A call session is managed using SIP INFO messages with JSON content. For now only muting/unmuting the session is supported. Next versions will add support for ejecting a participant, getting the list of participants, getting the call state (packet loss, RTT, audio/video quality, etc etc
The MCU could detect that a session is muted based on the RTP packets but it’s highly recommended to also send a SIP INFO message for confirmation. For audio-only sessions, muting a session without sending a SIP INFO could be interpreted as a crash or network issue which automatically disconnects the call.
When the “hangout” video pattern is selected the MCU renders the speaker's video with the highest quality and size. Detecting a speaker could be problematic when the participants are in a noisy environment. Manually muting/unmuting your session is a way to avoid such issues.
JSON content:
| Field name | Field value | Type | Availability |
| action | “req_call_mute” | String | Mandatory |
| enabled | < user defined > | Boolean | Mandatory |
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