Phone Service - R&S M3SR 4100 Series Operating Manual

M3SR Series 4100
After the VoIP gateway radio has transcoded the incoming message from the IP network, it
is then able to send the voice stream to all radios in this radio network. Three mechanisms
are supported for the activation of the VoIP gateway transmission:
• PTT bit in the RTP header extension
PTT bit indication in the RTP header extension is supported as defined by EUROCAE
137-1. Only if the PTT bit in the RTP header extension is set, will VoIP gateway radio
transmission be activated, i.e., voice stream will be broadcast to the radios in the radio
network.
• Use of phone keys
The use of phone keys indicating the activation and the deactivation of the VoIP gateway
radio transmission is also supported, for e.g. phone keys '1' and '2'. These keys can be
configured via R&S RNMS3000 and it is also possible to have the same phone key for
activation and deactivation. The transmission of the DTMF tones is done as RTP event
according to RFC2833.
• VOX
VOX functionality at the VoIP gateway radio is as well supported for the activation of the
VoIP gateway radio transmission.
Voice Communication inside the Radio Network during a Session
During an active VoIP connection the external VoIP user having a session with the VoIP
gateway radio is able to listen to any communication taking place inside the radio network,
which is also addressed to the VoIP gateway radio.
Only R&S M3TR: For the configuration of the Squelch Line see 'Switching the Squelch Line
On or Off'.
5.5.1.4

Phone Service

Phone book
Each radio in the radio network has a phone book, with preconfigured phone entries. Up to
20 phone book entries can be defined, each consisting of a name, a number and an address.
Phone book entries are selectable from the MMI (Man-Machine Interface) and by remote
command (GB2PP). Only the name of the phone book entry is displayed on the MMI.
Only R&S M3TR: The phone book is also used for a speed dial in the phone patch
mode for fixed frequency (see 'Basics of Phone Patch Mode'). In this case, the dialled
numbers are converted into DTMF (Dual Tone Multi Frequency) tones locally and sent
over the air. For the VoIP Service, the dialled number is sent over the air to the VoIP
gateway radio, which adapts this message into a SIP invite request.
5.110
Voice over IP (VoIP)
Operating Manual 6175.4760.02 – 01