R&S M3SR 4100 Series, Operating Manual

Page: 655 / 706

Operating Manual 6175.4760.02 – 01 5.109
M3SR Series 4100 Voice over IP (VoIP)
Scenario “Remote Phone Handle”
This configuration can be used to connect a single radio of a cell to a VoIP network to par-
ticipate in the conversation inside the radio cell using the VoIP connection.
No broadcast or directed connection establishment to particular other radios is possible, it is
just an “over IP extension” of the handle.
• SIP mode on the radio must be “on”
• Signalling mode “off”
• SIP settings have to be provided
The invite from the VoIP network sends URIs with a user “LOCAL” e.g. “INVITE LO-
CAL@192.168.1.100”, or empty user part 192.168.1.100.
5.5.1.2 General Constraints
• All radios belonging to the same radio network must have the same settings in the un-
derlying waveform (ALE-3G, SECOM-V), data rates, Voice-over-IP Data, etc.
• Since this service requires the IPoA service to be enabled, all the constraints defined to
the IPoA service must be met.
• VoIP gateway radio is able to handle one SIP (Session Initiation Protocol) call session
at a time.
• Service configuration, as the definition of the VoIP gateway radio and the activation of
the phone service should be performed via R&S RNMS3000.
• PBX or VoIP phone must not offer more than one CODEC. The preferred CODEC is
G.711 A-law.
5.5.1.3 Call Session
Real-time Transport Protocol (RTP) is used to transport packets over the IP network. The
RTP packets are encoded with G.711 A-law. Session Initiation Protocol (SIP) is used to ini-
tiate, modify and terminate the RTP call session and makes use of the Session Description
Protocol (SDP) for defining session parameters.
Only one call session at a time is supported by the VoIP gateway radio. If an external VoIP
user requests a second call session, it will get a busy signal. If a radio requests a call while
another is already being established, the second request will be ignored.
Voice Stream Conversion between Radio and VoIP Network
Once a SIP call session has been established, the VoIP gateway radio is responsible for the
conversion between voice stream (radio network side) and RTP stream (IP network side) as
well as for the transcoding. On IP network side, voice is sent in RTP packets encoded with
G.711. On the radio network side, voice communication occurs using the respective wave-
form and vocoder adopted in this radio network.