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
Once the save was completed, the call transfers were successful.
One thought on “Call drop on transfer – RTP Broken Connection”