Telco sip call cur when caller on mute

[iptuser55]

We have a sip trunk via our cube to a telco for external calls. If a user is on a call and on mute as with a conference call they get cut off after a while - telco does not get any rpt.

They suggested silence susuppression

I saw other disdiscussions but that was with mgcp

Any ideas 

[R0g22]

Add "no vad" under the VoIP dial-peers and then test.

[iptuser55]

Already has no vad

[iptuser55]

The telco mentioned something like rtp suppression but I've not heard of it

[R0g22]

Do you have any debugs for ccsip message to check ? When you say cut off, do you mean that the call drops or is there no RTP at all ? Why is your ISP complaining about not getting any RTP ?

[iptuser55]

The telco states that since the caller is on mute, there is no rtp and so
thinks there is no call, activity and disconnects. Caller takes off mute
then no problem as there is background noise chat etc and so the call
stays opens

[R0g22]

Got it. So basically your Telco is looking for some kind of activity to not drop calls.
One functionality is to have comfort noise. You would need to have VAD enabled for that. Once VAD will kick in, comfort noise will be added to already existing audio stream.
Now you can enable VAD on dial-peers and on SIP trunk from CUCM to test this. You can use "voice vad-time <time in msec>" command to tune the transition to comfort noise. Also a high value delays the VAD kick in. There are some cons to this as well. Read the following -
https://www.cisco.com/c/en/us/support/docs/voice/voice-quality/22388-hissing.html#topic3

[Mohammed al Baqari]

Can you check if media inactivity detection is enabled on CUBE or not. I
think this is the problem