VOP Support Notes. 24-port POTS/VOIP module for IES Version V3.53(BBT.0) July 2008 Edition 1

Transcription

1 VOP port POTS/VOIP module for IES-1000 Support Notes Version V3.53(BBT.0) July 2008 Edition 1

2 Index INTRODUCTION... 1 WHAT IS VOIP... 1 WHAT IS SIP... 1 SIP User Agent:... 1 SIP Server:... 1 SIP Call Flow:... 2 WHAT IS VOP PARAMETERS SPECIALLY FOR VOIP SERVICE... 4 AN EXAMPLE OF VOP CONFIGURATION... 8 INTRODUCTION... 8 CONFIGURATION STEPS... 8 CONFIGURATION STEPS... 9 CONFIGURE THE VOP ) System login ) Set the VOP s VoIP service IP address to ) Configure 802.1Q VLAN for VoIP service ) Create a SIP profile ) Create a Call Service profile ) Configure the VoIP settings of the ports ) Save your configuration ) VoIP Line Status and Info screen APPENDIX: HOW TO SIGN UP FOR A FWD SIP NUMBER... 22

3 Introduction 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 packetizes voice traffic from POTS signals to IP packets and performs 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 VOP , ZyXEL transforms its IES-1000 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 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. 1

4 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 What is VOP The VOP is a Voice over IP (VoIP) module for IES The VOP VoIP module is perfect for phone service via VoIP to subscribers while minimizing costs. One VOP provides VoIP telephony service for up to 24 subscribers over existing POTS telephone wiring. The line to and from the user carries either voice alone, or both voice and data (DSL) signals. When used in conjunction with the AAM1212 ADSL module, the AAM1212's internal splitter separates the high frequency DSL signal from the voice band signal and feeds the voice band signal to the VOP via the AAM1212's CO port. 2

5 The VOP supports SIP (the Session Initiation Protocol). Each VOP is an individual SIP UA (User Agent) for each physical port. It has an IP address for SIP and RTP protocols to communicate with SIP proxy servers. 3

6 Parameters specially for VoIP service The VOP is different form other xdsl series modules. Following table displays some novel attributes in VOP only. Operators should be careful with these parameters while deploying VoIP. Item IP Address DNS Default VoIP Gateway SIP profile Call service profile Description A unique IP address for VOP 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). Define the default outgoing gateway for VoIP service. 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 VOP 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 p 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). 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. 2. Password is used in SIP authentication while registering. If the 4

7 DSP profile 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. If 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 VOP supports four types of DTMF processing. User can configure to bypass DTMF signal as in-band 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. 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 , G , G , G , 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 and play the audio, which can reduce the jitter. Min Play Buffer 5

8 Number plan table Delay and Max Play Buffer Delay are the threshold of that buffer (10~500 ms, and max one mix.). 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. Countrycode Some VoIP parameters/settings have different values according to which country operator deploy the VOP To ease the complexity of configuration, we build a list of packages for choice in advance. The following table lists the accepted COUNTRY and COUNTRYCODE values. COUNTRY COUNTRYCODE USA 0 JAPAN 1 TAIWAN 2 AUSTRIA 3 BELGIUM 4 BULGARIA 5 CZECH 6 DENMARK 7 FINLAND 8 FRANCE 9 HUNGARY 10 ICELAND 11 ITALY 12 LUXEMBOURG 13 NETHERLANDS 14 6

9 NORWAY 15 POLAND 16 PORTUGAL 17 SLOVAKIA 18 SPAIN 19 SWEDEN 20 SWITZERLAND 21 UK 22 GERMANY 23 GREECE 24 AUSTRALIA 25 NEW_ZEALAND 26 HONGKONG 27 SINGAPORE 28 MOROCCO 29 IRELAND 30 MALAYSIA 31 RUSSIA 32 THAILAND 33 ISRAEL 34 UAE 35 CHINA 36 UKRAINE 37 SOUTH_AFRICA 38 SOUTH_KOREA 39 PHILIPPINES 40 INDIA 41 TURKEY 42 VIETNAM 43 BRAZIL 44 7

10 An example of VOP configuration Introduction Below is an example of using VOP to provide VoIP services. The VOP is installed in the operators estate. The voice signal between the access node and the customer site is still analog, the same as POTS. FWD SIP Server [PORT1] Splitter IP Core Network IES-1000 (w/ VOP & AAM installed) [PORT2] Analog Phone (B) Analog Phone (A) 8

11 Configuration Steps Configurations Soft Switch SIP Server IP Address: fwd.pulver.com, Port: 5060 Registrar Server IP Address: fwd.pulver.com, Port: 5060 Proxy Server IP Address: fwd.pulver.com, Port: 5060 Authentication: User A (for Analog phone A) SIP Number: Password: User B (for Analog phone B) SIP Number: Password: VOP VoIP Service IP Address: /27 Default VoIP Gateway: DNS: VLAN ID for VoIP Service: p Priority for VoIP Service: 7 Analog phone A is connected to port 1 Analog phone B is connected to port 2 The soft switch is connected to ENET1 We will use the built-in web configurator and take the following steps to configure the VOP It will allow User A and User B to communicate with each other via VoIP service. a. System login. b. Set the VOP s VoIP service IP address to /27. c. Configure 802.1Q VLAN for VoIP service. d. Create a SIP profile. e. Create a Call Service profile. f. Configure the VoIP settings of the ports. g. Save your configuration. 9

12 Configure the VOP ) System login. Connect to the VOP via Internet browser. The default IP address of VOP is 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. 10

13 2) Set the VOP s VoIP service IP address to VOP Support Notes Please take the following steps to set up the VOP 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 and for Default VoIP Gateway: enter , for DNS: c. Click the Apply button. 11

14 3) Configure 802.1Q VLAN for VoIP service. Do the following to assign ENET1 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 ENET1 field, select Fixed and clear the Tx Tagging check box. e. Click the Apply button. f. Click the VLAN Port Setting tab. g. Type 2 in the PVID field of ENET1. h. Select 7 from the Priority drop-down list box of ENET1. This will assign the highest 12

15 802.1p priority to the untagged VoIP frames received on ENET1. i. Click Apply. 13

16 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 fwd. c. Type fwd.pulver.com in the SIP IP / Doman Name field and 5060 in the Port field. d. Type fwd.pulver.com in the Registrar IP / Domain Name field and 5060 in the Port field. e. Type fwd.pulver.com in the Proxy Server IP / Domain Name field and 5060 in the Port field. f. Select 7 from the 802.1p Priority drop-down list box. g. Click the Add button to create the profile. 14

17 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 c. Type in the Password field, and retype it in the Retype Password to Confirm field. d. Click the Add button to create the profile for User A. e. Type in the Name field. f. Type in the Password field, and retype it in the Retype Password to Confirm field. g. Click the Add button to create the profile for User B. 15

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19 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 in the VoIP Tel Number field. c. Select fwd from the SIP Profile dorp-down list box. d. Select from the Call Service Profile drop-down list box. e. Click Apply. f. Select 2 from the Port drop-down list box. g. Click to select the Enable check box. h. Enter in the VoIP Tel Number field. i. Select fwd from the SIP Profile dorp-down list box. 17

20 j. Select from the Call Service Profile drop-down list box. k. Click Apply. You can see details of the VoIP settings configured on the two ports in the Port View screen. 18

21 7) Save your configuration. Click Config Save > Config Save on the navigation panel, and then click the Save button to save your configuration to nonvolatile memory. 19

22 8) VoIP Line Status and Info screen. After configuring all the ports, you can use this screen to see detailed information about the VoIP configuration currently active on the ports. Click VoIP > VoIP Line Status and Info on the navigation panel. The following screen appears. Now Service Status of port 1 is Idle. Some information of RTP and SIP is only shown when communication is established. For the following example, port 1 is the caller, and port 2 is the callee. Their communication is established. This screen shows port 1 s line status and information. The following screen shows port 2 s line status and information. 20

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24 Appendix: How to sign up for a FWD SIP number FWD (Free World Dialup) is a non-commercial VoIP network company. The service allows users to make free SIP telephone calls to other FWD users around the world. You can follow the step-by-step instructions below to sign up for a FWD SIP number. 1. Go to FWD s official site ( 2. Click the my.fwd tab. 22

25 3. Click << Sign Up for Fwd >>. 4. Follow the on-screen instructions to fill out the registration forms. 5. After registering, you will receive an with some more information in it, including the FWD number. 23

26 You are now registered with FWD, and you have a FWD SIP number. 24