How does mute work in a SIP call flow

[tonyppe]

When a SIP call is muted, media RTP is halted. What is meant to happen in terms of SIP signalling? Does this fall under RFC 6337 for media hold? 

I have an issue where the call flow is like: 
softphone1 -> SBC-> SIP trunk -> Phone system -> auto-attendant

When softphone1 goes on mute, a few minutes later, Phone system terminates the call normally. This is because, during mute, Phone system no longer receives RTP and does not receive any SIP signalling to communicate this mute condition. 

I think RFC 6337 covers the mute condition with a "re-INVITE" with attribute value a=inactive or “a=sendonly” but I am unsure. Can anyone confirm? 

Essentially, during a muted condition there must be something sent so the remote side doesnt terminate the call. 

Thanks,

[Roger Kallberg]

I can't name the specific RFC that this pertains to, but for sure the signaling is still ongoing between any of the endpoints and/or system(s) that are part of the call. Different vendors handles this differently, so there is no one answer to your specific question on the attributes in the SIP signaling.

[tonyppe]

Hi Roger - 

I was not able to find RFC or RFCs which explain the handling of mute at all. Only with regards to "media hold". So I was wondering if that indeed was what I was looking for. 

I agree with you, regardless of how it's implemented; something needs to be sent to maintain an active call. Either continuous media or a re-invite. This is my problem.