Transcript CCNP-IV-ONT_Mod_2_Lesson_3

Optimizing Converged
Cisco Networks (ONT)
Module 2: Cisco VoIP Implementations
© 2006 Cisco Systems, Inc. All rights reserved.
Lesson 2.3:
Encapsulating Voice
Packets for
Transport
© 2006 Cisco Systems, Inc. All rights reserved.
Objectives
 Compare and contrast voice transport in circuitswitched 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.
© 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
© 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
© 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
© 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.
© 2006 Cisco Systems, Inc. All rights reserved.
Protocols Used for VoIP
Feature
Voice
Needs
TCP
Reliability
No
Yes
Reordering
Yes
Timestamping
Yes
No
Overhead
As little as
possible
Contains
unnecessary
information
Multiplexing
Yes
© 2006 Cisco Systems, Inc. All rights reserved.
Yes
Yes


UDP
No

RTP
No

No
Yes

No
Yes

Low
Yes


Low

No
Voice Encapsulation
 Digitized voice is encapsulated into RTP, UDP, and IP.
 By default, 20 ms of voice is packetized into a single IP
packet.
© 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.
© 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
© 2006 Cisco Systems, Inc. All rights reserved.
cRTP Operation
Condition
Action
The change is
predictable.
The sending side tracks the predicted
change.
The predicted change The sending side sends a hash of the
is tracked.
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.
© 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).
© 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?
© 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.
© 2006 Cisco Systems, Inc. All rights reserved.
Q and A
© 2006 Cisco Systems, Inc. All rights reserved.
Resources
 Examples of Jitter impact on quality
http://www.voiptroubleshooter.com/problems/jitter.html
 Understanding Jitter in Packet Voice Networks (Cisco
IOS Platforms)
http://cisco.com/en/US/tech/tk652/tk698/technologies_tech_not
e09186a00800945df.shtml
© 2006 Cisco Systems, Inc. All rights reserved.
© 2006 Cisco Systems, Inc. All rights reserved.