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© 2006 Cisco Systems, Inc. All rights reserved. Optimizing Converged Cisco Networks (ONT) Module 2: Cisco VoIP Implementations.

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Presentation on theme: "© 2006 Cisco Systems, Inc. All rights reserved. Optimizing Converged Cisco Networks (ONT) Module 2: Cisco VoIP Implementations."— Presentation transcript:

1 © 2006 Cisco Systems, Inc. All rights reserved. Optimizing Converged Cisco Networks (ONT) Module 2: Cisco VoIP Implementations

2 © 2006 Cisco Systems, Inc. All rights reserved. Lesson 2.3: Encapsulating Voice Packets for Transport

3 © 2006 Cisco Systems, Inc. All rights reserved. Objectives  Compare and contrast voice transport in circuit- switched and VoIP networks.  Describe causes of and solutions for jitter.  Explain the issues with IP, TCP, and UDP when transporting voice packets.  Describe encapsulation overhead issues for VoIP.  Describe header compression and when and where it should be used.

4 © 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

5 © 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

6 © 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

7 © 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.

8 © 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

9 © 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.

10 © 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.

11 © 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

12 © 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.

13 © 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).

14 © 2006 Cisco Systems, Inc. All rights reserved. Self Check 1.What causes jitter? 2.Explain why IP is not well suited to voice transmission. 3.What issues does TCP have when considering it as the protocol for voice? 4.What guidelines can be used to determine when and where to use RTP header compression? 5.What are some disadvantages to using RTP header compression?

15 © 2006 Cisco Systems, Inc. All rights reserved. Summary  Circuit-switched calls use dedicated links. VoIP networks send voice in packets.  Jitter is caused by packets arriving at the destination at varying rates and not in the original order.  IP, TCP or UDP alone cannot be used for voice packets. IP does not guarantee reliability, flow control, error detection or error correction. TCP has unnecessary overhead. UDP needs additional functionality offered by RTP.  Encapsulation overhead for VoIP can be very large. Since voice packets are small, compression should be used to compress headers.

16 © 2006 Cisco Systems, Inc. All rights reserved. Q and A

17 © 2006 Cisco Systems, Inc. All rights reserved. Resources  Examples of Jitter impact on quality  Understanding Jitter in Packet Voice Networks (Cisco IOS Platforms) e09186a df.shtml

18 © 2006 Cisco Systems, Inc. All rights reserved.


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