VOICE over IP – H.

323
Advanced Computer Network
SS2005

Presenter : Vu Thi Anh Nguyet

1
 Outlines

1. Introduction
2. QoS in VoIP
3. H323
4. Signalling in VoIP
5. Conclusions

2
 1. Introduction to VoIP

Voice over IP – the transmission of digitalized voice over

packet-switched IP networks

PSTN

Class 5 City B
V
IP Network Class 5
V
City A
PSTN

3
VoIP Advantages

• Lower costs per call

• Lower infrastructure costs
• New advanced features

4
 VoIP Packet Format

• Link layer size vary per media

• Using UDP protocol without TCP
• Voice carried using the RTP protocol
• Payload size depend on codec type

5
 2. Quality of Service (QoS )

• QoS in a packet network is characterized by the main

parameters as:
- Bandwidth
- Delay
- Packet loss

6
 VoIP Bandwidth

• Total packet size = (L2 header: MP or FRF.12 or

Ethernet) + (IP/UDP/RTP header) + (voice payload size)
• PPS = (codec bit rate) / (voice payload size)
• Bandwidth = total packet size * PPS

7
 VoIP Bandwidth (cont.)
Example:
A G.729 call (8 Kbps codec bit rate) with cRTP and
the default 20 bytes of voice payload requires:
Total packet size (bytes) = (MP header of 6 bytes) + (compressed IP/UDP/RTP
header of 2 bytes) + (voice payload of 20 bytes) = 28 bytes
Total packet size (bits) = (28 bytes) * 8 bits per byte = 224 bits
PPS = (8 Kbps codec bit rate) / (160 bits) = 50 pps
(160 bits = 20 bytes (default voice payload) * 8 bits per byte

Bandwidth per call

= voice packet size (224 bits) * 50 pps = 11.2 Kbps

8
 Delay

• CODEC

Voice Path
• Packetization
• Output queuing
Loss • Access (up) link transmission
+
Delay • Backbone network transmission
• Access (down) link transmission
• Input queuing
• Jitter buffer
• CODEC

9
 Fixed Delay Components (cont.)
Propagation Delay
Serialization Delay—
Buffer to Serial Link
Processing Delay

• Propagation—6 microseconds per kilometer

• Processing
- Coding / compression
- Decoding / decompression
- Packetization
• Serialization
10
Variable Delay Components
(cont.)

Queuing Queuing Queuing

Delay Delay Delay

Jitter
Buffer

• Queuing delay
• Jitter buffer

11
 Jitter

Receiver
Sender
Network

Variation of interpacket arrival time

A B C Sender

A B C Receives

D1 D2 = D1 D3 = D2 t

12
 Total Delay Time

Total delay for above example : 167 ms

ITU-T: <150ms : not detectable
= 150 –200ms : Acceptatble quality
>200 - 300ms : unacceptable quality
13
 Packet Loss

missing packet
G.729 vocoder algorithm

• The total of number of lost packets can be accepted 5%

14
 QoS Remarks

• VoIP frames have to traverse an IP network which is

unreliable.
• Frames may be dropped as a result of network
congestion or data corruption.
• For real-time traffic like voice, retransmission of lost
frames at the transport layer is not practical because
of the additional delays.
•Voice terminals have to deal with missing voice
samples, also referred to as frame erasures.

15
 3. H.323 Standards

• H.323 is a standard that defines how voice and video

devices can communicate. It specifies both signaling
characteristics and host-to-host communication protocols

16
 H.323 Standards (cont.)

• The H.323 standard consists of the following components

and protocols:
Protocol: Feature:
• H.225 Call Signalling
• H.245 Media Control
• G.711,G.722, G.723,G.728,G.729 Audio Codes
• H.261, H.263 Video Codes
• T.120 Data Sharing
• RTP/RTCP Media Transport

17
 H.323 Components

H.323
MCU
GK
e

H.323 Packet H.323

Gatekeeper Network Terminal

H.323
V
Gateway

PSTN ISDN

V.70 H.324 Speech H.320 Speech

Terminal Terminal Terminal Terminal Terminal

18
 Gateway

• The H.323 gateway reflects the characteristics of a

Switches Circuit Network (SCN) endpoint and H.323
endpoint.
• It converts voice and fax calls, in real time, between the
PSTN and an IP network.
• Gateways work as an H323 terminal.
• Gateways are not needed unless the interconnection with
the PSTN is required.

19
 Gatekeeper

• An optional H.323 Component

• Defines H.323 Zone
• Provides Centralized Call Control
• Mandatory and Optional Services

20
 Gatekeeper Mandatory Services
(cont.)
• Address Translation
Translates H.323 aliases (e.g. sliu@cisco.com) or
E.164 addresses (standard phone numbers) into IP
transport addresses (e.g. 10.1.1.1 port 1720)
• Admissions Control
Authorizes access to the H.323 network
• Bandwidth Control
Manages endpoint bandwidth requirements
• Zone Management
Provides the above functions to all terminals,
gateways, and MCUs that register to it

21
 Gatekeeper Optional Services
(cont.)
• Call control signaling
Gatekeeper Routed Call Signaling (GKRCS)
• Call authorization
Restrict certain terminals, gateways, time of day
• Bandwidth management
Reject admission if bandwidth is not available
• Call management
Services include maintaining an active call list that
use to indicate busy terminals.

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 4. H. 323 Signaling

Admission Request H.225 (UDP)

RAS V
Admission Confirm
Gatekeeper

Setup
H.225 (TCP)
Alerting / Connect
Q.931
V
H.323
Gateway A Capabilities Exchange
H.245 (TCP)
Open Logical Channel V
Open Logical Channel Acknowledge H.323
Gateway B

RTP Stream
RTP Stream Media (UDP)
RTCP Stream

23
 RAS Messages

• RAS channel is established between endpoints and

Gatekeeper across an IP network.
• RAS channel is opend before any other channels which
are established.
• RAS messages are carried by the UDP connection,
perform registration, admission, bandwidth changes, etc.

24
 RAS Messages (cont.)

• GRQ/GCF/GRJ (Discovery)
GRQ : A multicast message sent by a GW GRQ
looking for the GK
GCF: The reply to a GW with it‘s transport
address GCF/GRJ

• RRQ/RCF/RRJ (Registration)
RRQ : sent from GW to GK RAS
channel address
RCF : sent from GK to GW to
confirm a GW registration

25
 RAS Messages (cont.)

• ARQ/ACF/ARJ (Admission)
ARQ:
– The GK assigned terminal identifier
– The type of call (point to point)
– The call model that the terminal is willing to use (direct or GK
routed)
– The destination address (Ex: E.164 address)
ACF:
– The call model in use
– The transport address and port to use for Q.931 call signalling
– The allowed bandwidth for the call
26
 RAS Messages (cont.)

• DRQ/DCF/DRJ (Disconnect)
Get rid of call state
• LRQ/LCF/LRJ (Location)
Stateless name - IP address resolution
Inter gatekeeper communication
• IRQ/IRR (Information Request)
Ping during active calls
Resource information for gateways
• BRQ/BCF/BRJ (Bandwidth)
Ask for more/less bandwidth during call
• URQ/UCF/URJ (Unregistration)
Get rid of registration state
27
 RAS Message Exchange
(cont.)
Gatekeeper A Gatekeeper B
LRQ
LCF

ACF ACF

IP Network

ARQ
H.225 (Q.931) Setup ARQ
H.225 (Q.931) Connect

H.245
V V
Gateway A
Gateway B
RTP Phone B

Phone A

28
H.225 Call Control (ISDN Q.931)

• Setup
Incoming call
• Call Proceeding
• Alerting
Phone is ringing
• Connect
Media cut through (used for billing)
• Release/Release Complete
Tear down call

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 H.245 System Control

• Capabilities Exchange
Exchange the capabilities between two entpoints –
entpoint‘s transmit and receive capabilities for audio,
video, data.
• Master/ Slave Determination
• Open Logical Channel/Ack
The channel is set up before the actual transmission to
ensure the entpoints are ready and capable of receiving
and decoding information.

30
 SS7 Interconnect for Voice
Gateway Call Setup
SC A SC B
GW A GK A GK B GW B

PSTN/SS7 PSTN/SS7

1. IAM
2. Setup H.323
3. Call proc
4. ARQ
Phone A 5. LRQ

6. LCF
7. ACF
8. H225
Setup
9. ARQ

SS7 Q. 931 10. ACF

11. Setup
12. IAM
14. H225
13. Call proc
Call proc
17. H225 15. ACM
16. Alerting
Alert
18. Alerting
19. ACM
20. ANM
21. Connect
23. H225
Connect 22. Con. ACK
24. Connect
Phone B
25. Con. ACK
26. ANM
Connection Established
31
 VoIP Configuration
POP A
SC GK

PSTN N x E1
N x E1 outbound
VV
V GW

E1
POP B

PSTN N x E1
VV
N x E1 V GW outbound
Router
SLT

E1
SC GK

PSTN N x E1
N x E1 outbound
VV GW
V GW
HNI
POPPOP
C

32
 5. Conclusions

• One of the major motivations of developing VoIP

networks is the cost benefit.
• QoS provides reduced delay and fewer dropped packets
of voice traffic to ensure the good voice quality to
customers.
• H.323 is probably the most important standard supporting
packetized voice technology. However it is also the
complex standard with many protocols .

33
 References

[1] J. Davidson, J. Peters, “Voice over IP Fundamentals“, Cisco Press,

2000.
[2] O. Hersent, D. Gurle, J. P. Petit, “IP Telephony Packet-based
multimedia communications systems“, Addison-Wesley, 2000.
[3] L. L. Peterson, B. S. Davie, “Computer Networks - A Systems
Approach“, 2nd Edition, Morgan Kaufmann, 2000.
[4] J. Walrand, P. Varaiya, “High-Performance Communication Networks“,
2nd Edition, Morgan Kaufmann, 2000.
[5] http://www.cisco.com/
[6] http://www.fcc.gov/voip/
[7] http://www.callback4u.com/voice-over-ip/
[8] Training documents, “Cisco Advance Services“, 2002.
[9] Training documents, “Cisco Voice over IP (CVOICE)“, 2002.
[10] Paul J. Fong, Eric Knipp, Charles Riley,“Configuring Cisco Voice over
IP“, Syngress, 2002.

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