#36 Audio loss in WebRTC

[David Suendermann-Oeft]

When using WebRTC via JSSIP or SIPML5, frequently, parts of the system audio (TTS or prerecorded prompt) are dropped.

There is the following systematic pattern: When a short prompt is played, the next prompt will be silent for about the duration of the short prompt. That is, if the second prompt is shorter than the first, no prompt will be audible. If it is longer, only the later portion will be audible.

However, there seems to be a minimum duration of the first prompt for which this bug vanishes. If the first prompt is longer than this minimum duration, the second prompt will not be subject to audio loss. For example, in Firefox v36 on SIPML5, this minimum duration was determined to be 3.8s.