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© 2006 Cisco Systems, Inc. All rights reserved. QOS Lecture 3 - Encapsulating Voice Packets for Transport.

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Presentation on theme: "© 2006 Cisco Systems, Inc. All rights reserved. QOS Lecture 3 - Encapsulating Voice Packets for Transport."— Presentation transcript:

1 © 2006 Cisco Systems, Inc. All rights reserved. QOS Lecture 3 - Encapsulating Voice Packets for Transport

2 © 2006 Cisco Systems, Inc. All rights reserved. Voice Transport in Circuit-Switched Networks  Analog phones connect to CO switches.  CO switches convert between analog and digital.  After call is set up, PSTN provides: End-to-end dedicated circuit for this call (DS-0) Synchronous transmission with fixed bandwidth and very low, constant delay

3 © 2006 Cisco Systems, Inc. All rights reserved. Voice Transport in VoIP Networks  Analog phones connect to voice gateways.  Voice gateways convert between analog and digital.  After call is set up, IP network provides: Packet-by-packet delivery through the network Shared bandwidth, higher and variable delays

4 © 2006 Cisco Systems, Inc. All rights reserved. Jitter  Voice packets enter the network at a constant rate.  Voice packets may arrive at the destination at a different rate or in the wrong order.  Jitter occurs when packets arrive at varying rates.  Since voice is dependent on timing and order, a process must exist so that delays and queuing issues can be fixed at the receiving end.  The receiving router must: Ensure steady delivery (delay) Ensure that the packets are in the right order

5 © 2006 Cisco Systems, Inc. All rights reserved. VoIP Protocol Issues  IP does not guarantee reliability, flow control, error detection or error correction.  IP can use the help of transport layer protocols TCP or UDP.  TCP offers reliability, but voice doesn’t need it…do not retransmit lost voice packets.  TCP overhead for reliability consumes bandwidth.  UDP does not offer reliability. But it also doesn’t offer sequencing…voice packets need to be in the right order.  RTP, which is built on UDP, offers all of the functionality required by voice packets.

6 © 2006 Cisco Systems, Inc. All rights reserved. Protocols Used for VoIP Feature Voice Needs TCPUDPRTP ReliabilityNoYes No ReorderingYes No Yes Time- stamping YesNo Yes Overhead As little as possible Contains unnecessary information Low MultiplexingYes No

7 © 2006 Cisco Systems, Inc. All rights reserved. Voice Encapsulation  Digitized voice is encapsulated into RTP, UDP, and IP.  By default, 20 ms of voice is packetized into a single IP packet.

8 © 2006 Cisco Systems, Inc. All rights reserved. Voice Encapsulation Overhead  Voice is sent in small packets at high packet rates.  IP, UDP, and RTP header overheads are enormous: For G.729, the headers are twice the size of the payload. For G.711, the headers are one-quarter the size of the payload.  Bandwidth is 24 kbps for G.729 and 80 kbps for G.711, ignoring Layer 2 overhead.

9 © 2006 Cisco Systems, Inc. All rights reserved. RTP Header Compression  Compresses the IP, UDP, and RTP headers  Is configured on a link-by-link basis  Reduces the size of the headers substantially (from 40 bytes to 2 or 4 bytes): 4 bytes if the UDP checksum is preserved 2 bytes if the UDP checksum is not sent  Saves a considerable amount of bandwidth

10 © 2006 Cisco Systems, Inc. All rights reserved. cRTP Operation ConditionAction The change is predictable. The sending side tracks the predicted change. The predicted change is tracked. The sending side sends a hash of the header. The receiving side predicts what the constant change is. The receiving side substitutes the original stored header and calculates the changed fields. There is an unexpected change. The sending side sends the entire header without compression.

11 © 2006 Cisco Systems, Inc. All rights reserved. When to Use RTP Header Compression  Use cRTP: Only on slow links (less than 2 Mbps) If bandwidth needs to be conserved  Consider the disadvantages of cRTP: Adds to processing overhead Introduces additional delays  Tune cRTP—set the number of sessions to be compressed (default is 16).

12 © 2006 Cisco Systems, Inc. All rights reserved. Factors Influencing Encapsulation Overhead and Bandwidth FactorDescription Packet rate–Derived from packetization period (the period over which encoded voice bits are collected for encapsulation) Packetization size (payload size) –Depends on packetization period –Depends on codec bandwidth (bits per sample) IP overhead (including UDP and RTP) –Depends on the use of cRTP Data-link overhead–Depends on protocol (different per link) Tunneling overhead (if used) –Depends on protocol (IPsec, GRE, or MPLS)

13 © 2006 Cisco Systems, Inc. All rights reserved. Bandwidth Implications of Codecs  Codec bandwidth is for voice information only.  No packetization overhead is included. CodecBandwidth G kbps G.726 r3232 kbps G.726 r2424 kbps G.726 r1616 kbps G kbps G.7298 kbps

14 © 2006 Cisco Systems, Inc. All rights reserved. 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)

15 © 2006 Cisco Systems, Inc. All rights reserved. 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) 40 VoIP packet size (bytes) Packet rate (pps)

16 © 2006 Cisco Systems, Inc. All rights reserved. Data-Link Overhead Is Different per Link Data-Link Protocol Ethernet Frame Relay MLP Ethernet Trunk (802.1Q) Overhead [bytes]

17 © 2006 Cisco Systems, Inc. All rights reserved. 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.

18 © 2006 Cisco Systems, Inc. All rights reserved. Extra Headers in Security and Tunneling Protocols ProtocolHeader Size (bytes) IPsec transport mode30–53 IPsec tunnel mode50–73 L2TP/GRE24 MPLS4 PPPoE8

19 © 2006 Cisco Systems, Inc. All rights reserved. Example: VoIP over IPsec VPN  G.729 codec (8 kbps)  20-ms packetization period  No cRTP  IPsec ESP with 3DES and SHA-1, tunnel mode

20 © 2006 Cisco Systems, Inc. All rights reserved. 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

21 © 2006 Cisco Systems, Inc. All rights reserved. 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.

22 © 2006 Cisco Systems, Inc. All rights reserved. Bandwidth Calculation Example

23 © 2006 Cisco Systems, Inc. All rights reserved. Quick Bandwidth Calculation Total packet size Total bandwidth requirement ————————— = ———————————————— Payload size Nominal bandwidth requirement Total packet size = All headers + payload ParameterValue Layer 2 header6 to 18 bytes IP + UDP + RTP headers40 bytes Payload size (20-ms sample interval)20 bytes for G.729, 160 bytes for G.711 Nominal bandwidth8 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

24 © 2006 Cisco Systems, Inc. All rights reserved. VAD Characteristics  Detects silence (speech pauses)  Suppresses transmission of “silence patterns”  Depends on multiple factors: Type of audio (for example, speech or 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

25 © 2006 Cisco Systems, Inc. All rights reserved. VAD Bandwidth-Reduction Examples Data-Link Overhead Ethernet 18 bytes Frame Relay 6 bytes Frame Relay 6 bytes MLPP 6 bytes IP overheadno cRTP 40 bytes cRTP 4 bytes no cRTP 40 bytes cRTP 2 bytes CodecG kbps G kbps G kbps G kbps Packetization20 ms 160 bytes 30 ms 240 bytes 20 ms 20 bytes 40 ms 40 bytes Bandwidth without VAD 87.2 kbps66.67 kbps26.4 kbps9.6 kbps Bandwidth with VAD (35% reduction) kbps43.33 kbps17.16 kbps6.24 kbps

26 © 2006 Cisco Systems, Inc. All rights reserved. Enterprise Voice Implementations  Components of enterprise voice networks: Gateways and gatekeepers Cisco Unified CallManager and IP phones

27 © 2006 Cisco Systems, Inc. All rights reserved. Deploying 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.

28 © 2006 Cisco Systems, Inc. All rights reserved. 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.

29 © 2006 Cisco Systems, Inc. All rights reserved. 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

30 © 2006 Cisco Systems, Inc. All rights reserved. Cisco Unified CallManager Functions Call processing Dial plan administration Signaling and device control Phone feature administration Directory and XML services Programming interface to external applications Cisco IP Communicator

31 © 2006 Cisco Systems, Inc. All rights reserved. Example: Signaling and Call Processing

32 © 2006 Cisco Systems, Inc. All rights reserved. Enterprise IP Telephony Deployment Models Deployment ModelCharacteristics Single site–Cisco Unified CallManager cluster at the single site –Local IP phones only Multisite with centralized call processing –Cisco Unified CallManager cluster only at a single site –Local and remote IP phones Multisite with distributed call processing –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

33 © 2006 Cisco Systems, Inc. All rights reserved. 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.

34 © 2006 Cisco Systems, Inc. All rights reserved. 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.  CAC is used to limit the number of VoIP calls, and AAR is used if WAN bandwidth is exceeded.  Cisco SRST is located at the remote branch. Note: Cisco Unified CallManager cluster can be connected to various places depending on the topology.

35 © 2006 Cisco Systems, Inc. All rights reserved. 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.

36 © 2006 Cisco Systems, Inc. All rights reserved. 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.

37 © 2006 Cisco Systems, Inc. All rights reserved. Basic Cisco IOS VoIP Voice Commands

38 © 2006 Cisco Systems, Inc. All rights reserved. Voice-Specific Commands dial-peer voice tag type router(config)#  Use the dial-peer voice command to enter the dial peer subconfiguration mode. destination-pattern telephone_number router(config-dial-peer)#  The destination-pattern command, entered in dial peer subconfiguration mode, defines the telephone number that applies to the dial peer.

39 © 2006 Cisco Systems, Inc. All rights reserved. Voice-Specific Commands (Cont.) 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. 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)#


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