EXFO EXpert VoIP Test Tools User Manual
Page 19
Selecting and Starting a Test
EXpert IP
13
SIP
SIP
The SIP (Session Initiation Protocol) test measures the performance of
Voice over IP (VoIP) services within a network. The test uses the SIP for call
signaling and the RTP for digitally transporting the encoded audio.
The SIP signaling portion of the test can be transported over UDP/TCP. SIP
calls transported over UDP/TCP can be routed through a proxy server. The
test's RTP functionality conforms to RFC 3550.
The SIP test measures call statistics for a SIP call and initiates a call to or
answers a call from a given endpoint. It reports call setup and media path
statistics between the test set and a SIP endpoint or another EXpert VoIP
SIP enabled test set.
The SIP test operates in two modes.
Network Active Test
Service Active Test
The SIP network active test is performed, if EXFO devices (i.e. EXpert IP
Test Tools devices) are selected on both ends, that is, on the Controller side
and the endpoint. The SIP network active test operates peer-to-peer in UDP
communication mode along with third-party SIP proxy servers to set up
calls between two endpoints, a Controller, which initiates the call, and a
Responder, which is responsible for completing the call and, optionally,
echoing media packets sent from the Controller. For the SIP network active
test in TCP communication mode third-party SIP proxy servers are
mandatory. A SIP test cycle consists of the initiation and termination of an
audio session (simulated audio data) between two endpoints for which it
reports call setup and audio stream statistics.
The SIP test measures the call statistics. It initiates and answers a SIP call
from a given endpoint. It reports call setup and media path statistics
between the test set and a SIP endpoint or another EXpert VoIP SIP
enabled test set.