Optimizing Converged Cisco Networks (ONT)

Transcription

1 Optimizing Converged Cisco Networks (ONT) Module 2: Cisco VoIP Implementations (Deploy)

2 Calculating Bandwidth Requirements for VoIP

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.

4 Factors Influencing Encapsulation Overhead and Bandwidth 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)

5 Bandwidth Implications of Codecs Codec bandwidth is for voice information only. No packetization overhead is included. Codec Bandwidth G.711 (PCM) 64 kbps G.726 r32 (ADPCM) 32 kbps G.726 r24 (ADPCM) 24 kbps G.726 r16 (ADPCM) 16 kbps G.728 (LDCELP) 16 kbps G.729 (CS-ACELP) 8 kbps

6 How the Packetization Period Impacts VoIP Packet Size and Rate High packetization period results in: Larger IP packet size (adding to the payload) and more delay Lower packet rate (reducing the IP overhead)

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. Characteristics of IPsec and tunneling protocols are: The original frame or packet is encapsulated into another protocol. The added headers result in larger packets and higher bandwidth requirements. The extra bandwidth can be extremely critical for voice packets because of the transmission of small packets at a high rate.

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

11 Example: VoIP over IPsec VPN G.729 codec (8 kbps) 20-ms packetization period (20B) No crtp (40B) IPsec ESP with 3DES and SHA-1, tunnel mode ( )

12 Total Bandwidth Required for a VoIP Call Total bandwidth of a VoIP call, as seen on the link, is important for: Designing the capacity of the physical link Deploying Call Admission Control (CAC) Deploying QoS

13 Total Bandwidth Calculation Procedure Gather required packetization information: Packetization period (default is 20 ms) or size Codec bandwidth Gather required information about the link: crtp enabled Type of data-link protocol IPsec or any tunneling protocols used Calculate the packetization size or period. Sum up packetization size and all headers and trailers. Calculate the packet rate. Calculate the total bandwidth.

14 Bandwidth Calculation Example

15 Quick Bandwidth Calculation 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

16 VAD (Voice Activity Detection) Detects silence (speech pauses) Suppresses transmission of silence patterns Depends on multiple factors: Type of audio (for example, speech or Music on Hold (MoH)) Level of background noise Other factors (for example, language, character of speaker, or type of call) Can save up to 35 percent of bandwidth (statistical distribution of at least 24 calls)

17 VAD Bandwidth-Reduction Examples Data-Link Overhead Ethernet 18 bytes Frame Relay 6 bytes Frame Relay 6 bytes MLPP 6 bytes IP overhead no crtp crtp no crtp crtp 40 bytes 4 bytes 40 bytes 2 bytes Codec G.711 G.711 G.729 G kbps 64 kbps 8 kbps 8 kbps Packetization 20 ms 30 ms 20 ms 40 ms 160 bytes 240 bytes 20 bytes 40 bytes Bandwidth without VAD Bandwidth with VAD (35% reduction) 87.2 kbps kbps 26.4 kbps 9.6 kbps kbps kbps kbps 6.24 kbps

18 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?

19 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.

20 Resources Voice Over IP - Per Call Bandwidth Consumption h_note09186a ae2.shtml#topic1 Voice Codec Bandwidth Calculator

21 Implementing VoIP in an Enterprise Network

22 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 Cisco Unified CallManager. Identify common enterprise IP telephony deployment models. Identify basic Cisco IOS VoIP configuration commands.

23 Enterprise Voice Implementations Components of enterprise voice networks: Gateways (PSTN connectivity) and gatekeepers (scalability) Cisco Unified CallManager (PBX-like features) and IP phones

24 Deploying Call Admission Control (CAC) CAC artificially limits the number of concurrent voice calls. CAC prevents oversubscription of WAN resources caused by too much voice traffic. CAC is needed because QoS cannot solve the problem of voice call oversubscription: 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.

25 Example: CAC Deployment IP network (WAN) is only designed for two concurrent voice calls. If CAC is not deployed, a third call can be set up, causing poor quality for all calls. When CAC is deployed, the third call is blocked.

26 Voice Gateway Functions on a Cisco Router 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 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

27 Cisco Unified CallManager Functions Call processing Dial plan administration (MGCP: Media Gateway Control Protocol ) Signaling and device control Phone feature administration Directory and XML services Programming interface to external applications Cisco IP Communicator

28 Example: Signaling and Call Processing RTP RTP

29 Enterprise IP Telephony Deployment Models Deployment Model Single site Characteristics Cisco Unified CallManager cluster at the single site Local IP phones only Multisite with centralized call processing Multisite with distributed call processing 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 Clustering over WAN 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

30 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.

31 Multisite with Centralized Call Processing Cisco Unified CallManager 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. Call Admission Control (CAC) is used to limit the number of VoIP calls, and Automated alternate routing (AAR) is used if WAN bandwidth is exceeded. Cisco Survivable Remote Site Telephony (SRST) is located at the remote branch. Note: Cisco Unified CallManager cluster can be connected to various places depending on the topology.

32 Multisite with Distributed Call Processing Cisco Unified CallManager 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. Note: Cisco Unified CallManager cluster can be connected to various places, depending on the topology.

33 Clustering over WAN Cisco Unified CallManager 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). 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. Note: Cisco Unified CallManager cluster can be connected to various places, depending on the topology.

34 Basic Cisco IOS VoIP Voice Commands

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_number The destination-pattern command, entered in dial peer subconfiguration mode, defines the telephone number that applies to the dial peer.

36 Voice-Specific Commands (Cont.) router(config-dial-peer)# port port-number The port command, entered in POTS dial peer subconfiguration mode, 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 The session target command, entered in VoIP dial peer subconfiguration mode, defines the IP address of the target VoIP device that applies to the dial peer.

37 Self Check 1. What is CAC? 2. What can happen is CAC is not used? 3. What command is used to define the telephone number that applies to the dial peer? 4. List 4 deployment options when using the Cisco Unified CallManager.

38 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 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 Q and A

40 Resources Video: The ABCs of VoIP (16 min.) Voice and Unified Communications VoIP Call Admission Control l/cac.htm

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