ZyXEL Communications IES-1248-51V Manual

IES-1248-51V
48-port ADSL2+ remote IP DSLAM with embedded Media
Gateway
Support Notes
Version 3.53
November 2007

IES-1248-51V Support Notes
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Index
Application Notes............................................................................................................................ 3
How to configure the Voice over IP (VoIP) features...................................................3
What is VoIP ..................................................................................................................... 3
What is SIP ....................................................................................................................... 3
Parameters specially for VoIP service............................................................................... 5
How to set up a VoIP environment ................................................................................... 8
Provide Different DSL Port Speeds to different subscribers..................................... 24
How to apply the profile to ports.................................................................................... 24
How to configure 802.1Q VLAN...............................................................................30
How to set up a VLAN environment.............................................................................. 31
Triple play Application.............................................................................................. 34
How to set up a Multiple PVCs environment................................................................. 35
802.1x Application....................................................................................................43
How to set up an 802.1x environment............................................................................ 43
Setting up the Syslog Server ...................................................................................48
How to set up a Syslog server......................................................................................... 48
Setting up the Ring Environment .............................................................................50
How to set up a Ring Environment................................................................................. 51
Setting up the IGMP Snooping/IGMP Filtering.........................................................57
How to set up IGMP snooping/IGMP filtering............................................................... 58
Static Multicast.........................................................................................................60
How to configure Static Multicast filter ......................................................................... 62
Limit the users behind the certain DSL port............................................................. 66
How to set up MAC Filter/Port Security........................................................................ 67
DHCP Relay Option 82 Application..........................................................................69
How to set up DHCP Relay Option 82 Environment ..................................................... 69
Filter Some Certain Packet...................................................................................... 79
How to Filter Some Specific Packet............................................................................... 79
Cascade two IES-1248-51V (Daisy chain mode)..................................................... 81
How to setup Daisy-chain mode..................................................................................... 81
FAQ ................................................................................................................................................ 82
What is the IES-1248-51V? .....................................................................................82
What is the maximum line rate of IES-1248-51V? ................................................... 82
How many sessions are allowed to access IES-1248-51V at the same time?......... 82
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What is the power requirements of IES- 1248-51V?................................................ 82
What kind of the ADSL standards does IES-1248-51V comply?.............................. 82
What kind of the Multicasting functionalities does IES-1248-51V support? .............83
What kind of the Media Gateway Signaling protocols are supported?..................... 83
What kind of the Media Gateway Communication protocols are supported?...........83
What are the limitations for eachADSL port of IES-1248-51V?............................... 84
What are the system limitations of IES-1248-51V?..................................................84
What are the VoIP limitations of IES-1248-51V?......................................................85
What kind of the voice CODECs does the IES-1248-51V support?.........................85
What is the ADSL chipset of IES-1248-51V? ........................................................... 85
How many VLAN ID and Static VLAN entry does IES-1248-51V support?.............. 86
What does it mean if the ALM LED lights on?..........................................................86
What is the default console rate of IES-1248-51V? ................................................. 86
How to upgrade the firmware of IES-1248-51V? .....................................................86
Appendix A – Installation of SER................................................................................................ 87
RPM Package Installation........................................................................................ 87
Real-time monitor..................................................................................................... 88
mySQL Setup...........................................................................................................89
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Application Notes
How to configure the Voice over IP (VoIP) features
The IES-1248-51V is an IP-based DSLAM that connects ADSL and voice subscribers
to the Internet. It provides 48 aggregated lines of POTS connectivity, designed to
connect the subscriber with the PSTN. Each telephone line interface is a FXS port
connecting to the subscriber’s telephone via copper wire. The analog voice signal
from the subscriber is converted to voice data packets and transmitted towards the
callee across the IP packet-switched network. The IES-1248-51V uses SIP for
network signaling to establish or tear down a voice call. DTMF signals are also
translated into SIP signals (or transmitted in the voice band).
What is VoIP
Traditional voice switching business has long been the mainstay for a relatively small
group of global equipment suppliers. But times has changes, carriers are adopting a
standardized IP Multimedia Subsystem architecture for next-generation networks with
distributed softswitch call control and independent Media Gateways. In addition to the
legacy circuit switch equipment suppliers, the equipment market reshuffles and opens
to new vendors.
Media Gateway is a network element which packetize voice traffic from POTS signals
to IP packets and perform critical voice network features such as echo cancellation,
silence detection and suppression, etc. Softswitch is the one which provides call
control to Media Gateways in a packet voice network.
Through IES-1248-51V, ZyXEL transforms its IP DSLAM portfolio into multi-service
DSLAM and starts to enter the Media Gateway market worldwide.
VoIP is the sending of voice signals over the Internet Protocol. This allows you to
make phone calls and send faxes over the Internet at a fraction of the cost of using
the traditional circuit-switched telephone network.
What is SIP
The Session Initiation Protocol (SIP: RFC 3261) is an ASCII-based, peer-to-peer
protocol designed to provide rendezvous services over the Internet. SIP is an Internet
Engineering Task Force (IETF) specification that was derived from Hypertext Transfer
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Protocol (HTTP: RFC 2616) and Simple Mail Transfer Protocol (SMTP: RFC 821).
SIP, along with Media Gateway Control Protocol (MGCP: RFC 2705), and H.323 (an
International Telecommunications Union (ITU) specification), is one of three
commonly used open protocols for VoIP implementations.
SIP User Agent:
User Agents (UA’s) are specified in RFC 3261 as applications such as SIP phones
and software that initiate and receive calls over a SIP network.
SIP Server:
Servers are specified in RFC 3261 as application programs that accept requests,
service requests and send back responses to those requests.
SIP Call Flow:
The following figure describe the SIP messages exchanged for call establishment and
tear down.
Step Description
1 Invite: User A initiates a call to user B. This message is an invitation for B
to participate in a SIP telephone call.
2 Ringing: User B sends a response to user A and indicates that the phone is
ringing.
3 OK: B sends a OK response after the call is answered.
4 ACK: User A then sends an ACK message to acknowledge that B has
answered the call.
5 Voice: a two-way voice channel is established over real-time protocol (RTP)
and a conversation takes place between user A and B.
6 Bye: User A hangs up.
7 OK: User B replied a OK message confirming receipt of the Bye request
and the call is terminated.
For more information about SIP messages, please see RFC 3261.
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Parameters specially for VoIP service
The IES-1248-51V is different form other xDSL series pizza-box. Following table
display some novel attributes in IES-1248-51V only. Operators should be careful with
these parameters while deploying VoIP.
Item Description
IPAddress A unique IP address for IES-1248-51V
DNS Specify the IP address of the Domain Name System server for VoIP
service. You do not need to enter this if your SIP server uses IP
addresses in SIP messages (not domain names).
Default VoIP
Gateway
Define the default outgoing gateway for VoIP service.
SIP profile 1. Setup the SIP IP / Domain Name of sip server, registrar IP and
proxy server IP. The default port number is 5060 but can be
customized in both SIP server and the IES-1248-51V.
2. “URI Type” can be “sip” or “tel”, it’s used in the signaling packet
header and SIP server must recognize the URI format when read the
packets. Otherwise the requests will be discarded if SIP server
cannot support. Default “URI Type” is SIP.
3. “802.1p Priority” (0~7) and “DSCP” (Differentiated Service Code
Point, 0~63) are used to define the QoS in VLAN tag and IP header
respectively.
4. “Keep Alive” is defined in RFC4028 as a keep-alive mechanism
for SIP sessions. UAs send periodic re-INVITE or UPDATE
requests to keep the session alive. In “Session Expiration”, you can
enter the minimum number of seconds after which the IES tears
down the session (if no successful session refresh has occurred).
5. “Provisional Response ACK” is used to set whether the IES sends
provisional acknowledgment messages (ON), or does not send them
(OFF).
Call service profile This setting allows operator to define the authentication password
and what call features to be active/inactive.
1. “Name” is the name of this call service profile. If you select
“ON” in the Password for SIP Registration field, the name is also
used for user authentication.
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2. “Password” is used in SIP authentication while registering. If the
SIP account this profile uses does not require a password for user
authentication, you do not need to configure it.
3. “Do not Disturb” rejects all incoming calls via automatically
reply “486 Busy” message to caller.
4. “Call Hold” service allows user to accept another call on waiting
or transfer to another remote party. Typically, user needs to press
flash button to put remote party on hold or resume conversation. To
enable “Call Wait” and “Call Transfer”, this feature must be active
in advance.
5. “Call Wait” allows the indication of third-party incoming call. I
f
turn on, the subscriber will be notified with call waiting tone that an
incoming call is waiting, instead of replying “Busy” directly. When
a call awaits, user should decide to (1)reject it, (2)accept it and
terminate original conversation, (3)accept it and hold original
dialogue.
6. “Call Transfer” is a mechanism to transfer current correspondent
to another remote user. After transferring, the two parties can chat.
7. DTMF (digital tone multi frequency) Relay is voice band signal
normally used to identify which dial button is pressed on the phone.
Now IES-1248-51V supports four types of DTMF processing. User
can configure to bypass DTMF signal as in-
b
and voice, relay it as
RFC2833/RFC2833 like (Cisco compatible) RTP packets, or relay it
by SIP INFO messages.
8. Two types of FAX services – bypass FAX signal with G.711
CODEC or relay according to ITU-T T.38. User can configure
which type of FAX service to be used or disable service.
DSP profile In this configuration operator can define what CODEC to use, how
much the play buffer and echo cancellation period.
1. Currently the available CODECs are G.711a, G.711mu, G.723,
G.726-16, G.726-24, G.726-32, G.726-40, G.729ab. Besides, both
caller and callee need to negotiate a consistent CODEC to be
applied in their voice conversation, thus the order of CODECs mean
the priority of selection.
2. “Echo Tail” provides the echo cancellation period (i.e. 8, 16, 32
and 128 ms) to decrease the acoustic echo arising in circuit or
handset.
3. “Play Buffer Delay” is designed to store the received steaming
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and play the audio, which can reduce the jitter. “Min Play Buffer
Delay” and “Max Play Buffer Delay” are the threshold of that buffer
(10~500 ms, and max one ≧mix.).
Number plan table The telephone number is composed of three parts, country code,
national destination code, and subscriber number, which is defined
in ITU-T E.164. Generally user can omit to dial country code and
national destination since local exchange should transfer the dialed
number as E.164 format before transmitting.
Number plan consists of numbers of entries. Each entry has two
fields, first is number pattern and second one is translating rule. The
entry order plane is significant. The numbering plan feature
compares the dialed number with the defined pattern one by one.
Once matched, the dial number is translating into E.164 number
according to the corresponding rules.
Country code Some VoIP parameters/settings have different values according to
which country operator deploy the IES-1248-51V. To ease the
complexity of configuration, we build a list of packages for choice
in advance.
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How to set up a VoIP environment
Below is an example of using IES-1248-51V to provide VoIP services. The
IES-1248-51V is installed in the operators’ estate. The voice signal between the
access node and the customer site is still analog, the same as POTS.
Configurations
Soft Switch
SIP Server IPAddress: 192.168.10.1, Port: 5060
Registrar Server IPAddress: 192.168.10.1, Port: 5060
Proxy Server IPAddress: 192.168.10.1, Port: 5060
Authentication:
User A (for Analog phone A)
SIP Number: 111
Password: 111
User B (for Analog phone B)
SIP Number: 112
Password: 112
IES-1248-51V
VoIP Service IPAddress: 192.168.10.11/24
VLAN ID for VoIP Service: 2
802.1p Priority for VoIP Service: 7
Analog phoneA is connected to port 1
Analog phone B is connected to port 2
The soft switch is connected to ENET2
IES-1248-51V
A
nalo
g
p
hone A
PC
25K ~ 2.2M Hz
0 ~ 4K Hz
Copper line Splitter
X
Data packets
Upper Network
Soft Switch
PSTN
Managed Net
A
nalo
g
p
hone B
Internet VoIP packets
Data packets
Analog voice
STB
TV
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We will use the built-in web configurator and take the following steps to configure the
IES-1248-51V. It will allow User A and User B to communicate with each other via
VoIP service.
1) System login.
2) Set the IES-1248-51V’s VoIP service IP address to 192.168.10.11.
3) Configure 802.1Q VLAN for VoIP service.
4) Create a SIP profile.
5) Create a Call Service profile.
6) Configure the VoIP settings of the ports.
7) Save your configuration.
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Configure the IES-1248-51V
1) System login.
Connect to the IES-1248-51V via Internet browser. The default IP address of
IES-1248-51V is 192.168.1.1. Enter the default username and password to access
the device:
Username: admin
Password: 1234
The main screen is the first screen that displays when you access the web
configurator.
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2) Set the IES-1248-51V’s VoIP service IP address to 192.168.10.11.
Please take the following steps to set up the IES-1248-51V’s VoIP IP configuration.
a. In the navigation panel, click Basic Setting > IP Setup. The IP Setup screen
appears.
b. In the VoIP field, for IP Address: enter 192.168.10.11 and for Default VoIP
Gateway: enter 192.168.10.1.
c. Click the Apply button.
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3) Configure 802.1Q VLAN for VoIP service.
Do the following to assign ENET2 to be a member of the VLAN 2 group.
a. Click Advanced Application > VLAN on the navigation panel. The VLAN Status
screen appears. Click the Static VLAN Settings tab.
b. Click VID 2’s index number.
c. Click to select the Active check box.
d. In the ENET2 field, select Fixed and clear the Tx Tagging check box.
e. Click the Add button.
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f. Click the VLAN Port Setting tab.
g. Type 2in the PVID field of ENET2.
h. Select 7from the Priority drop-down list box of ENET2. This will assign the
highest 802.1p priority to the untagged VoIP frames received on ENET2.
i. Click Apply.
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4) Create a SIP profile.
Take the following steps to create a SIP profile for the soft switch.
a. Click VoIP > SIP Profile on the navigation panel. The SIP Profile screen appears.
b. Type a name for the SIP profile. For our example, it is sipserver.
c. Type 192.168.10.1 in the SIP IP / Doman Name field and 5060 in the Port field.
d. Type 192.168.10.1 in the Registrar IP / Domain Name field and 5060 in the Port
field.
e. Type 192.168.10.1 in the Proxy Server IP / Domain Name field and 5060 in the
Port field.
f. Select 7from the 802.1p Priority drop-down list box.
g. Click the Add button to create the profile.
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5) Create a Call Service profile.
Please do the following to create Call Service profiles for User A and User B
respectively.
a. Click VoIP > Call Service Profile on the navigation panel.
b. Type a name for the profile. For our example, it is 111.
c. Type 111 in the Password field, and retype it in the Retype Password to
Confirm field.
d. Click the Add button to create the 111 profile for User A.
e. Type 112 in the Name field.
f. Type 112 in the Password field, and retype it in the Retype Password to
Confirm field.
g. Click the Add button to create the 112 profile for User B.
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6) Configure the VoIP settings of the ports.
Click VoIP > VoIP Port Setup on the navigation panel. The Port View screen
appears. Click port 1’s index number.
a. Click to select the Enable check box.
b. Enter 111 in the VoIP Tel Number field.
c. Select sipserver from the SIP Profile dorp-down list box.
d. Select 111 from the Call Service Profile drop-down list box.
e. Click Apply.
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f. Select 2from the Port drop-down list box.
g. Click to select the Enable check box.
h. Enter 112 in the VoIP Tel Number field.
i. Select sipserver from the SIP Profile dorp-down list box.
j. Select 112 from the Call Service Profile drop-down list box.
k. Click Apply.
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You can see details of the VoIP settings configured on the two ports in the Port View
screen.
All contents copyright (c) 2007 ZyXEL Communications Corporation.