Hello all,
I am working on an Android app that uses PJSIP 2.6 to make VOIP calls. We have encountered issues with our app when it hosts a 3 way conference call, especially with Wifi networks that can be spotty and earlier versions of Android devices. We notice that when the call quality is bad, sound frames are often lost and synthetic frames must be generated, as follows:
16:21:36.231 Master/sound Underflow, buf_cnt=56, will generate 1 frame
16:21:34.422 strm0xee995814 !Frame lost, recovered!
16:20:21.336 strm0xee996414 Jitter buffer starts returning normal frames (after 1 empty/lost)
There are more of these messages that show up when the call quality is especially bad, indicating that perhaps the sound device/codec is not receiving enough frames(we use a ptime of 20ms). The audio issues we face are generally that it breaks up a lot and is of low volume. We are not sure how much of it is an issue with the Android device/pjsip vs network issues. We have tried adjusting some settings, including PJMEDIA_SOUND_BUFFER_COUNT, PJMEDIA_DELAY_BUF_SIMPLE_FIFO, PJMEDIA_CODEC_SPEEX_DEFAULT_QUALITY,
and others. They have not created much improvement in the audio quality across the Android devices we've tested.
We do not any issues with the delay in the frames received, so I would like to know what I can do on the pjsip side to optimize call quality at the expense of some added delay. The idea is to make our app more robust to changes in network quality so that the call quality can remain as optimal as possible regardless of changes in network strength or signal(barring a complete loss of connectivity of course).
-manav95
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