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SIP, VoIP and QoS

PLEASE READ CAREFULLY
This PDF may help you make notes as you work through the online course. It is
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being revoked plus any additional action that Vocale Ltd deems necessary. 1

Topics Topics

Voice over IP (VoIP) Refresher

Voice sampling and Codecs

The Real Time Protocol (RTP) and

The Real Time Control Protocol
(RTCP)

Quality of Service and VoIP

Assured SIP Services (AS-SIP)

Where does SIP and SDP fit in?

2

What is VoIP?

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What is Voice over IP?

Router
Switch Router Switch

WAN

Traditional Call Path, Dedicated

A B

VOIP Call Path, Full utilisation of links

A B
C D
E F
4

VoIP – ‘A Basic Call’

Hmm, no TCP and UDP

ACK – better
resend

Data 1

Data 2

Data 3

Data 4

Internet ‘Black Hole’

Transmission Control Protocol

User Datagram Protocol

Voice 1
No resends!
Voice 2
No Acknowledgements
Voice 3

Voice 4
‘Real Time’ Applications
6

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VoIP and TCP / UDP

SIP RFC 3261 states:

“All SIP elements MUST implement UDP and TCP. SIP elements
MAY implement other protocols.”

UDP is implemented in a lot of SIP elements as it was the required

protocol in the older SIP RFC 2543. As a lot of SIP products now
produce headers that are too big for UDP, TCP is being used more
and more as it has the ability to break up messages, re-assemble
them at the destination and cope with packet loss with
retransmissions.

E.g. Microsoft’s OCS and Lync systems only support TCP for
signalling.

A Note on TCP

This SIP Message is too big for a TCP Segment

8

VoIP over the Internet

Internet

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Branch to Branch VoIP

Private Data Network

Private
WAN
(Gateway ) Gateway
IP to TDM (TDM to IP)

10

Signaling Paths
IP PBX DHCP
DNS
TFTP
NTP
Analog
SIP
T1/E1
PRI
PSTN

Access / Media Trunk Gateway

Gateway

‘Branch’ IP PBX ITSP

Private Internet
IP WAN
SIP trunks

Softphone on PC

IP Phones
11

Speech Paths
IP PBX

Analog
SIP?
T1/E1
PRI
PSTN

Access / Media Trunk Gateway

Gateway

‘Branch’ IP PBX ITSP

Private Internet
IP WAN

VoIP/RTP
VoIP/RTP

Softphone on PC

IP Phones
12

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IP PBX Advantages

ITSP
PSTN
PSTN
VoIP

PRI line?

SIP trunk

IP

Single Cable infrastructure

Flexibility in Moving devices
IP Applications
Integrated Voicemail / Attendant
Unified Messaging
SIP trunking 13

Voice Sampling and Codecs

14

Encoding 1

This analogue waveform represents the words “Hi, how are you?”
and takes about 3 seconds to say.

Encoding is all about taking an analogue waveform, converting

it into digital information before it is sent to the intended recipient

15

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Encoding 2

Let’s zoom in on a very small

0.001 second part of the waveform

= a Sample

0.001 second

0.001 second of analogue speech would be represented by the bit stream

00110100 00110011 11001100 11001101 11001110 11001110 11001111 11001110
Digital telephony requires 64000 bits / second for normal speech
16

Encoding 3

= a Sample

0.001 second

0.001 second of analogue speech would be represented by the bit stream

00110100 00110011 11001100 11001101 11001110 11001110 11001111 11001110

This analogue signal has been sampled at a rate of 8 thousand samples a

second, each sample is represented by an 8 bit number.

A Codec that produces Binary data at a rate of 64Kbps is (usually)

working to the G.711 specification.

17

Encoding 4

Analogue
Analogue
waveform
waveform

Encoder
Binary
e.g. Data
G.711

Analogue waveform

De-coder
e.g. G.711

18

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Codecs for Voice

Codec Description MOS RTP/AVP

Payload Type
G.711 µ-Law This is an uncompressed codec for calls in North America and Japan 4.3 0

G.711 A Law This is the uncompressed version of the µ-Law codec for calls to 4.3 8
areas other than North America and Japan

G.729 This is a codec produces compressed voice by using a special model 3.7 18
(Annex A,B or J) called Code Excited Linear Prediction (CELP). Sample rate of (8
kbit/s)
G.723 This is a codec that compresses voice and uses Voice Activity 3.9 4
Detection (MOS for 6.3kbits sampling variant)

G.722 In it’s G.722.2 variant this codec can adapt it’s sampling rate in 4.3 9
reaction to network congestion. The less congestion the higher the
quality of the samples. It also samples at 16Khz to produce a superior
quality to other codecs

iLBC Internet Low Bit Rate Codec. This is a relatively new codec designed 3.8 Dynamic
to work well over the Internet

19

Try the ‘Codec Test’

[Link]

20

HD Voice

Consonants
Presence, Depth such as ‘s’ and So you think a MOS
Consonants of 4.3 is good
and Richness of ‘f’ can be clearly
sound heard enough now?

150 Hz 300 Hz 3.4 kHz 7 kHz

21

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Let’s add some Music

50 Hz >>>>> <<<< 19.2 kHz

[Link] 22

Wideband (HD) codecs (some of them)

G.722 [Link]

G.722.1 [Link]

G.722.2 [Link]

AMR-WB + [Link]

Speex [Link]

RTAudio [Link]

SILK [Link]

Opus [Link]

I can support G.722.1/G.722.2/SILK

G.729/G.711/G.722.2
I use
23

Opus IETF Audio Codec

24

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Opus examples

25

The Real Time Protocol (RTP)

26

The Real Time Protocol (RTP)

RTP is designed to support ‘real-time’ traffic such as voice and video

that is time sensitive. It can work with both Unicast and Multicast
applications. It provides services that include information such as:

– Payload Type Identification

– Sequence Numbering
– Timestamps
– And Delivery Information

RTP runs over UDP as TCP would be no good for ‘Real-Time’

operation

– RTP ‘Does Not’ provide any service that guarantees timely delivery of
the payload
– It ’Does Not’ provide any quality of service guarantees.
– It ‘Does’ rely on lower layer protocols to provide these extra services

27

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RTP Encapsulation

Voice element – G.711 encoding

= 20ms (approx 160bytes)
Layer 5 Binary DATA RTP RTP Time Stamp

Sequence Numbers

Layer 4 DATA UDP Header

Layer 3 DATA IP Header

Layer 2 CRC DATA LLC & MAC

Layer 1 10010101010010010101010011110011001

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Real Time Control Protocol (a)

The Real Time Control Protocol (RTCP) can be used alongside RTP in order to
provide information on the session and the participants

It carries an ID for a device that is transmitting the RTP data and this is called a canonical name or
CNAME (also known as an alias). Even if the Synchronization source changes (in the case of another
stream being introduced from the same device or machine) the CNAME remains the same. This helps
to identify participants in a session and is useful as these participants can join and leave sessions
dynamically.

30

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Real Time Control Protocol (b)

The Real Time Control Protocol (RTCP) can be used alongside RTP in order to
provide information on the session and the participants

It carries an ID for a device that is transmitting the RTP data and this is called a canonical name or
CNAME (also known as an alias). Even if the Synchronization source changes (in the case of another
stream being introduced from the same device or machine) the will CNAME remains the same. This
helps to identify participants in a session and is useful as these participants can join and leave
sessions dynamically.
The Types of RTCP Packet are:

SR - This is the Sender report. Shows the statistics on transmission and reception for the
participants in the session that are actively sending data

RR - This is the Receiver report and it shows the reception statistics from participants that
are not actively sending data in the session

SDES - The Source description items including identifying information such as the
CNAME and allows the binding of SSRC value with an actual id of the user.

GOODBYE - This Indicates end of participation in a session

APP - This denotes Application specific functions that will effect/interact with the session

XR – This is an RTCP extension that can help provide a ‘rich set’ of data for voice
management – if it is implemented in VoIP devices
31

RTCP-XR (Extended Reports)

RTCP XR is a VoIP management protocol that defines a set of metrics that contain information
for assessing VoIP call quality and diagnosing problems.

RTCP XR is defined in RFC 3611.

RTCP XR messages containing key call-quality-related metrics are exchanged periodically

between IP phones and gateways and this allows analyzing equipment to monitor these metrics to
assist in call quality analysis and troubleshooting.

The protocol measures VoIP call quality using these following key metrics:

Packet loss and discard rate and the distribution of lost and discarded
packets

Round-trip delay

Signal, noise and echo levels

Call quality in terms of estimated R factor or mean opinion score (MOS)

Configuration data such as jitter buffer size and configuration, and the type of
packet-loss concealment algorithm in use.

32

RTP / RTCP and UDP Ports

UDP (Layer 4)

30000 30001 40392 40393

RTP RTCP

Please note: RTP and RTCP use two different

sets of UDP port numbers
33

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Quality of Service (QoS)

34

What is Quality of Service?

Quality of Service is vitally important

when looking at provisioning Real
Time communication such as Voice
and Video across a network that is
being shared with Data.

QoS is not a single mechanism rather

it is something that can be achieved if
all of the elements in your network
(and your wide area network providers
network) are configured to recognise
Real Time communication streams and
give them the treatment or ‘priority’
they need in order to get to their
destination on time.
35

36

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37

38

QoS Issues

For ‘BEST’ voice quality

• One way end to end voice delay should be no
more than 150ms (ITU Recommendation)
• No more than 30ms Jitter (watch out at WAN
connections.)
• Test for delay and jitter with Ping

Pinging [Link] with 218 bytes of data:

Ping –l 218 [Link] or Ping –s on linux/unix

Reply from [Link]: bytes=218 time=40ms TTL=64

Reply from [Link]: bytes=218 time<10ms TTL=64
Reply from [Link]: bytes=218 time=10ms TTL=64
Reply from [Link]: bytes=218 time<10ms TTL=64 39

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Measuring Delay

Delay will be different for every site install dependent on the network installed. In a
Pinging [Link] with 238 bytes of data
fully switched Ethernet environment delay will not be a problem in normal
to simulate a G.711 voice frame
operation. However, routers will insert more delay into a network so voice over IP
connections via a router require careful engineering

Most manufacturers recommend that end to end delay be no more than 200ms.
Of this 200ms up to 80ms when an IP Set is connected through an IP PBX.
Allowing for up to 40ms delay in the PSTN without echo cancellation leaves 80ms
available for use in the customer’s network.

The International Telecommunication Union Telecommunication Standardization

Sector (ITU-T) G.114 recommendation actually specifies that for good voice
quality, no more than 150 ms of one-way, end-to-end delay (also called Latency)
should occur. Let’s work with that for the rest of this module

Network Reply
delay locations are varied, but include
from [Link]: such areas
bytes=238 as -: TTL=64
time=40ms
Reply from [Link]: bytes=238 time=40ms TTL=64

Reply from
Hardware [Link]:
Delay’s bytes=238 time=40ms TTL=64
through the network

Variable
Reply delays,
from e.g. Router Queues.
[Link]: bytes=238 time=40ms TTL=64

Transit delays (the time to traverse networks) 40

Processing delay in the end devices.

Jitter and Packet Loss

Reply from [Link]: bytes=238 time=25ms TTL=64

Reply from [Link]: bytes=238 time=120ms TTL=64
Reply from [Link]: bytes=238 time=40ms TTL=64
Reply from [Link]: bytes=238 time=100ms TTL=64

Packet loss means that a voice packet was sent, lost in transit and the far end never received
it. As long as packet loss stays low i.e. below 5%, this should not affect speech quality.

41
[Link]

General VoIP Acceptance Criteria

42

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QoS Everywhere!

Customer Site 1
IP/MPLS
Layer 3 Provider Network
Layer 2
Classification
Classification

LSR (Label Switching Routers)

Edge Routers
Customer Site 2

Check here!
Delay/Jitter/Packet Loss
Customer Site 2

Private WAN
Leased Line?

Layer 3

Layer 2
43

802.1Q – VLANs (1)

If the switch is
left in it’s default
state, i.e. VLANS
are not configured
then a broadcast
from any device
can hit any other
device connected
to the switch.

This may effect IP

phone (and VoIP)
performance

44

802.1Q - VLANs (2)

VLAN 1 VLAN 2
(Default)

The answer to this is to ‘split’ the switch internally (and logically)

into two separate Virtual LANs or VLANS. A VLAN has exactly
the same characteristics as a LAN and Traffic may only get from
one VLAN to the other via a device such as a Router (or Default
Gateway)
45

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802.1Q/P Tagging

CRC DATA Type - 0800 MAC Addresses

CRC DATA Type - 8100 Priority - VLAN ID MAC Addresses

Trunk or Uplink

MAC Addresses Type - 0800 DATA CRC

46

802.1P - L2 Classification

7
6

5
4
FTP RTP RTP

3
0

2
1

ingress egress

47

TOS and DiffServe

IP Datagram / Header

Header
Version Type of Service / DSCP
length

7 6 5 4 3 2 1 0

ToS

1 0 1 =5

DSCP

1 0 1 1 1 0 = 46

48

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Layer 3 Classification

HIGH

MEDIUM
Classify
Typical
NORMALDatagram Value = 0 Egress port

LOW
Voice Datagram Value = 46

Expedited Forwarding (EF) DSCP

49

Codecs and Bandwidth

Codec Bandwidth Bandwidth on an IP Switched Network MOS RTP/AVP

Payload
Type
G.711 µ-Law 64Kbit/second 4.3 0
Example:
G.711 A Law 64Kbit/second A G.729
G.711 Codec samples at a rate of 4.3 8
64Kbits/secondwith
8Kbits/second withaapayload
payloadgenerated
generated
every 20ms of 20bytes
160bytes/ /payload
payload
G.729 8Kbits/second 3.7 18
(Annex A,B or J) L2 Ethernet Header @ 18bytes

G.723 6.3Kbits/second G.711

L3 IP Header @ 20 bytes
L4 UDP Header @ 8 bytes
L5 RTP Header @ 12 bytes 3.9 4

Total of bytes per packet = 78

218bytes
bytes
G.722.1 48Kbits/second 4.3 9
Transmitting @ 50 Packets per second

Equals a total of 3900

10900bytes
bytes/ /second
secondoror
iLBC 13.33Kbits/second 3.8 Dynamic
87.2Kbits/second on the network
31.2Kbits/second

50

Symmetric DSL (SDSL)

SDSL Link from

Internet ISP
Download Bandwidth

ITSP
(for Trunks)

Upload Bandwidth

Caller
Symmetric DSL
• Same Upload as Download Speeds
• Usually lower contention ratios
• Often come with SLA
• Better calculate ‘VoIP allowance’
51

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Testing your link

[Link] 52

53

Attacks and Responses

54

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Service Provider Architecture

PROXY 1 PROXY 2

UA 1 AR 1 AR 2 UA 2

[Link]
55

Proxy and Access Router functions

PROXY 1 PROXY 2
Proxy
• Processing Authentication
• Maintaining ‘State’ information
• Understanding ‘other services’
• Verifying UA re: Precedence
requests
UA 1 AR 1 • Working ARwith
2 the Access Router
UA 2
• When to Pre-empt?
• Maintain all records

Access Router
• Transfer packets
• Packets, pass or stop?
56

Resource-Priority

INVITE sip:UserB@[Link] SIP/2.0

Via: SIP/2.0/TCP [Link];branch=z9hG4bK74bf9
Max-Forwards: 70
From: BigGuy ;tag=9fxced76sl
To: LittleGuy
Call-ID:3848276298220188511@[Link]
CSeq: 1 INVITE
Require: resource-priority (lowest) [Link]
Resource-Priority: [Link]
[Link]
Contact:
[Link]
[Link]
[Link]/rfc/[Link]
(highest) [Link]-override

57

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Accept-Resource-Priority

PROXY 1 200OK PROXY 2

200OK 200OK

Accept-Resource-Priority: [Link]

UA 1 AR 1 AR 2 UA 2

58

Cisco and Lync example video

[Link]
Jason Opdycke
59

Reason Header for Pre-emption Events

SIP UA1 SIP UA2 SIP UA3

INVITE

200 OK

ACK

Media Stream
INVITE

Reason: preemption ;cause=1 ;text="UA Preemption"

BYE
200 OK

200 OK
ACK

Media Stream

[Link]
header-for-preemption-04 60

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Proxy receives message

Doesn’t recognize namespace = Drop message
Does Recognizes and authorized it uses the
priority levels.
Not authorized = rejected
If UA doesn’t have a Priority level set, the Proxy
can assign a default.
If Auth and resources are all ok, treats as
normal
If Auth and resources not available – priority is
utilized
Proxy Maintains ‘state’ of all sessions

61

Multi-Level Pre-emption and Precedence

PROXY 1 PROXY 2

Better get
confirmation SIP Trunks x 5 I guess this
from Dave Capacity ~ 500Kbps Total is important
Assume 100k / call John?

UA 1 AR 1 AR 2 UA 2

I’ve been
dropped!

62

AS-SIP AS-SIP, Summary

Defines a Network Architecture

Built to carry secure signalling and media within

a domain / area,

MLPP that defines how important calls can get

through etc.

Priority – User defined / Network defined

Resource Priority

Accept Priority

Reason Header for Pre-emption

Open standard

AS-SIP
REASON HEADER
RESOURCE PRIORITY
MLPP 63

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SIP, SDP and VoIP

64

SIP, SDP and VoIP

65

Where does SIP fit in?

SDP Media Coding [G.711, G.723, G.720, iLBC etc.]

Applications SIP RTP RTCP DNS DHCP

Transport TCP UDP

Network IPv4, IPv6

Link & Physical ATM Ethernet Wireless

66

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67

68

Audio and Video in the SDP body

UDP (Layer 4)

49077 43713 40114 48227

Audio

Video

INVITE (SDP Element) 200 OK (SDP Element)

m=audio 49077 RTP/AVP 0 8 97 m=audio 40114 RTP/AVP 0
a=rtpmap:0 PCMU/8000 a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000 m=video 48227 RTP/AVP 32
m=video 43713 RTP/AVP 31 32 a=rtpmap:32 MPV/90000
a=rtpmap:31 H261/90000
a=rtpmap:32 MPV/90000

69

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Summary

In this module we have looked at Voice over IP along with the types of implementation you may
1 come across

We have looked at how voice is sampled along with focusing on some of the main codecs in use
2 along with a few suggestions on which to use for VoIP

As you’ve seen, RTP carries the actual data payload and RTCP carries control information. Again,
3 we looked at a trace to see exactly what is happening on the network re: RTP and RTCP

Quality of Service is something that all VoIP implementers must strive for. It is not a single solution
4 or device it is a culmination of good practices across the whole of the IP network

Assured SIP services may be something worth looking at if you want to be able to prioritize traffic
5 to ensure important call completion when traffic levels are high.

Of course we are really concerned about SIP and SDP so we looked at how SDP can convey
6 information to help SIP devices select functions such as which codec to use in a session
70

71

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73

End

74

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