Call drop on transfer - RTP Broken Connection

Environment:

Skype for Business Enterprise Edition

Trunk Configuration

Identity : PSTN:GatewayIP
OutboundTranslationRulesList : {}
SipResponseCodeTranslationRulesList : {}
OutboundCallingNumberTranslationRulesList : {}
PstnUsages : {}
Description :
ConcentratedTopology : True
EnableBypass : False
EnableMobileTrunkSupport : False
EnableReferSupport : False
EnableSessionTimer : True
EnableSignalBoost : False
MaxEarlyDialogs : 20
RemovePlusFromUri : False
RTCPActiveCalls : False
RTCPCallsOnHold : False
SRTPMode : Optional
EnablePIDFLOSupport : False
EnableRTPLatching : False
EnableOnlineVoice : False
ForwardCallHistory : False
Enable3pccRefer : False
ForwardPAI : False
EnableFastFailoverTimer : True
EnableLocationRestriction : False
NetworkSiteID

Audiocodes (Remote Site ) <-> Skype for Business Mediation Pool <-> Skype for Business Enterprise Pool

Issue:

During a recent migration from Audiocodes <-> SBA <-> SFB FE Pool to the current Audiocodes <-> SFB Mediation Pool <-> SFB FE Pool, users in the remote site experienced call drops during a transfer.

Regardless of the call flow ie. the call entering via Gateway -> SFB -> Auto Attendant or directly to the users DID when that user transferred the call to another user the call would drop.

Resolution:

We enabled tracing to a syslog server and captured the call information, as we were looking through the logs we found that the Gateway was dropping the call in the Bye event with the

Reason: Q.850 ;cause=31 ;text=”local, RTP Broken Connection”

To fix this issue here is what was done:

Logged into the Audiocodes Gateway
Click Setup -> Signaling&Media -> Expand Coders & Profiles -> Click IP Profiles -> Edit your SFB Profile -> Broken Connection Mode -> Select Ignore -> Click Apply
Expand SIP Definitions -> Click SIP Definitions General Settings -> Broken Connection Mode -> Select Ignore -> Click Apply -> Click Save