4  
Features for Internet telephony 
 

4.1  
Voice over IP 
 

Introduction  
Voice over IP (VoIP) is used to transfer voice data over IP-based networks (packet-switched networks).
To make calls, communication must be established between the workpoints involved before the actual call takes place. This type of communication is known as signaling and is possible over
 •   the comprehensive standard H.323.
In addition to signaling protocols, this standard includes protocols for exchanging workpoint functionality, call control, exchanging status information, and data flow control.
H.323 is a protocol defined by the ITU with limited expansion options.
 •   the Session Initiation Protocol (SIP)
The SIP protocol is an ASCII-based signaling protocol used to configure sessions in an IP network.
The SIP protocol is based on well-established Internet technologies, such as, HTML and e-mail, and provides seamless integration into the Internet protocol architecture. SIP can be used for various application scenarios while the protocol itself is restricted to handling signaling functions. SIP can, therefore, be used in conjunction with other protocols for different purposes.
The SIP protocol has prevailed over H.323 in Internet telephony. Almost all key ITSPs exclusively implement the SIP protocol. HiPath 3000 supports both SIP and H.323 in the same network.
Apart from analog and digital trunk connections (POTS or ISDN), stations can also be reached via ITSPs for connections to the CO. HiPath implements Internet telephony from ITSP via DSL telephone connections to the LAN. Internet telephony connections are set up with SIP. HiPath supports both Internet telephony user connections and Internet telephony system connections. Internet telephony connections can be established in HiPath standalone environments and HiPath networks.

Gateway, gatekeeper, registrar, SIP server  
HiPath supports the following functions:
 •   Gateway
 •   Gatekeeper (for internal IP workpoints)
 •   Registrar (for the internal SIP telephone)
 •   SIP server (Back to Back User Agent) (for the internal SIP telephone)
Gateways are required for communication between IP-supported networks (LAN, intranet, Internet) and line-switched networks (POTS, ISDN).
Version 6.0 SMR-09 always requires the use of a gateway. It is only possible to connect to the SIP provider a DSP, irrespective of whether a TDM or IP phone is connected to an SIP provider. Version 7 supports direct payload connection to the SIP provider.
 •   In HiPath 3000/5000 V7 R2 and later systems, IP/IP e2e payload via enterprise proxy is supported for Internet telephony. This means that for Internet telephony, no data packets need be processed by the system's digital signal processors. The calls are routed over LAN, router and the Internet Telephony Service Provider (ITSP). HiPath 3000/5000 sets up a call to the remote station. This means that no gateway channels (DSP channels) are required for ITSP connections. The maximum number of simultaneous calls to the ITSP depends on the codec used and the bandwidth available on the DSL connection. The maximum number of simultaneous calls is 30.
 •   With Internet telephony, it is possible to send faxes via T.38 channels from HiPath 3000/5000 V7 R4.
Gatekeepers perform the following tasks:
 •   IP workpoint registration
   -   System clients such as OpenStage IP, optiPoint 600, OpenScape Personal Edition, optiPoint 150 S, optiPoint 410 and optiPoint 420
   -   Connection setup
   -   Access control
A registrar (gatekeeper/HG 1500 board) is needed to register SIP clients, such as optiPoint 150 S, optiPoint 410 S, optiPoint 420 S on the system.
A SIP server performs connection control and access control.

HiPath 3000/5000 V9, Feature Description, Issue 7   up Top
Disclaimer & Copyright   ID: P31003H3590F100017618   2012-06-25