Make timestamp conform to https://tools.ietf.org/html/draft-ietf-payload-rtp-opus-11

[Jeyhey]

Just a hint: According to https://tools.ietf.org/html/draft-ietf-payload-rtp-opus-11 page 6 paragraph 4, for OPUS the timestamp has to be increased by 960 every frame regardless of the frame-size and sample rate. You can easily implement it by introducing in RtpStreamSender on line 568 the following two lines:

else if (p_type.codec.number() == 98) 
                time += 960;

This will make lumicall compatible with webRTC ; ). Otherwise the voice will be strange on the webRTC side.
(By the way webRTC uses the number 111 for opus).