We are proud to announce that RestComm Media Server 3.0.3-M1 has been released to the public.
This is the first milestone of the upcoming 3.0.3 release that will focus on solidifying the current audio implementation and addressing technical issues like memory leaks and CPU consumption.
One of the most notable issues addressed by this release was the increasing latency in WebRTC calls which is now gone! Users can now connect their mobile clients, SIP phones, landline phones and browsers and have a smooth conference call – RestComm Media Server bridges them all!
At the time of this release, we are aware of audio glitches that can be heard in conference calls.
The rate at which the Media Server reads incoming packets had to be increased to comply with WebRTC calls that multiplex RTP, RTCP and STUN in a single channel. This breaks the 20ms rule on which the Media Server’s internal scheduler has been built on top of, so adjustments need to be done. We expect to have this problem fixed in the near future.
The RestComm Media Server 3.0.3-M1 binary is available for download here and the source code can be found on github.
We would like to ask our community to help testing and to provide their feedback in our public forum discussion. Feel free to submit your questions to StackOverflow as well.
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