Hello Cisco-Community !
We actually face a severe problem concerning mostly our cellphone or landline call-in users in WebEx meetings or even conversations from PSTN (mostly vodafone mobile) to normal CUCM SIP registered internal office phones. It randomly but often happens that the calling in users will not hear any audio after exactly 15minutes (900 seconds) of being dialed in to our infrastructure. They still can be heard by the other meeting participants though ! Anyway they will still have to hang up their phone and call in again to join again for possibly just the next 15 minutes ... After some internal tests we can confirm that if the issue occurs and you (as called in cellphone user) place the call on hold and unhold it again, you will be able to hear everything again. As we are still in the VOIP migrations process we actually run a CUCM 12.5 cluster with 2 redundant Cisco ISR Voice gateways that are connected to our old Sophos telephone system (which is the central PSTN transfer point) via E1. From various captures at the ISR we can tell the RTP stream itself doesn´t break down and you can not see anything going on when placing the call on hold. We are at the moment in the process of decoding a captured RTP stream from the Voice Gateway at the moment to check if the actual audio is already lost at this point or possibly just not transmitted at another.
We consider the whole issue may be a timer problem somewhere and therefor enabled the sip session refresh on the ISR as well as set min-se 1800 session-expires 1800 without any changes.
Is there anybody out here that may faced the same issues and can give us an idea on where to go on ?
Best and thanks a lot !
