Symptom

voip_rtp_allocate_port:Possible port leak? , when call goes on hold

Conditions

Software Version: 20160620_090152_V16_3_0_237 Noticed bunch of following message in log buffer during load run. Jun 8 13:27:59.389 PDT: voip_rtp_allocate_port:Possible port leak? callID(18446744073709551615), port(38164) socket(0x0) Topology: PhoneA----CUCM--------(CUBE)---------SIPTrunk-------- (GW ISR4451)------------T1 PRI ---------(GW ISR )------SIPTrunk--- Found following call scenario that causing the leak; SEA-PSTN-EMU-02#sh voip rtp connections VoIP RTP Port Usage Information: Max Ports Available: 19999, Ports Reserved: 101, Ports in Use: 0 Port range not configured Min Max Ports Ports Ports Media-Address Range Port Port Available Reserved In-use ------------------------------------------------------------------------------ Global Media Pool 8000 48198 19999 101 0 ------------------------------------------------------------------------------ No active connections found 1) PhoneA makes a call, Gateway ISR4451 receives an EO INVITE with following four m-lines , call gets established with ?RTP Ports In Use=3? m=audio 39252 RTP/AVP 0 8 18 101 m=video 13282 RTP/AVP 119 98 m=application 41636 UDP/BFCP * m=video 32932 RTP/AVP 119 99 2) When Phone A goes on hold. And Gateway receives midcall INVITE, *Jun 8 14:13:51.606 PDT: //1/DC4E1C000001/SIP/State/ccsip_cnfsm_debugs: IWF:next_state:CNFSM_NO_STATE_CHANGE *Jun 8 14:13:51.606 PDT: //1/DC4E1C000001/SIP/Info/info/1/sipSPIIncomingCallSDP: SDP already there use old SDP and update media if needed *Jun 8 14:13:51.606 PDT: //1/DC4E1C000001/SIP/Info/verbose/1/sipSPIUpdateSrcSdpFixedPart: RTP port already reserved for stream 1, src_port=8000 *Jun 8 14:13:51.606 PDT: voip_rtp_allocate_port:Possible port leak? callID(18446744073709551615), port(8002) socket(0x0) *Jun 8 14:13:51.606 PDT: //-1/xxxxxxxxxxxx/SIP/Media/sipSPIReserveRtpPort: reserved port 8006 for stream 2 *Jun 8 14:13:51.606 PDT: //1/DC4E1C000001/SIP/Info/verbose/1/sipSPIUpdateSrcSdpFixedPart: Reserving rtp port for stream 2, src_port=8006 *Jun 8 14:13:51.606 PDT: //1/DC4E1C000001/SIP/Info/info/1/sipSPIUpdateSrcSdpFixedPart: stream 3 is dead, src_port=0 *Jun 8 14:13:51.606 PDT: //1/DC4E1C000001/SIP/Info/verbose/1/sipSPIUpdateSrcSdpFixedPart: RTP port already reserved for stream 4, src_port=8004 3) PhoneA resume and then ends the call; SEA-PSTN-EMU-02#sh voip rtp connections VoIP RTP Port Usage Information: Max Ports Available: 19999, Ports Reserved: 101, Ports in Use: 1 Port range not configured Min Max Ports Ports Ports Media-Address Range Port Port Available Reserved In-use ------------------------------------------------------------------------------ Global Media Pool 8000 48198 19999 101 1 ------------------------------------------------------------------------------ No active connections found

Workaround

none

Further Problem Description