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Chapter 16 VoIP
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251
3
Client device
A
then sends the call invitation to client device
C
.
Figure 106
SIP Redirect Server
SIP Register Server
A SIP register server maintains a database of SIP identity-to-IP address (or
domain name) mapping. The register server checks your user name and password
when you register.
RTP
When you make a VoIP call using SIP, the RTP (Real time Transport Protocol) is
used to handle voice data transfer. See RFC 3550 for details on RTP.
Pulse Code Modulation
Pulse Code Modulation (PCM) measures analog signal amplitudes at regular time
intervals and converts them into bits.
SIP Call Progression
The following figure displays the basic steps in the setup and tear down of a SIP
call. A calls B.
Table 65
SIP Call Progression
A
B
1. INVITE
2. Ringing
3. OK
4. ACK
1
2
3
A
B
C
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252
1
A
sends a SIP INVITE request to
B
. This message is an invitation for
B
to
participate in a SIP telephone call.
2
B
sends a response indicating that the telephone is ringing.
3
B
sends an OK response after the call is answered.
4
A
then sends an ACK message to acknowledge that
B
has answered the call.
5
Now
A
and
B
exchange voice media (talk).
6
After talking,
A
hangs up and sends a BYE request.
7
B
replies with an OK response confirming receipt of the BYE request and the call is
terminated.
Voice Coding
A codec (coder/decoder) codes analog voice signals into digital signals and
decodes the digital signals back into analog voice signals. The ZyXEL Device
supports the following codecs.
•
G.711 is a Pulse Code Modulation (PCM) waveform codec. PCM measures analog
signal amplitudes at regular time intervals and converts them into digital
samples. G.711 provides very good sound quality but requires 64 kbps of
bandwidth.
•
G.726 is an Adaptive Differential PCM (ADPCM) waveform codec that uses a
lower bitrate than standard PCM conversion. ADPCM converts analog audio into
digital signals based on the difference between each audio sample and a
prediction based on previous samples. The more similar the audio sample is to
the prediction, the less space needed to describe it. G.726 operates at 16, 24,
32 or 40 kbps.
•
G.729 is an Analysis-by-Synthesis (AbS) hybrid waveform codec that uses a
filter based on information about how the human vocal tract produces sounds.
G.729 provides good sound quality and reduces the required bandwidth to 8
kbps.
5.Dialogue (voice
traffic)
6. BYE
7. OK
Table 65
SIP Call Progression (continued)
A
B
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PSTN Call Setup Signaling
Dual-Tone MultiFrequency (DTMF) signaling uses pairs of frequencies (one lower
frequency and one higher frequency) to set up calls. It is also known as Touch
Tone®. Each of the keys on a DTMF telephone corresponds to a different pair of
frequencies.
Pulse dialing sends a series of clicks to the local phone office in order to dial
numbers.
3
MWI (Message Waiting Indication)
Enable Message Waiting Indication (MWI) enables your phone to give you a
message–waiting (beeping) dial tone when you have a voice message(s). Your
VoIP service provider must have a messaging system that sends message waiting
status SIP packets as defined in RFC 3842.
16.9.3
Quality of Service (QoS)
Quality of Service (QoS) refers to both a network's ability to deliver data with
minimum delay, and the networking methods used to provide bandwidth for real-
time multimedia applications.
Type of Service (ToS)
Network traffic can be classified by setting the ToS (Type of Service) values at the
data source (for example, at the ZyXEL Device) so a server can decide the best
method of delivery, that is the least cost, fastest route and so on.
DiffServ
DiffServ is a class of service (CoS) model that marks packets so that they receive
specific per-hop treatment at DiffServ-compliant network devices along the route
based on the application types and traffic flow. Packets are marked with DiffServ
Code Points (DSCP) indicating the level of service desired. This allows the
intermediary DiffServ-compliant network devices to handle the packets differently
depending on the code points without the need to negotiate paths or remember
state information for every flow. In addition, applications do not have to request a
particular service or give advanced notice of where the traffic is going.
4
3.
The ZyXEL Device does not support pulse dialing at the time of writing.
4.
The ZyXEL Device does not support DiffServ at the time of writing.
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DSCP and Per-Hop Behavior
DiffServ defines a new DS (Differentiated Services) field to replace the Type of
Service (TOS) field in the IP header. The DS field contains a 2-bit unused field and
a 6-bit DSCP field which can define up to 64 service levels. The following figure
illustrates the DS field.
DSCP is backward compatible with the three precedence bits in the ToS octet so
that non-DiffServ compliant, ToS-enabled network device will not conflict with the
DSCP mapping.
Figure 107
DiffServ: Differentiated Service Field
The DSCP value determines the forwarding behavior, the PHB (Per-Hop Behavior),
that each packet gets across the DiffServ network.
Based on the marking rule,
different kinds of traffic can be marked for different priorities of forwarding.
Resources can then be allocated according to the DSCP values and the configured
policies.
VLAN Tagging
Virtual Local Area Network (VLAN) allows a physical network to be partitioned into
multiple logical networks. Only stations within the same group can communicate
with each other.
Your ZyXEL Device can add IEEE 802.1Q VLAN ID tags to voice frames that it
sends to the network. This allows the ZyXEL Device to communicate with a SIP
server that is a member of the same VLAN group. Some ISPs use the VLAN tag to
identify voice traffic and give it priority over other traffic.
16.9.4
Phone Services Overview
Supplementary services such as call hold, call waiting, and call transfer. are
generally available from your VoIP service provider. The ZyXEL Device supports
the following services:
•
Call Hold
•
Call Waiting
•
Making a Second Call
•
Call Transfer
•
Three-Way Conference
DSCP
(6-bit)
Unused
(2-bit)
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255
•
Internal Calls
•
Do not Disturb
Note: To take full advantage of the supplementary phone services available through
the ZyXEL Device's phone ports, you may need to subscribe to the services
from your VoIP service provider.
The Flash Key
Flashing means to press the hook for a short period of time (a few hundred
milliseconds) before releasing it. On newer telephones, there should be a "flash"
key (button) that generates the signal electronically. If the flash key is not
available, you can tap (press and immediately release) the hook by hand to
achieve the same effect. However, using the flash key is preferred since the timing
is much more precise. With manual tapping, if the duration is too long, it may be
interpreted as hanging up by the ZyXEL Device.
You can invoke all the supplementary services by using the flash key.
Europe Type Supplementary Phone Services
This section describes how to use supplementary phone services with the
Europe
Type
Call Service Mode
. Commands for supplementary services are listed in the
table below.
After pressing the flash key, if you do not issue the sub-command before the
default sub-command time-out (2 seconds) expires or issue an invalid sub-
command, the current operation will be aborted.
Table 66
European Flash Key Commands
COMMAND
SUB-
COMMAND
DESCRIPTION
Flash
Put a current call on hold to place a second call.
Switch back to the call (if there is no second call).
Flash
0
Drop the call presently on hold or reject an incoming call
which is waiting for answer.
Flash
1
Disconnect the current phone connection and answer the
incoming call or resume with caller presently on hold.
Flash
2
1. Switch back and forth between two calls.
2. Put a current call on hold to answer an incoming call.
3. Separate the current three-way conference call into
two individual calls (one is on-line, the other is on hold).
Flash
3
Create three-way conference connection.
Flash
*98#
Transfer the call to another phone.

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