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Page 256 / 412 Scroll up to view Page 251 - 255
Chapter 21 Voice
VMG8924-B10A and VMG8924-B30A Series User’s Guide
256
SIP Call Progression
The following figure displays the basic steps in the setup and tear down of a SIP call. A calls B.
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.
SIP Call Progression Through Proxy Servers
Usually, the SIP UAC sets up a phone call by sending a request to the SIP proxy server. Then, the
proxy server looks up the destination to which the call should be forwarded (according to the URI
requested by the SIP UAC). The request may be forwarded to more than one proxy server before
arriving at its destination.
The response to the request goes to all the proxy servers through which the request passed, in
reverse sequence. Once the session is set up, session traffic is sent between the UAs directly,
bypassing all the proxy servers in between.
Table 119
SIP Call Progression
A
B
1. INVITE
2. Ringing
3. OK
4. ACK
5.Dialogue (voice traffic)
6. BYE
7. OK
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Chapter 21 Voice
VMG8924-B10A and VMG8924-B30A Series User’s Guide
257
The following figure shows the SIP and session traffic flow between the user agents (
UA 1
and
UA
2
) and the proxy servers (this example shows two proxy servers,
PROXY 1
and
PROXY 2
).
Figure 154
SIP Call Through Proxy Servers
The following table shows the SIP call progression.
1
User Agent 1
sends a SIP INVITE request to
Proxy 1
. This message is an invitation to
User
Agent 2
to participate in a SIP telephone call.
Proxy 1
sends a response indicating that it is trying
to complete the request.
2
Proxy 1
sends a SIP INVITE request to
Proxy 2
.
Proxy 2
sends a response indicating that it is
trying to complete the request.
3
Proxy 2
sends a SIP INVITE request to
User Agent 2
.
4
User Agent 2
sends a response back to
Proxy 2
indicating that the phone is ringing. The response
is relayed back to
User Agent 1
via
Proxy 1
.
Table 120
SIP Call Progression
UA 1
PROXY 1
PROXY 2
UA 2
Invite
Invite
100 Trying
Invite
100 Trying
180 Ringing
180 Ringing
180 Ringing
200 OK
200 OK
200 OK
ACK
RTP
RTP
BYE
200 OK
UA 1
UA 2
PROXY 1
PROXY 2
SIP
SIP
SIP
SIP
&
RTP
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Chapter 21 Voice
VMG8924-B10A and VMG8924-B30A Series User’s Guide
258
5
User Agent 2
sends an OK response to
Proxy 2
after the call is answered. This is also relayed
back to
User Agent 1
via
Proxy 1
.
6
User Agent 1
and
User Agent 2
exchange RTP packets containing voice data directly, without
involving the proxies.
7
When
User Agent 2
hangs up, he sends a BYE request.
8
User Agent 1
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 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.
Voice Activity Detection/Silence Suppression
Voice Activity Detection (VAD) detects whether or not speech is present. This lets the Device reduce
the bandwidth that a call uses by not transmitting “silent packets” when you are not speaking.
Comfort Noise Generation
When using VAD, the Device generates comfort noise when the other party is not speaking. The
comfort noise lets you know that the line is still connected as total silence could easily be mistaken
for a lost connection.
Echo Cancellation
G.168 is an ITU-T standard for eliminating the echo caused by the sound of your voice
reverberating in the telephone receiver while you talk.
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.
Page 259 / 412
Chapter 21 Voice
VMG8924-B10A and VMG8924-B30A Series User’s Guide
259
Custom Tones (IVR)
IVR (Interactive Voice Response) is a feature that allows you to use your telephone to interact with
the Device. The Device allows you to record custom tones for the
Early Media
and
Music On Hold
functions. The same recordings apply to both the caller ringing and on hold tones.
Recording Custom Tones
Use the following steps if you would like to create new tones or change your tones:
1
Pick up the phone and press “****” on your phone’s keypad and wait for the message that says
you are in the configuration menu.
2
Press a number from 1101~1105 on your phone followed by the “#” key.
3
Play your desired music or voice recording into the receiver’s mouthpiece. Press the “#” key.
4
You can continue to add, listen to, or delete tones, or you can hang up the receiver when you are
done.
Listening to Custom Tones
Do the following to listen to a custom tone:
1
Pick up the phone and press “****” on your phone’s keypad and wait for the message that says
you are in the configuration menu.
2
Press a number from 1201~1208 followed by the “#” key to listen to the tone.
3
You can continue to add, listen to, or delete tones, or you can hang up the receiver when you are
done.
Deleting Custom Tones
Do the following to delete a custom tone:
1
Pick up the phone and press “****” on your phone’s keypad and wait for the message that says
you are in the configuration menu.
2
Press a number from 1301~1308 followed by the “#” key to delete the tone of your choice. Press
14 followed by the “#” key if you wish to clear all your custom tones.
Table 121
Custom Tones Details
LABEL
DESCRIPTION
Total Time for All Tones
900 seconds for all custom tones combined
Maximum Time per
Individual Tone
180 seconds
Total Number of Tones
Recordable
5
You can record up to 5 different custom tones but the total time must be 900
seconds or less.
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Chapter 21 Voice
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You can continue to add, listen to, or delete tones, or you can hang up the receiver when you are
done.
21.10.1
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 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.
3
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 155
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.
21.10.2
Phone Services Overview
Supplementary services such as call hold, call waiting, and call transfer. are generally available from
your VoIP service provider. The Device supports the following services:
3.
The Device does not support DiffServ at the time of writing.
DSCP
(6-bit)
Unused
(2-bit)

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