The amount of bandwidth saved depends on multiple factors: Type of audio (for example, speech or MoH) Level of background noise (if too high, VAD cannot detect silence) Other factors (for example, language, character of speaker, or type of call) Can save up to 35 percent of bandwidth under certain statistical distribution of call types, which is usually achieved only if a link carries at least 24 calls. If fyou are calculating l bandwidth hfor fewer calls, you should not take VAD into account. 16
17 VAD Bandwidth-Reduction Examples Data-Link Overhead Ethernet Frame Relay Frame Relay MLPP 18 bytes 6 bytes 6 bytes 6 bytes IP overhead no crtp crtp no crtp crtp 40 bytes 4 bytes 40 bytes 2 bytes Codec G kbps G kbps G kbps G kbps Packetization 20 ms 160 bytes 30 ms 240 bytes 20 ms 20 bytes 40 ms 40 bytes Bandwidth without t 87.2 kbps kbps 26.4 kbps kbps VAD Bandwidth with VAD (35% reduction) kbps kbps kbps 6.24 kbps
18 Lesson 2.5: Implementing VoIP in an Enterprise Network reserved.
19 Objectives List the common components of an enterprise voice implementation. Describe Call Admission Control and how it differs from QoS. Describe the functions of the ecisco Unified CallManager. a age. Identify common enterprise IP telephony deployment models. Identify basic Cisco IOS VoIP configuration commands. 19
20 Enterprise Voice Implementations Components of enterprise voice networks: Gateways (interconnect traditional telephony systems) Gatekeepers (scalability of dial plans and for bandwidth management when using the H.323 protocol) Cisco Unified CallManager (PBX-like features to IP phones) and IP phones 20
21 Deploying CAC CAC artificially limits the number of concurrent voice calls. Prevents oversubscription of WAN resources caused by too much voice traffic. Is needed because QoS cannot solve the problem of voice call oversubscription: i QoS gives priority only to certain packet types (RTP versus data). QoS cannot block the setup of too many voice calls. Too much voice traffic results in delayed voice packets. 21
22 Example: CAC Deployment IP network (WAN) is only designed for two concurrent voice calls. If CAC is not deployed, d a third call can be set up, causing poor quality for all calls. When CAC is deployed, the third call is blocked. With the CAC configuration, no voice quality problems should be experienced. 22
23 Voice Gateway Functions on a Cisco Router Cisco routers, especially the ISR, are voice capable. These routers can be equipped with traditional telephony interfaces to act as gateways for analog and digital devices. The gateway functions are: dual tone multifrequency (DTMF) Connects traditional telephony devices to VoIP Converts analog signals to digital format Encapsulates voice into IP packets Performs voice compression Provides DSP resources for conferencing and transcoding Cisco Survivable Remote Site Telephony (SRST) Supports fallback scenarios for IP phones (Cisco SRST) Acts as a call agent for IP phones (Cisco Unified CallManager Express) Provides DTMF relay and fax and modem support 23
24 Cisco Unified CallManager Functions Cisco Unified CallManager is the IP-based PBX in an IP telephony solution. Acts t as a call agent for IP phones and MGCP gateways and can also interact with H.323 or SIP devices. There are six main functions performed by Cisco Unified CallManager: Call processing Dial plan administration Signaling and device control Phone feature administration Directory and XML services Programming interface to external applications: Cisco IP Communicator, Cisco IP Interactive Voice Response (IVR), Cisco Personal Assistant, and Cisco IP Communicator Cisco Unified CallManager Attendant Console. 24
25 Example: Signaling and Call Processing 1. User dials the telephone number, the IP phone sends signaling messages to the Cisco Unified CallManager cluster. 2. The CUCM server processes the call by looking up the called number in the CUCM s call routing table. 25
26 Example: Signaling and Call Processing 3. The CUCM server determines the IP address of the destination telephone, and sends a signaling message to the destination i telephone. The destination telephone starts ringing, and the user who is being called can accept the call. 4. After the call is accepted, the telephones start sending and receiving RTP packets that carry audio signals. 26
27 Enterprise IP Telephony Deployment Models Deployment Model Single site Multisite with centralized call processing Multisite with distributed call processing Clustering over WAN Characteristics Cisco Unified CallManager cluster at the single site Local IP phones only Cisco Unified CallManager cluster only at a single site Local and remote IP phones Cisco Unified CallManager clusters at multiple sites Local IP phones only Single Cisco Unified CallManager cluster distributed over multiple sites Usually local IP phones only Requirement: Round-trip delay between any pair of servers not to exceed 40 ms 27
28 Single Site Cisco Unified CallManager servers, applications, and DSP resources are located at the same physical location. IP WAN is not used for voice. PSTN is used for all external calls. Note: Cisco Unified CallManager cluster can be connected to various places depending on the topology. 28
29 Multisite it with Centralized Call Processing CUCM servers and applications are located at the central site while DSP resources are distributed. IP WAN carries data and voice (signaling for all calls, media only for intersite calls). PSTN access is provided at all sites. 29
30 Multisite it with Centralized Call Processing CAC is used to limit the number of VoIP calls, and Automated Alternate Routing (AAR) is used if WAN bandwidth is exceeded to reroute the calls through the PSTN Cisco SRST is located at the remote branch. Note: CUCM cluster can be connected to various places depending on the topology. 30
31 Multisite it with Distributed ib t Call Processing Note: CUCM cluster can be connected to various places, depending on the topology. CUCM servers, applications, and DSP resources are located at each site. IP WAN carries data and voice for intersite calls only (signaling and media). PSTN access is provided at all sites; rerouting to PSTN is configured if IP WAN is down. CAC is used to limit the number of VoIP calls, and AAR is used if WAN bandwidth is exceeded. 31
32 Clustering over WAN Note: CUCM cluster can be connected to various places, depending on the topology. CUCM servers of a single cluster are distributed among multiple sites while applications and DSP resources are located at each site. Intracluster communication (such as database synchronization) is performed over the WAN. IP WAN carries data and voice for intersite calls only (signaling and media). 32
33 Clustering over WAN Note: CUCM cluster can be connected to various places, depending on the topology. PSTN access is provided at all sites; rerouting to PSTN is performed if IP WAN is down. CAC is used to limit the number of VoIP calls; AAR is used if WAN bandwidth is exceeded. 33
34 Basic Cisco IOS VoIP Voice Commands 34
35 Voice-Specific Commands router(config)# dial-peer voice tag type Use the dial-peer voice command to enter the dial peer subconfiguration mode. router(config-dial-peer)# destination-pattern telephone_numbernumber Defines the telephone number that applies to the dial peer. 35
36 Voice-Specific Commands (Cont.) router(config-dial-peer)# port port-number Defines the port number that applies to the dial peer. Calls that are routed using this dial peer are sent to the specified port. router(config-dial-peer)# session target ipv4 :ip-address Defines the IP address of the target VoIP device that applies to the dial peer. 36
37 Self Check 1. Describe the relationship between packetization period and packet size and packet rate. 2. How does the data-link protocol used effect bandwidth considerations? 3. What is the default packetization period on Cisco devices? 4. What is VAD? 5. How much bandwidth can be saved, on average, using VAD? 6. What is CAC? 7. What can happen is CAC is not used? 8. What command is used to define the telephone number that applies to the dial peer? 9. List 4 deployment options when using the Cisco Unified CallManager. 37
38 Summary VoIP packet size and rate are determined by the packetization period. Data-link overhead must be considered with calculating bandwidth requirements. Different links have different overhead requirements. Adding a tunneling protocol header effects the bandwidth requirements for voice packets. This additional overhead must be considered when calculating bandwidth requirements. Voice Activity Detection (VAD) is a process used to detect silence in order to save bandwidth. VAD can save 34% on average. 38
39 Summary Enterprise voice implementations use components such as gateways, gatekeepers, Cisco Unified CallManager, and IP phones. Call Admission Control (CAC) extends the functionality of QoS to ensure that t an additional call is not allowed unless bandwidth is available to support it. Enterprise IP Telephony deployment models include single site, multisite with centralized call processing, multisite with distributed call processing, and clustering over the WAN. 39
40 Resources VoiceOverIP - Per Call Bandwidth Consumption ae2.shtml#topic1 Voice Codec Bandwidth Calculator Video: The ABCs of VoIP (16 min.) Voice and Unified Communications VoIP Call Admission Control p 40
41 41 41
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