Voice over IP (VoIP) SIP Telephony
■ The MGW interface card contains the required software in its own Flash memory. This software is updated using a separate file in the Configurator via the SYS Configuration: Components.
Operational Information
The MGW interface card must be correctly inserted and configured (see also
Installing Interface Cards starting on page 29).
Each MGW interface card requires its own IP address. This can either be assigned statically or obtained via DHCP.
1. In the Configurator, open the PBX Configuration: Ports: Slots page.
2. Click on the slot number in the table column containing the desired interface card.
3. Enter the desired static IP address in the IP address configured field. Enter
“0.0.0.0” to obtain an IP address via DHCP.
4. Confirm with Apply.
The configuration page displays the MGW interface card’s MAC address. You will need this for the static assignment of an IP address via a DHCP server.
Note: During the commissioning of a MGW interface card, the software MGW function is automatically switched off.
9.3
SIP Telephony
The SIP Internet (Session Initiation Protocol) protocol provides you with a low-cost, standardised option for telephoning via the Internet. OpenCom 100 enables you to use external SIP telephone connections (SIP trunk lines). The PBX Configu-
ration: Trunks: Route menu gives you the option of making an SIP connection on a normal ISDN line in case of bundle overflow, breakdown or over-occupancy. You can also set up your system to route certain types of calls, such as international calls, to an SIP connection.
Note: You will need a Media Gateway card for SIP telephony.
You will also need a fast Internet connection such as DSL for SIP telephony.
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Voice over IP (VoIP) SIP Telephony
You will also usually need the services of a SIP provider. A SIP provider operates a special server (the SIP Registrar) to handle connections. The SIP provider also operates a gateway to the ordinary telephone network which users pay to use and which enables the SIP provider to provide calls to the telephone network.A SIP connection can also accept incoming calls from the telephone network.
The same voice transmission techniques as those explained in Fundamentals starting on page 100 are used for SIP telephony. SIP telephony has the following
distinctive features:
■ Subscribers are identified through an e-mail-like “SIP ID” such as
[email protected] or [email protected].
■ SIP transmits dialling numbers always in a single data package (block dialling).
Dialling can therefore be concluded with the hash key # on the system terminal, or the end of the number will be indicated by a time-out. The value for this time-out can be defined for each SIP provider separately.
■ You must log on (“Login”) to the SIP registrar before you can use SIP telephony.
Use the OpenCom 100 to manage important information for the registration
(user name and password) of one or more SIP accounts. It is possible to make several calls simultaneously using a single SIP account.
■ A SIP connection causes constant Internet data traffic, so do not use SIP with
Internet access which is paid for according to the time used.
■ RTP call data is also exchanged directly between terminals for SIP telephony, so different codecs can be used for sending and for receiving. It is also possible to change codecs dynamically during a call. You should use every codec available in the VoIP profile at least once, because this will enable you to establish connections with as many SIP subscribers as possible.
■ Fairly large packet lengths are quite normal on the Internet. They compensate for the longer packet propagation delay.
■ A bidirectional RTP data stream with a dynamically-assigned UDP port number is used to set up calls between subscribers. For this reason, incomng RTP calls often fail to get past the Firewall or NAT configuration of the Internet gateway product used. Do not use OpenCom 100 as an Internet gateway if the product used is to be compatible with SIP telephony. These products provide a “Full
Cone NAT” setting for this application.
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