HUAWEI UC Technologies and Standards Sales Specialist Training

HUAWEI UC Technologies and Standards
Sales Specialist Training
Topics




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

Overview
Basic Concepts of Enterprise Voice Systems
Voice System Networks and Reliability
VoIP Concepts and Protocols
Basic Concepts of Video
Basic Concepts of VoIP QoS
VoIP and IP Networks
2
What Can IP Telephony & Unified
Communications Do?

Facilitate voice communication among users within
enterprises







Unified numbering within the dedicated network,
simplifying memorization, and dialing
More telephony services; for example, Calling Line
Identification Presentation (CLIP) and callback
...
Facilitate communication with customers

Unique PBX number that is easy to remember
The PBX can help customers reach a desired
enterprise user
...
Improve team collaboration capabilities




Organization-matching functions, including searching a
group, secretarial service, and call pickup.
Voice and data conferencing functions
...
3

Save costs for enterprises

VoIP calling saves toll call fees

Save training fees

Enable efficient call routing

Billing

...
Enhance work efficiency

Integrate communication services
into service systems


...
Support enterprise business
development

Hotel

Production and dispatching

...
IP Telephony (IPT) and
Unified Communications (UC)
enable enterprises to improve
work efficiency, service quality,
and their enterprise image
while reducing costs.
Evolution of Enterprise Communications

The first modernized Private Branch Exchange (PBX) or Private Automatic Branch Exchange (PABX) was developed
in the middle 1970s, during the same period as Carriers’ “stored program control” exchanges. The PBX has changed
in the following stages: IP-enabled PBX, IP PBX, and SIP-based IP PBX. Unified Communications (UC) was derived
from PBX. Unified Communications and Collaboration (UC&C) evolved from UC.
Commercially
available
Mature
Commercially
available
Mature
Commercially
available
Mature
Commercially
available
Mature
1970s
1980s
1998
2008
2005
2015
2010
2020
PBX
•More efficient
communication tool
•Reduced communication
costs
IPT
•Beginning of IP era; reduced
network costs
•Further reduced
communications costs
•More flexible deployment
4
UC
•Integrates communication
tools
•Changes communication
modes
•Improves working
efficiency
UCC/CEBP
•Integrates communication
services into service systems
•Changes work modes
•Improves Enterprise
competitiveness
Differences between Enterprise and Carrier
Voice Communications Services



PBXs concentrate on users’ experience, user information security, convenience of creating their own private
networks, and ensure service quality over transmission networks with limited performance. PBXs have less
impact on O&M and billing.
Currently, there are hundreds of PBX services provided by mainstream vendors. However, Carriers can
provide only some supplementary services.
In addition to providing telephony services, PBXs:







Permit enterprise users to make internal calls free of charge
Help enterprises create private voice networks, reducing toll call fees
Permit enterprises to control their own private voice networks
Conveniently add applications such as voice mailbox, unified messaging, voice dispatching, emergency paging system, and
hotel voice mail applications
Easily interconnect with system functions such as broadcasting, electric control switch, alarms, and wireless trunking
...
These advantages of PBXs ensure that enterprises’ voice communication services will be a growing market
for a long time to come.
5
Benefits of Enterprise Communications
Endpoints
•Enterprise endpoints offer comprehensive functions. The popular IP phone provides some unique functions.
However, most of the functions have been available since the digital phone era.
•Using dedicated enterprise endpoints, users can manage audio services on PBXs simply by pressing buttons.
This frees users from memorizing complicated function access codes. The price of a medium-range endpoint
ranges from USD $100 to USD $200.
6
Topics


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


Overview
Basic Concepts of Enterprise Voice
Systems
Voice System Networks and Reliability
VoIP Concepts and Protocols
Basic Concepts of VoIP QoS
VoIP and IP Networks
7
Telephone Switching
Addressing and
connecting the call
Placing a call
Answering the call
Program control era: converting from analog to digital and from manual to computerized IP era:
converting from Time Division Multiplex (TDM) to Packets (basic principles and structure have not
changed).
8
PSTN and PBX

Public Switched Telephone Network (PSTN) is a telecommunications network established to
perform telephone services for public subscribers

PBX

A device on the user side from a Carrier’s perspective

Evolved from the Integrated Services Digital Network (ISDN) in the TDM era, and quickly
transitioned from H.323 to Session Initiated Protocol (SIP) in the IP era
Analog
POTS
Handset
Trunks
POTS
Handset
Trunks
Trunks
SS7
PSTN
PBX
9
SS7
Switch B
ISDN PRI/BRI
Switch A
POTS
Handset
Switch C
Signaling System No. 7 (SS7)
•Communications devices follow a specified communications protocol to transmit control information to destination
devices in a secure, reliable, and efficient manner. The transmitted information is called “protocol control
information” in computer networks, and “signal” or “signaling” in telecommunications networks.
•Signaling is divided into subscriber signaling and inter-office signaling. Subscriber signaling is used between user
terminals (for example, telephones and PBXs) and the PSTN, while inter-office signaling is used within the PSTN.
•SS7 is an inter-office signaling for Carriers and is termed Common Channel Signaling (CCS). SS7 plays a crucial
role in intelligent communication network development.
ITU-T definition
• SS7 core modules
Telephone User Part (TUP)
ISDN User Part (ISUP)
Only Huawei’s PBXs
support SS7, which is
our bidding
specification advantage
in China.
10
R2 Signaling and CNo.1 Signaling

R2 signaling is channel-associated signaling. It uses a signaling channel associated with the speech channel or the
speech channel itself to transmit the required control signals such as a busy signal, answer signal, release signal, and
dialing signal. In other words, R2 signaling uses one channel to transmit both speech information and associated
signaling.

China No.1 (CNo.1) signaling is a subset of the International R2 signaling system and is widely used by PSTN
networks in China

Outside of China, Mexico, and Brazil, R2 signaling is a special subset of the International R2 signaling system

Inter-office signaling and user-side signaling are not differentiated with R2, so all mainstream PBXs support R2

R2 signaling has simple functions, poor scalability, low efficiency, and small capacity, so R2 signaling has been totally
replaced by SS7 for PSTN
11
E1 and T1



E1 and T1 are standards for transmitting data over physical lines and are used for
signaling.
E1 is dominant in Europe. T1 is dominant in the U.S., Canada, Hong Kong, Taiwan,
and Japan (named J1 by some vendors).
 Similarities: the same sampling frequency (8 kHz), bits per code (8 bits), and timeslot
bit rate (64 kbit/s)
 Differences: an E1 has 32 timeslots and a data transmission rate of 2.048 Mbit/s. A
T1 has 24 timeslots and a data transmission rate of 1.544 Mbit/s.
 E1 adopts A-law coding/decoding of 13-segment while T1 adopts µ-law
coding/decoding of 15-segment.
Interfaces: unbalanced 75 ohm coaxial cable and balanced 120 ohm twisted pair based
on G.703.
12
PRI/BRI interface

PRI and BRI are used for subscriber signaling and are also called Digital Signal System 1 (DSS1) signaling.
PRI and BRI are products of the ISDN era and final products of the TDM voice era. The PSTN is used to
connect PBXs or endpoints

Primary Rate Interface/Primary Rate Access (PRI/PRA): 30B+D interface

Basic Rate Interface/Basic Rate Access (BRI/BRA): 2B+D interface and two types of physical interfaces

S/T interface: 4 lines and a transmission distance of 1.2 km (the BRI interfaces in some countries are S/T interfaces)

U interface: 2 lines, a transmission distance of 5 km, and converted to S/T interface through NT1
Type
B Channel (Media, kbit/s)
D Channel (Signaling, kbit/s)
BRA 2B+D
2B = 2 x 64 = 128
D = 16
PRA Europe 30B+D
30B+D = 30 x 64 = 1,920
D = 64
PRA North America 23B+D
23B+D = 23 x 64 = 1,472
D = 64
13
Q Signaling (QSIG)

QSIG is a protocol for Integrated Services Digital Network (ISDN) communications based on
the Q.931 standard, and is used for signaling between PBXs

QSIG supports a variety of functions such as basic calling, number display, name display, call
transfer, call forwarding, call back, message notification, and route optimization
14
FXO and FXS
FXO
FXS
PSTN
PBX

Foreign Exchange Office (FXO), also called an analog trunk, is an interface on a PBX to connect to
the PSTN. The FXO interface helps simulate the PBX as an analog phone to interact with Carrier
networks. In FXO connection mode, users can connect to extensions only through an operator service
or the automatic switchboard.

Foreign Exchange Station (FXS), also called a Plain Old Telephone Service (POTS) port, is an
interface on the PBX to connect to analog phones. This interface supplies battery power, provides dial
tone, and generates ringing voltage.
15
Differences between Digital Phones and
IP Phones
Digital Phone
IP Phone
Physical
connection
Twisted pair
Ethernet cable
Power supply
Centralized
PoE or local power supply
Transmission
protocol
2B+D
IP
IPT service
Comprehensive
Comprehensive+
Bandwidth service
N/A
Imaginative
Transmission
distance
A maximum of 1,200 meters
Distance that IP networks can reach
Power-off survival
Unsupported
Unsupported
Phone relocation
Troublesome
Simple
Extension mobility
Partly supported
Supported
Compatibility
Self-developed and incompatible
High compatibility with basic services
Hardware
specification
Low hardware configurations (for example,
screen and Bluetooth are not supported)
Technology keeps current with industry
16
Digital phones and IP phones do not
have significant differences in
appearance, but may have three
versions: digital phone, H.323, or SIP.
Topics







Overview
Basic Concepts of Enterprise Voice Systems
Voice System Networks and Reliability
VoIP Concepts and Protocols
Basic Concepts of Video
Basic Concepts of VoIP QoS
VoIP and IP Networks
17
Traditional TDM PBX (Digital SPC)
E1/T1/BRI/SS7/R2/
PRI digital trunk
User cable
PSTN
Equipment room
MDF
FXO analog trunk



Cables are the main disadvantage for
a traditional TDM because of
investment costs and maintenance.
Security and reliability of cables are
also poor. For example, cables are
prone to lightning strikes.
The advantages of analog phones
are that power can be supplied in a
unified manner, and phones can be
bridged.
SPC
Floor and building
distribution frame
Twisted pair
phone cable
18
Multiple twisted
pairs phone cable
Analog User Access with IP PBX
E1/T1/BRI/SS7/R2/
PRI digital trunk
IP PBX
User cable
PSTN
Equipment room
MDF
FXO analog trunk

Multiple twisted
pairs phone cable
Using Integrated Access
Devices (IADs) to provide
IAD
access for analog users, IP
PBX eliminates the
Twisted pair
phone cable
disadvantage of user cables.
19
Floor and building
distribution frame
IP User Access with IP PBX
E1/T1/BRI/SS7/R2/
PRI digital trunk
IP PBX
PSTN
Switch
FXO analog
trunk
IP network

IP phones can be powered
by switches that provide
Floor and building
switch
Power over Ethernet (PoE),
or use local power supplies

IP phones cannot be
bridged
20
Integrated
cabling
Topics
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Overview
Basic Concepts of Enterprise Voice Systems
Voice System Networks and Reliability
VoIP Concepts and Protocols
Basic Concepts of Video
Basic Concepts of VoIP QoS
VoIP and IP Networks
21
Main VoIP Standards: H.323 and SIP

H.323 is a general standard developed by the International Telecommunication Union (ITU) for sharing audio, video, and
data over data packet (IP) networks. Initially, H.323 was used in multimedia conferences. Later, it was expanded and
used by IP phones.



Has typical telecom features with advantages from TDM development to IP for Carrier networks

Provides unified processing and management
SIP is a multimedia signal protocol developed by the Internet Engineering Task Force (IETF).

Simple, modular, good expandability, and closely associated with Internet applications

Pushes the complexity of network devices to the edge of the network

Focuses on building VoIP networks based on the Internet
Common ground between H.323 and SIP:

Provide multimedia services over IP networks

Run on IP networks, use TCP, and UDP sessions to transmit signals; use RTP to transmit voice and video streams

Use existing protocols (such as G.711 and G.729) for encoding and decoding; do not require new encoding/decoding methods

Use a server as an intermediary for establishing sessions
•
H.323 VoIP network: a gatekeeper provides address translation, bandwidth control, certificate control, and area management functions
•
SIP network: an agent server processes and sends requests of user agents, directly establishes sessions with other user agents, and calls traditional
PSTN users through a gateway
22
Differences between H.323 and SIP

An important feature of SIP is that it does not define the types of sessions to be established. SIP defines only the method
for managing sessions. Based on this flexibility, SIP can be used in diversified applications and services, including
interactive games, on-demand music and video applications, and for voice, video, and data conferences.

By design, SIP is a distributed call model, and provides a distributed multicast function. The multicast function not only
facilitates conference control, but also simplifies user positioning, group invitation, and saves bandwidth. H.323 does not
support multicast.


Advantages of SIP:

Simple, easy to understand: SIP messages are in text format, while H.323 messages are in ASN.1 format

Extensible: many parts of SIP can be customized by users, while H.323 cannot

Expandable: SIP supports multi-domain searches

High efficiency: the process for establishing a SIP call is easier than for H.323

The cost for establishing an audio and video environment is low
Disadvantage of SIP:

The simpler,
the more popular
Lack of standards
23
Digitization Basics: PCM Voice Encoding

Pulse Code Modulation (PCM) samples analog signals such as
voice or image signals periodically to make them discrete, round,
and quantized sample values. Then, sample values are converted
to binary code to represent amplitude values of sample pulses.


Encoding process

Sampling

Quantizing

Coding
Traditionally, voice encoding uses an 8 kHz sampling rate, an 8-bit
depth to coding quantized values, and uses A-law or µ-law in the
coding process to finally obtain 64 kbit/s voice code.
24
Opus

Opus is a lossy audio codec, which was developed by the IETF for real-time voice transmission over networks.

In the competition among lossy audio formats, Advanced Audio Coding (AAC) was once very popular. However, Opus quickly
upstaged AAC. In low-bit-rate encoding, Opus outperforms HE AAC. In middle-bit-rate encoding, Opus is comparable to AAC with
30% higher bit rate. In high-bit-rate encoding, Opus is closer to the original voice. Therefore, Opus has broad uses in the future.


6 kbit/s to 510 kbit/s bit rate

8 kHz (narrowband) to 48 kHz (full-band) sampling rate

2.5 ms to 60 ms frame size

Constant Bit Rate (CBR) and Variable Bit Rate (VBR) support

Audio bandwidth from narrowband to full-band

Voice and music support

Monaural and stereo support

Up to 255 channels (frames of multiple data streams)

Dynamically adjustable bit rate, audio bandwidth, and frame size

Good robustness and Packet Loss Compensation (PLC)

Floating point and fixed-point implementation
When NetATE is enabled, packetization time and bit rate are automatically adjusted based on network conditions. This applies only
to Opus. Therefore, the bandwidth required by Opus is not fixed.
25
Topics







Overview
Basic Concepts of Enterprise Voice Systems
Voice System Networks and Reliability
VoIP Concepts and Protocols
Basic Concepts of Video
Basic Concepts of VoIP QoS
VoIP and IP Networks
26
International Standards for Video Encoding

Moving Picture Experts Group (MPEG) standards: include MPEG-1, MPEG-2, and MPEG-4 standards
developed by the International Organization for Standardization (ISO) and International Electrotechnical
Commission (IEC)

ITU-T standards: include H.261 and H.263 developed by ITU-T for video phones and conferences

H.264/Advanced Video Coding (AVC) (MPEG-4 Part 10) jointly developed by the ISO and ITU-T
27
H.263 (Past)

H.263 is a low-bit-rate video codec developed by the ITU-T for video conferences.

Initially, H.263 was designed to transmit data based on H.324 systems (that is, conducting video conferences or calls over a
PSTN or other networks that are based on circuit switching). Later, it was found that H.263 also can be successfully applied in
H.323 (video conference system based on an RTP/IP network), H.320 (video conference system based on an ISDN), RTSP
(streaming media transmission system), and SIP (video conference system based on the Internet).

H.263 provides better image quality than H.261 (designed for ISDN) in low bit rates. Here are the differences between
H.263 and H.261:

H.263 motion compensation uses half-pixel precision, while H.261 uses full-pixel precision and loop filter.

Some parts of the data stream hierarchy structure are optional in H.263. This allows a lower bit rate or better error correction to
be configured for H.263.

H.263 contains four negotiable options to improve performance.

H.263 uses unrestricted motion vectors and syntax-based arithmetic encoding.

H.263 uses the same frame prediction method as the P-B frame in MPEG.

H.263 supports five resolutions: in addition to Quarter Common Intermediate Format (QCIF) and Common Intermediate Format
(CIF) supported by H.261, H.263 also supports Sub-Quarter Common Intermediate Format (SQCIF), 4 x Common Intermediate
Format (4CIF), and 16 x Common Intermediate Format (16CIF). The resolution of SQCIF is half that of QCIF, while the
resolutions of 4CIF and 16CIF are 4 and 16 times that of CIF, respectively.
28
H.264 (Now)

After H.263, the next-generation video codec developed by the ITU-T (jointly with the MPEG) was H.264, which is also
called AVC or MPEG-4 Part 10.

H.264 introduces many new compression technologies such as multiple reference frames, multi-block type, integral
transform, and intra-frame prediction, and uses finer sub-pixel motion vectors (1/4 and 1/8) and a next-generation loop
filter to improve compression performance and provide a complete system.

Compared with H.263+ and MPEG-4 SP, H.264 saves up to 50% of the bit rate, greatly reducing storage capacity.

H.264 provides better video quality in various resolutions and bit rates.

H.264 adopts a simple and clear design, uses simple syntax description, avoids excess options and configurations,
and utilizes existing encoding modules (as many as possible).

H.264 can be easily combined with low-bit-rate codecs such as G.729 for a complete system.

H.264 features low delay and flexibly uses appropriate delay limits for different services.

H.264 enhances error code and packet loss processing to improve the decoder’s error correction.

H.264 provides higher network adaptability, using network infrastructure and syntax to adapt to both IP and mobile
networks as well as applications.
29
H.265 High-Efficiency Video Coding (Future)

Huawei holds various core patents built on H.265, and is the dominant player.

H.265 aims to transmit network videos with higher quality using limited bandwidth. Using half the
bandwidth originally required, H.265 can provide videos with the same quality. Compared with H.264,
H.265 can reduce video size from 39% to 44%, while ensuring the same video quality.

When the bit rate is reduced by 51% to 74%, H.265 can still provide video quality equal to or even better
than H.264. Essentially, H.265 provides better Peak Signal to Noise Ratio (PSNR) than expected.

H.265 also supports ultra HD videos such as 4k (4096 x 2160) and 8k (8192 x 4320) videos.

H.263 can transmit SD broadcast digital TV videos (720 x 576 that comply with CCIR601 and CCIR656)
at a bandwidth of 2 Mbit/s to 4 Mbit/s. Based on algorithm optimization, H.264 can transmit SD digital
images at bandwidth lower than 2 Mbit/s. H.264 HD can transmit 1080p full HD videos at a bandwidth
lower than 1.5 Mbit/s.
30
Topics







Overview
Basic Concepts of Enterprise Voice Systems
Voice System Networks and Reliability
VoIP Concepts and Protocols
Basic Concepts of Video
Basic Concepts of VoIP QoS
VoIP and IP Networks
31
Voice Quality Factors Using VoIP
Common methods for improving voice quality:
•
Packet Loss Compensation (PLC)
•
Dynamic Jitter Buffer (DJB)
•
Automatic Echo Cancellation (AEC)
•
Automatic Noise Suppression (ANS)
•
Automatic Silence Compression (ASC)
•
Voice Activity Detection (VAD)
•
Comfort Noise Generation (CNG)
•
Automatic Gain Control (AGC)
Voice quality evaluation method: Mean Opinion Score (MOS) test
Main factors affecting VoIP voice quality
Now we can see why vendors do not
guarantee VoIP voice quality over 3G,
4G, Wi-Fi networks, or the Internet!
32
Level
MOS Score
User Experience
Excellent
5.0
Very clear, no distortion, and no delay
Good
4.0
Clear, low delay, and a little noise
Fair
3.0
Not very clear, and obvious delay, noise, and distortion
Poor
2.0
Not very clear, loud noise, intermittent, and serious
distortion
Bad
1.0
Silent or absolutely unclear, and very loud noise
User Experience with Delay
Delay defined by the ITU in G.114
Range (ms)
Description
0–150
Acceptable to most users and
applications
150–400
Acceptable in certain conditions
> 400
Unacceptable
User experience
Range (ms)
Description
0–100
Not felt by most people
100–300
Slightly affected
> 300
Obvious feel
Note: Delay refers to end-to-end delay. Considering other factors, such as encoding/decoding
and buffering, network delay should be controlled within 200 ms.
33
Packet Loss Concealment (PLC) Algorithm
Missing frame
Restored frame
Packet loss
detector
1 or 0
Frame
estimator
Frame
buffer
4
5
6
8
9
Frame count
Feature vector stream from network


To recognizer
VoIP transmits voice data over an IP network using UDP. On an IP network, packet loss will unavoidably occur. To
minimize effects from packet loss, the PLC algorithm can be used to reconstruct missed frames based on the
correlation inside voice information, ensuring received voice quality.
If the packet loss rate is high, not all missed frames can be compensated through calculation. However, voice
services are not as sensitive to packet loss.
34
Dynamic Jitter Buffer (DJB)
PBX
MG
IP network
Jitter buffer
20 ms
20 ms
20 ms

23 ms
20 ms
20 ms
C
B
A
C
B
A
C
B
A
50
30
10
50
30
10
50
30
10
RTP timestamp
20 ms frame interval

IP users
RTP timestamp
20 ms frame interval
RTP timestamp
20 ms frame interval
A dynamic jitter buffer refers to a certain size buffer allocated by a gateway’s RTP media receiver, where RTP media packets
are buffered, sorted, and discarded. The main function of the buffer is to reduce the impact of network jitter on voice service.
When there is no network jitter, the buffer can be disabled; when network jitter is large, the size of the buffer can be increased.
Buffering results in network delay. A larger delay is better to filter jitter. A DJB using an excellent algorithm can find the balance
between delay and jitter, and enable good queuing and timely discards to obtain perfect Internet voice quality.
35
Topics







Overview
Basic Concepts of Enterprise Voice Systems
Voice System Networks and Reliability
VoIP Concepts and Protocols
Basic Concepts of Video
Basic Concepts of VoIP QoS
VoIP and IP Networks
36
Power Supply Methods for IP Phones
Independent power
module
Customer LAN
Desktop PC
switch
This situation illustrates what to do when a customer’s LAN switch is old. The wall-mounted power supply for IP phones must
have Uninterruptible Power Supply (UPS) capability. A PoE module can be directly attached to a LAN switch port for use with
old devices.
Desktop PC
Any PoE switch that complies with the 802.3af PoE standard
or media gateway with a built-in PoE switching port
Mainstream IP phones have PoE capability. PoE enables UPS deployment and LAN switch requirements in a unified
manner, which simplifies desktop cabling.
37
Why Don’t H.323 and SIP Packets
Support NAT Traversal?

H.323 and SIP protocol packets write original address information at the application layer.
Network Address Translation (NAT) only converts addresses at the network layer. When the
destination end receives a packet, and finds that the address at the application layer (original
address) is different than the address at the network layer (address after NAT), the destination
end discards the packet.

A Session Border Controller (SBC) functions as the NAT device at the application layer. An SBC
is required because it converts addresses at the network layer in addition to the application layer.
38
HUAWEI ENTERPRISE ICT SOLUTIONS A BETTER WAY
Copyright © 2014 Huawei Technologies Co., Ltd. All Rights Reserved.
The information in this document may contain predictive statements including, without limitation, statements regarding the future financial and operating results, future product portfolio,
new technology, etc. There are a number of factors that could cause actual results and developments to differ materially from those expressed or implied in the predictive statements.
Therefore, such information is provided for reference purpose only and constitutes neither an offer nor an acceptance. Huawei may change the information at any time without notice.
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