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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.
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