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Gateways Part 3
CONFIGURATION OF VOICE
PORTS
Voice Over IP: Chapter 1 Introducing Voice Gateways
Digital Trunks
The information about line and device states (on-hook,
off-hook, etc.) is communicated over digital lines using
signaling that emulates analog networks (FXS, FXO,
E&M)
For signaling to pass from a circuit-switched network
(like the PSTN) and a packet-switched network (like a
WAN) both networks must use the same type of
signaling
Digital Trunks
Lets review - digital lines use two types of signaling:
Channel Associated Signaling (CAS): Takes place within the
voice channel itself and is associated to each channel
Common Channel Signaling (CCS): Sends signaling
information over a dedicated channel and is not typically sent if
a channel is not in use
T1 CAS
Recall that a single digital voice channel requires 64
kbps of bandwidth (called a DS0):
64 kbps (64,000 bps) = 8000 samples/sec * 8 bits/sample
T1 CAS
T1 CAS uses in-band signaling by borrowing bits in the
actual voice channel to transmit signaling information
(sometimes referred to as robbed-bit signaling, or
RBS)
A bit is taken from every sixth frame of the voice data to
communicate on- or off-hook status, wink-start, groundstart, dialed digits, and other information about the call
Notice that these signaling types are the same that are
used by analog voice ports. They are simply
transmitted differently across digital trunks
T1 CAS
The eighth bit on every sixth sample in each DS0 is
stolen for signaling
E1 R2 CAS
An E1 circuit is similar to a T1 circuit: it is a TDM circuit
that carries several DS0s in one connection
The main difference between an E1 and a T1 is that an
E1 bundles 32 time slots instead of 24, resulting in
2.048 Mbps of bandwidth (T1 was 1.544 Mbps)
E1 circuits can be deployed using R2 signaling for CAS
(called an E1 R2 trunk)
These trunks use the E1 multiframe format
In this format, only 30 channels are used for audio
streams. The other two channels are used for framing
and signaling
Voice Over IP: Chapter 1 Introducing Voice Gateways
E1 Multiframe Format
A multiframe consists of 16 consecutive frames, each
carrying 32 time slots
The first time slot is used exclusively for frame
synchronization
Time slots 2 through 16 and 18 through 32 carry the
actual voice traffic
Time slot 17 is used for R2 signaling
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E1 Multiframe Format
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ISDN Services
Integrated Services Digital Network (ISDN) is used to
transmit voice and data over ordinary telephone copper
wires
In contrast to CAS and R2 signaling which provide only
DNIS (called number information), ISDN offers
additional supplementary services like call waiting and
Do Not Disturb (DND)
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http://business.telus.com/enterprise/bc/mlb-business-voice-local/mlb-pri-service
Voice Over IP: Chapter 1 Introducing Voice Gateways
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ISDN Signaling
ISDN uses Q.921 as its Layer 2 signaling protocol and
Q.931 as its Layer 3 signaling protocol
Q.921 (also known as LAPD) is very similar to HDLC
(the default encapsulation type on Cisco serial
interfaces)
Q.931 is used at Layer 3 for call-establishment, calltermination, information, and miscellaneous messaging
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Digital Trunks
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ECHO
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Echo Cancellation
Echo is the sound of your own voice reverberating in
the telephone receiver while you are talking
When timed properly, echo is not a problem in a
conversation
However, if the echo interval exceeds approximately
25ms, it can be distracting to the speaker
In the traditional telephony network, echo is generally
caused by an impedance mismatch when the four-wire
network is converted to the two-wire local loop
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Talker Echo
Talker echo happens with the speech sent by a talker
sent down the transmit path is coupled into the
receiving path
Talkers then hear their own voice, delayed by the total
delay of the path
This is the most common type of echo and is a direct
result of two- to four-wire conversion
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Listener Echo
Listener echo occurs at the far end and is caused by
the echo being echoed.
The voice of the talker is echoed by the receiving end,
and then echoed back again by the transmitting end
The person listening hears both the talker and the echo
of the talker
This is much less common
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Echo Cancellation
An echo canceller is a tool that you can use to control
echo
An echo canceller reduces the level of echo that leaks
from the receive path into the transmit path
Echo cancellation is implemented in the DSP firmware
on Cisco voice gateways and is independent of the
other DSP configurations
Enabled using the command echo-cancel enable in
voice-port configuration mode (enabled by default)
In voice packet-based networks, echo cancellers are
built into the low-bit-rate codecs
Voice Over IP: Chapter 1 Introducing Voice Gateways
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Codecs
A codec is a device or program capable of performing
encoding and decoding on a digital data stream or
signal
In essence, the codec is the method used to convert
the analog signal to a digitized, packetized format, and
back again
Various types of codecs are used to encode and
decode or compress and decompress data that would
otherwise use large amounts of bandwidth on WAN
links
Codecs are especially important on lower-speed serial
links, where every bit of bandwidth is needed
Voice Over IP: Chapter 1 Introducing Voice Gateways
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Codecs
Capacity planning is one of the most important factors
to consider when building a voice network
You must understand how much bandwidth is used for
each VoIP call
To understand that, you must know which codec is
being used
Coding techniques are standardized by the ITU with the
ITU-T G-series codecs being the most popular
standards (G.711, G.729, etc.)
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Codecs
These are the codecs supported by Cisco IOS
gateways:
G.711: The international standard for encoding telephone audio
on a 64-kbps digital channel (such as a DS0 channel on a T1).
It is a PCM scheme operating at an 8-kHz sample rate, with 8
bits per sample.
With G.711, the encoded voice is already in the correct format
for digital voice delivery in the PSTN or through PBXs
There are two subsets of the G.711 codec, -law (pronounced
mu-law) used in North America/Japan, and a-law used
everywhere else
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Codecs
These are the codecs supported by Cisco IOS
gateways:
G.726: Uses a special kind of PCM called Adaptive Differential
PCM. Essentially, rather than sending the value of the current
sample we send the change in this sample from the one before
it .
Available at 40, 32, 24, and 16 kbps variants and often referred
to by the bit size of a sample (5, 4, 3, and 2 bits respectively)
G.728: Uses an algorithm called LDCELP to compress the
voice stream to 16 kbps
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Codecs
These are the codecs supported by Cisco IOS
gateways:
G.729: Uses CS-ACELP voice compression algorithm to code
voice into 8 kbps streams
G.729a (Annex A) requires less computation but speech quality
is marginally worsened
G.729b (Annex B) adds support for VAD and CNG making it
more efficient in its bandwidth usage
G.729ab combines the features of Annex A and Annex B
There are also variants (Annex D and Annex E, and others) that
provide different bit rates
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Codecs
These are the codecs supported by Cisco IOS
gateways:
G.723.1: Comes in two bitrates:
r63: uses 24-byte frames at 6.3 kbps
r53: uses 20-byte frames at 5.3 kbps
The higher bitrate provides a slightly better quality
GSM Full Rate Codec (GSMFR): Operates at 13 kbps. Used
with VoiceXML scripts that can be used for simple voice-mail
systems
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Codecs
These are the codecs supported by Cisco IOS
gateways:
Internet Low Bit Rate Codec (iLBC): Has a payload bit rate of
13.33 kbps or 15.20 kbps. Its a free open source codec, (used
in Google Talk and many other applications).
This codec enables graceful speech quality degradation in the
case of lost frames, which occurs in connection with lost or
delayed IP packets.
In other words, when packets are lost, the speech quality is
much better when using iLBC than other low-bitrate codecs.
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Bytes_per_Sample = 160
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30ms
20ms
30ms
20ms
30ms
20ms
30ms
20ms
30ms
20ms
30ms
20ms
Voice Over IP: Chapter 1 Introducing Voice Gateways
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Bytes_per_Sample = 20 bytes
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Codec Quality
These are the average MOSs for most typical codecs
(under ideal network conditions)
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Evaluating Overhead
Several factors must be included in calculating the
overhead of a VoIP call.
Layer 2, Layer 3, and security protocols significantly
add to the packet size
BW_per_call = (Voice_payload + L3/4_overhead + L2_overhead) *
Packet_ratio) * 8 bits/byte
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VPN Overhead
VPN encapsulation adds additional overhead to the
VoIP packets
Encapsulating Security Payload (ESP): Adds typically a 50to 57-byte overhead (depending on the encryption and
authentication algorithms used)
Generic Routing Encapsulation (GRE), Layer 2 Tunneling
Protocol (L2TP): Adds a 24-byte header
Multiprotocol Label Switching (MPLS): Adds a 4-byte header
for every label carried in the packet. A label stack might include
multiple labels in an MPLS VPN or traffic engineering
environment
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G.711 has 160 bytes of payload per packet (found from the
previous formula)
Layer 3/4 is 40 bytes under normal circumstances
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DSP Chip
The DSP chip comes in several form factors, the
modular packet voice DSP module (PVDM) being the
most common
The PVDM can have multiple DSPs on the module
Currently, there are two major types of high-density
PVDMs: PVDM generation 2 (PVDM2) and PVDM
generation 3 (PVDM3)
The 2800/3800 routers only support PVDM2 chips
The newer 2900/3900 routers support both types, but
not in certain combinations
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Codec Complexity
Codec complexity refers to the amount of processing
that is required to perform voice compression
Codec complexity affects call density, which is the
number of calls that are able to be processed at once
With higher codec complexity, fewer calls can be
processed
Codecs that perform a lot of compression typically have
higher complexity
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Codec Complexity
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DSP Calculator
For easier DSP calculation, a DSP calculator tool is
available at the following URL (cisco.com login
required):
http://cisco-apps.cisco.com/web/applicat/dsprecal/dsp_calc.html
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