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Voice over IP (VoIP) | SIP Telephony |
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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 62 are used for SIP telephony. SIP telephony has the following distinctive features:
■Subscribers are identified through an
■SIP transmits dialling numbers always in a single data package (block dialling). Dialling can therefore be concluded with the hash key #on the system ter- minal, or the end of the number will be indicated by a
■You must log on (“Login”) to the SIP registrar before you can use SIP telephony. Use the Aastra 800 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 con- nections 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
These products provide a “Full Cone NAT” setting for this application.
■To enable the use of multiple devices on a single Internet connection, the IP addresses used in a LAN (often 192.168.x.x) are translated to a valid IP address using address translation (NAT - Network Address Translation), but no status
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