Voip Connection Establishment - R&S 4200 Series Operating Manual

Audio via VoIP (Optional)
3.15.7

VoIP Connection Establishment

Introduction
Session Initiation
Audio
Communication
This chapter explains the way from session initiation to audio transmission
via VoIP. The explanation of the connection establishment is independent
from the radio modules.
VoIP is a session-based communication standard. This means that audio
data transmission requires a mechanism to synchronize the two endpoints
(RX/TX and VCS/R&S GB4000V) also called user agents (UA). This
synchronization of endpoint capabilities is done by SIP (see chapter 3.15.1)
and SDP (see chapter 3.15.1). In this case SIP is used to initiate a session
and SDP is used for negotiating the endpoint capabilities.
The connection establishment between two user agents contains two
phases:
 Session initiation (see chapter 3.15.6)
 Audio communication (see chapter 3.15.6)
During session initiation the user agent one (VCS or R&S GB4000V) tries to
connect to the user agent two (R&S S4200) by using the session initiation
protocol (SIP). The identification between the two user agents is done by
unified resource identifier (URI). The user agent two (R&S S4200) checks
its Emergency and Normal VoIP ACLs whether the requesting user agent is
allowed to connect. The VoIP ACL contains the URIs being allowed to
connect.
Note: The VoIP ACL can be used as a whitelist. This allows all user agents
to connect to the radio (no URI restriction).
By using the session description protocol (SDP) both user agents negotiate
their capabilities (e.g. used codec etc.). If the whole negotiating process
succeeds, the session between both user agents is established. This
establishment is the basis for the audio communication. This means that
without an established SIP session no audio data can be transferred.
Note: The SIP session keeps existing until one of the user agents
terminates the session
After successful establishment of a SIP session the result of the negotiating
process between both user agents will be used to configure the RTP-based
audio transmission (e.g. codec to use, ports to use etc.).
Note: The real time transport protocol uses even ports higher than 5000.
If one user agent terminates the SIP session, the corresponding RTP
connection will also be terminated.
3.109
or the network
R&S XU 4200
connection fails.
6166.5368.02.01