* Added SDP signaling of TIAS modifier for AAC
* Setting marker bit for every RTP packet containing AAC frames
* Ensuring that audio buffer size in RtpStreamSender is at least 4 KB for AAC and Opus

Signed-off-by: jhowes <howes.julian@gmail.com>
Merge branch 'master' of https://github.com/juho0006/sipdroid-pro