ZyXEL Communications IES-1248-51V Handbuch

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

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

IES-1248-51V Support Notes

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

IES-1248-51V Support Notes

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

IES-1248-51V Support Notes

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.

IES-1248-51V Support Notes

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.

IES-1248-51V Support Notes

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

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-

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

IES-1248-51V Support Notes

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.

IES-1248-51V Support Notes

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

nalo

hone A

25K ~ 2.2M Hz

0 ~ 4K Hz

Copper line Splitter

Data packets

Upper Network

Soft Switch

PSTN

Managed Net

nalo

hone B

Internet VoIP packets

Data packets

Analog voice

STB

IES-1248-51V Support Notes

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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