Fix Mute timeout on PSTN. Send RTCP or Keepalives rather than No Media
After several months of troubleshooting, Microsoft have finally admitted that the issue of call drops and conference participant drops when muting, are by design. When a participant mutes their client (mobile, desktop, or web), the SIP traffic stops sending media, but doesn;t send any for of keepalive to appease Tier 1 carriers that implement NO RTP timers.
The science behind this is that there are Communications RFC’s from the IETF that all providers agree to. One of these suggests that in the event of no RTP being received from the far end, a timer will be set in order to prevent a stale call session. Some carriers implement a matter of minutes or hours, it’s entirely at their discretions. In the case of BT in the UK, they have their timers set to 120 seconds.
This post is to request that Microsoft amend their SIP signalling or Mute behaviour, in order to fall in line with other PBX vendors and carriers and follow the IETF RFC's on comms.
In light of this, our current workaround is that if clients have a compatible headset, they can retract the mic boom and the hardware will mute itself, or the noise cancelling is so effective that it’s as if you are on mute. For any other methods, the answer is to not use mute for longer than 2 minutes.