|
|
|
|
Note:
The VoIP analysis module is only available to VoIP license users or
evaluation version users who selected VoIP evaluation mode.
|
|
|
|
|
|
|
|
|
|
The call playback functionality can be used for assessing the audio
quality experienced by the parties participating in a VoIP call. In
most of the cases, VoIP analyzer allows you to play captured calls
(this depends on the support of the specific codec(s) being used in
the given VoIP call). To play a call, select the desired call
record in the VoIP analyzer window, select the
RTP Streams
tab, and click on the
Play
button:
Alternatively, you can select any item on the right pane containing
the list of RTP streams (for example, the
RTP Streams
category), select one or several streams, right-click on them, and
select the
Play Selected
menu item. That way, it's possible to interrelate and play back the
streams for which the signaling session is either absent, or the
signaling protocol is not supported (i.e. the protocol is not SIP
or H.323).
|
|
|
|
Note:
Simultaneous playback of RTP streams that belong to different calls
initiated at different times usually won't work. The main problem
is the significant time discrepancy between the streams that belong
to different VoIP calls, aside from the fact that it makes no sense
to listen to unrelated audio that is part of unrelated calls. The
functionality that allows selecting arbitrary RTP streams for
subsequent playback is provided solely for manual recovery of a
call from several streams in cases where parent SIP or H.323
sessions are not available.
|
|
|
|
|
|
|
|
|
|
After clicking on the
Play
button, the Media Stream Player window will be opened:
Click on the double-arrow button to have the application display
more detailed information about the audio stream(s) and access to
manual codec mapping. For each of the RTP streams you can:
·Manually
synchronize a stream by time, i.e. set the starting time of
playback in relationship to other streams. To do that, move the
small triangle to the left or to the right.
·Select
the correct sound codec for each of the payload types in the RTP
stream. In most cases, Media Stream Player will automatically
select the correct codec. However, when working with "orphan" RTP
streams that lack parent SIP or H.323 sessions, and, therefore,
information on the codecs being used, you will have to manually
select the correct codec from the drop-down list. If you find it
difficult to pick the correct codec, try clicking on the
Try to Guess
button and Media Stream Player will try to select the codec by
itself.
Note that sometimes, it's not possible to play back audio from RTP
streams, as these streams may be encrypted or use proprietary
codecs or codecs not supported by CommView.
The
Volume
control allows you to adjust the sound volume. The
Jitter buffer size
control allows you to simulate the jitter buffer used in real world
VoIP end nodes. A typical jitter buffer is 30 ms to 50 ms in size.
Increasing the buffer size improves the voice quality but increases
the delay.
|