Guys,
Calls put on hold are dropping after 30 seconds.
My topology: SIP server -> Cisco VG 3825 -> PSTN.
VG 3825 runs the c3825-spservicesk9-mz.151-3.T.bin IOS. The call is dropped because of the RTP/RTCP timer (error message: RTP/RTCP receive timer expired. Disconnect the call.) Below is my config and trace for one dropped call.
Could you please help? Why is this happening? It's says a=inactive for the RTP stream and it drops the call anyway.
Voice gateway configuration:
dial-peer voice 3130 pots (Incoming from PSTN)
permission orig
description Incoming from PSTN
huntstop
incoming called-number .
direct-inward-dial
port 0/0/0:15
!
dial-peer voice 1250000 voip (Outgoing to SIP server)
permission term
description SIP server
preference 4
destination-pattern 3...$
session protocol sipv2
session target ipv4:x.x.x.11
voice-class codec 1
dtmf-relay rtp-nte
no vad
!
voice vad-time 1000
!
gateway
media-inactivity-criteria all
timer receive-rtcp 5
timer receive-rtp 1200
!
Hold re-invite:
Feb 29 21:16:16.337 GMT: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:9@x.x.x.12:5060 SIP/2.0
To: "unknown" <sip:9@x.x.x.12>;tag=13FF4F98-13C5
From: <sip:3777@x.x.x.11>;tag=SEC11-4a0d9509-4b0d9509-1-8Sy9M4cD97c7
Call-ID: 6F4BCA8F-625111E1-BE5CD8D5-9E39E534@x.x.x.12
CSeq: 1 INVITE
Contact: <sip:3777@x.x.x.11:5060;transport=udp;maddr=x.x.x.11>
Via: SIP/2.0/UDP x.x.x.11:5060;branch=z9hG4bKSEC-4a0d9509-4b0d9509-1-dfYDM2uB7r
Content-Type: application/sdp
Content-Length: 301
X-Siemens-Call-Type: ST-insecure
Accept-Language: en;q=0.0
Allow: REGISTER, INVITE, ACK, BYE, CANCEL, NOTIFY, REFER, INFO
Date: Wed, 29 Feb 2012 21:16:16 GMT
Max-Forwards: 70
v=0
o=MxSIP 0 165574749 IN IP4 x.x.x.51
s=SIP Call
c=IN IP4 x.x.x.51
t=0 0
m=audio 16400 RTP/AVP 18 8 0 101
a=rtpmap:18 G729/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=silenceSupp:off - - - -
a=fmtp:18 annexb=no
a=fmtp:101 0-15
a=inactive
Feb 29 21:16:16.349 GMT: //9054239/6F48BDC8B14E/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP x.x.x.11:5060;branch=z9hG4bKSEC-4a0d9509-4b0d9509-1-dfYDM2uB7r
From: <sip:3777@x.x.x.11>;tag=SEC11-4a0d9509-4b0d9509-1-8Sy9M4cD97c7
To: "unknown" <sip:9@x.x.x.12>;tag=13FF4F98-13C5
Date: Wed, 29 Feb 2012 21:16:16 GMT
Call-ID: 6F4BCA8F-625111E1-BE5CD8D5-9E39E534@x.x.x.12
CSeq: 1 INVITE
Allow-Events: telephone-event
Server: Cisco
Content-Length: 0
Feb 29 21:16:16.349 GMT: //9054239/6F48BDC8B14E/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK
Via: SIP/2.0/UDP x.x.x.11:5060;branch=z9hG4bKSEC-4a0d9509-4b0d9509-1-dfYDM2uB7r
From: <sip:3777@x.x.x.11>;tag=SEC11-4a0d9509-4b0d9509-1-8Sy9M4cD97c7
To: "unknown" <sip:9@x.x.x.12>;tag=13FF4F98-13C5
Date: Wed, 29 Feb 2012 21:16:16 GMT
Call-ID: 6F4BCA8F-625111E1-BE5CD8D5-9E39E534@x.x.x.12
CSeq: 1 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Contact: <sip:9@x.x.x.12:5060>
Supported: replaces
Supported: sdp-anat
Server: Cisco
Supported: timer
Content-Type: application/sdp
Content-Length: 270
v=0
o=CiscoSystemsSIP-GW-UserAgent 3635 7861 IN IP4 x.x.x.12
s=SIP Call
c=IN IP4 x.x.x.12
t=0 0
m=audio 18022 RTP/AVP 18 101
c=IN IP4 x.x.x.12
a=inactive
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
Feb 29 21:16:16.361 GMT: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:9@x.x.x.12:5060 SIP/2.0
To: "unknown" <sip:9@x.x.x.12>;tag=13FF4F98-13C5
From: <sip:3777@x.x.x.11>;tag=SEC11-4a0d9509-4b0d9509-1-8Sy9M4cD97c7
Call-ID: 6F4BCA8F-625111E1-BE5CD8D5-9E39E534@x.x.x.12
CSeq: 1 ACK
Contact: <sip:3777@x.x.x.11:5060;transport=udp;maddr=x.x.x.11>
Via: SIP/2.0/UDP x.x.x.11:5060;branch=z9hG4bKSEC-4a0d9509-4b0d9509-1-WmWhf716Y6
Date: Wed, 29 Feb 2012 21:16:16 GMT
Max-Forwards: 70
Content-Length: 0
Feb 29 21:16:42.130 GMT: //9054239/6F48BDC8B14E/SIP/Info/ccsip_indicate_rt_packet_stats: Processing stats for callid=9054239, proc_id=9
Feb 29 21:16:43.390 GMT: //9054239/6F48BDC8B14E/SIP/Info/ccsip_set_cc_cause_for_spi_err: Categorized cause:41, category:185
Feb 29 21:16:43.390 GMT: //9054239/6F48BDC8B14E/SIP/Info/ccsip_set_release_source_for_peer: ownCallId[9054239], src[6]
Feb 29 21:16:43.390 GMT: //9054239/6F48BDC8B14E/SIP/Info/sipSPIRtpDiscTimerExpired: RTP/RTCP receive timer expired. Disconnect the call.
Feb 29 21:16:43.390 GMT: //9054239/6F48BDC8B14E/SIP/Media/sipSPIUpdateRtpSession: stun is disabled for stream:69EA02C0
Feb 29 21:16:43.390 GMT: //9054239/6F48BDC8B14E/SIP/Info/ccsip_call_statistics: Requesting stats for callid=9054239
Feb 29 21:16:43.390 GMT: //9054239/6F48BDC8B14E/SIP/Info/sipSPI_ipip_antiTrombone: Entered Antitrombone service
Feb 29 21:16:43.390 GMT: //9054239/6F48BDC8B14E/SIP/Info/sipSPI_ipip_antiTrombone: Antitrombone service not configured
Feb 29 21:16:43.390 GMT: //9054239/6F48BDC8B14E/SIP/Info/act_active_disconnect: Disconnect deferred, as stats request pending
Feb 29 21:16:43.398 GMT: //9054239/6F48BDC8B14E/SIP/Info/ccsip_indicate_rt_packet_stats: Processing stats for callid=9054239, proc_id=1
Feb 29 21:16:43.398 GMT: //9054239/6F48BDC8B14E/SIP/Info/sipSPI_ipip_antiTrombone: Entered Antitrombone service
Feb 29 21:16:43.398 GMT: //9054239/6F48BDC8B14E/SIP/Info/sipSPI_ipip_antiTrombone: Antitrombone service not configured
Feb 29 21:16:43.398 GMT: //9054239/6F48BDC8B14E/SIP/Info/sipSPIStopHoldTimer: Stopping hold timer
Feb 29 21:16:43.398 GMT: //9054239/6F48BDC8B14E/SIP/Info/sipSPIStopHoldTimer: Stopping hold timer
Feb 29 21:16:43.398 GMT: //9054239/6F48BDC8B14E/SIP/Info/sipSPISendBye: Associated container=0x69C40820 to Bye
Feb 29 21:16:43.398 GMT: //9054239/6F48BDC8B14E/SIP/Transport/sipSPISendBye: Sending BYE to the transport layer
Feb 29 21:16:43.398 GMT: //9054239/6F48BDC8B14E/SIP/Transport/sipSPIGetSwitchTransportFlag: Return the Dial peer configuration, Switch Transport is FALSE
Feb 29 21:16:43.398 GMT: //9054239/6F48BDC8B14E/SIP/Transport/sipSPITransportSendMessage: msg=0x69B3F668, addr=x.x.x.11, port=5060, sentBy_port=0, local_addr=x.x.x.12, is_req=1, transport=1, switch=0, callBack=0x61AC0344
Feb 29 21:16:43.402 GMT: //9054239/6F48BDC8B14E/SIP/Transport/sipSPITransportSendMessage: Proceedable for sending msg immediately
Feb 29 21:16:43.402 GMT: //9054239/6F48BDC8B14E/SIP/Transport/sipTransportLogicSendMsg: switch transport is 0
Feb 29 21:16:43.402 GMT: //9054239/6F48BDC8B14E/SIP/Transport/sipTransportLogicSendMsg: Set to send the msg=0x69B3F668
Feb 29 21:16:43.402 GMT: //9054239/6F48BDC8B14E/SIP/Info/sentByeDisconnecting: Sent Bye Request, starting DisconnectTimer
Feb 29 21:16:43.402 GMT: //9054239/6F48BDC8B14E/SIP/State/sipSPIChangeState: 0x69705A68 : State change from (STATE_ACTIVE, SUBSTATE_NONE) to (STATE_DISCONNECTING, SUBSTATE_NONE)
Feb 29 21:16:43.402 GMT: //9054239/6F48BDC8B14E/SIP/Msg/ccsipDisplayMsg:
Feb 29 21:16:43.402 GMT: //9054239/6F48BDC8B14E/SIP/Msg/ccsipDisplayMsg:
Sent:
BYE sip:3777@x.x.x.11:5060;maddr=x.x.x.11;transport=udp SIP/2.0
Via: SIP/2.0/UDP x.x.x.12:5060;x-route-tag="tgrp:tg_voice_pri";branch=z9hG4bK1EE7AE312
From: "unknown" <sip:9@x.x.x.12>;tag=13FF4F98-13C5
To: <sip:3777@x.x.x.11>;tag=SEC11-4a0d9509-4b0d9509-1-8Sy9M4cD97c7
Date: Wed, 29 Feb 2012 21:16:16 GMT
Call-ID: 6F4BCA8F-625111E1-BE5CD8D5-9E39E534@x.x.x.12
User-Agent: Cisco
Max-Forwards: 70
Timestamp: 1330550203
CSeq: 102 BYE
Reason: Q.850;cause=102
P-RTP-Stat: PS=201,OS=4020,PR=194,OR=3880,PL=0,JI=0,LA=0,DU=31
Content-Length: 0
