Call Playback

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:

sip2_and_play1

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:

player_en

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.