Hi list, there must be a hidden timeout between the initial INVITE and the first provisional response with SDP - it seems to equal 60 seconds. Therefore you hear nothing if you pick up your call after 60 seconds. I tried to document a whole call, going to Voicemail after 61 seconds. While doing my tests I also played around with some undocumented setting, without success (relay_recover_interval).
Here the call trace (please note that the same call is fine if VM picks up the call within 60 seconds): # Call from PSTN (0212345678) to local user (0276543210, # [email protected], vmbox: 12345): 13:49:50 OpenSIPS[762]: New dialog from trusted peer - M=INVITE RURI=sip:[email protected] F=sip:[email protected] T=sip:[email protected] SRC=80.2.3.4:5060 [email protected] 13:49:50 OpenSIPS[762]: Callee was aliased, PSTN - M=INVITE RURI=sip:[email protected] F=sip:[email protected] T=sip:[email protected] SRC=80.2.3.4:5060 [email protected] 13:49:50 OpenSIPS[762]: Setting ring timeout to 60 secs - M=INVITE RURI=sip:[email protected] F=sip:[email protected] T=sip:[email protected] SRC=80.2.3.4:5060 [email protected] # Two branches are forked, however there is only one of them in my logs # (need to move that xlog()-statement to branch_route): 13:49:50 OpenSIPS[762]: Local user online - M=INVITE RURI=sip:[email protected]:1025;line=iqdxv6hz F=sip:[email protected] T=sip:[email protected] SRC=80.2.3.4:5060 [email protected] 13:49:50 OpenSIPS[762]: Request leaving server, D-URI='sip:111.2.3.5:51076' - M=INVITE RURI=sip:[email protected]:1025;line=iqdxv6hz F=sip:[email protected] T=sip:[email protected] SRC=80.2.3.4:5060 [email protected] 13:49:50 media-dispatcher[1697]: debug: Issuing "update" command to relay at 90.1.2.3 # On one of the relay hosts the call is prepared: 13:49:50 media-relay[690]: debug: Received new SDP offer 13:49:50 media-relay[690]: mediaproxy.mediacontrol.StreamListenerProtocol starting on 54436 13:49:50 media-relay[690]: mediaproxy.mediacontrol.StreamListenerProtocol starting on 54437 13:49:50 media-relay[690]: mediaproxy.mediacontrol.StreamListenerProtocol starting on 54438 13:49:50 media-relay[690]: mediaproxy.mediacontrol.StreamListenerProtocol starting on 54439 13:49:50 media-relay[690]: debug: Added new stream: (audio) 100.3.4.5:60846 (RTP: Unknown, RTCP: Unknown) <-> 90.1.2.3:54436 <-> 90.1.2.3:54438 <-> Unknown (RTP: Unknown, RTCP: Unknown) 13:49:50 media-relay[690]: debug: created new session [email protected]: [email protected] (69D9B578-1DC8) --> [email protected] # Both devices are ringing: 13:49:50 OpenSIPS[769]: Reply - S=100 D=Trying F=sip:[email protected] T=sip:[email protected] SRC=111.2.3.4:61000 [email protected] 13:49:50 OpenSIPS[771]: Reply - S=180 D=Ringing F=sip:[email protected] T=sip:[email protected] SRC=111.2.3.5:51076 [email protected] 13:49:50 OpenSIPS[757]: Reply - S=180 D=Ringing F=sip:[email protected] T=sip:[email protected] SRC=111.2.3.4:61000 [email protected] # Timeout happens, 60 secs are over, call is redirected to VM Server, # the two active branches are cancelled 13:50:51 OpenSIPS[779]: Request leaving server in transaction - M=INVITE RURI=sip:[email protected] F=sip:[email protected] T=sip:[email protected] SRC=80.2.3.4:5060 [email protected] 13:50:51 OpenSIPS[776]: Reply - S=100 D=Trying F=sip:[email protected] T=sip:[email protected] SRC=90.1.2.1:5060 [email protected] 13:50:51 OpenSIPS[758]: Reply - S=487 D=Request Terminated F=sip:[email protected] T=sip:[email protected] SRC=111.2.3.5:51076 [email protected] 13:50:51 OpenSIPS[767]: Reply - S=487 D=Request Cancelled F=sip:[email protected] T=sip:[email protected] SRC=111.2.3.4:61000 [email protected] # VM Server waits 1 additional second and then picks up the call (early # media): 13:50:52 OpenSIPS[770]: Reply - S=183 D=Session Progress F=sip:[email protected] T=sip:[email protected] SRC=90.1.2.1:5060 [email protected] 13:50:52 media-dispatcher[1697]: debug: Issuing "update" command to relay at 90.1.2.3 # The update reaches the relay - no RTP info yet: 13:50:52 media-relay[690]: debug: Got traffic information for stream: (audio) 100.3.4.5:60846 (RTP: Unknown, RTCP: Unknown) <-> 90.1.2.3:54436 <-> 90.1.2.3:54438 <-> Unknown (RTP: 90.1.2.1:10262, RTCP: Unknown) 13:50:52 media-relay[690]: debug: updating existing session [email protected]: [email protected] (69D9B578-1DC8) --> [email protected] 13:50:52 media-relay[690]: debug: Received updated SDP answer 13:50:52 media-relay[690]: debug: Got initial answer from callee for stream: (audio) 100.3.4.5:60846 (RTP: Unknown, RTCP: Unknown) <-> 90.1.2.3:54436 <-> 90.1.2.3:54438 <-> 90.1.2.1:10262 (RTP: 90.1.2.1:10262, RTCP: Unknown) # Traffic starts in one direction, however I'm unable to hear something: 13:50:57 media-relay[690]: debug: Got traffic information for stream: (audio) 100.3.4.5:60846 (RTP: Unknown, RTCP: Unknown) <-> 90.1.2.3:54436 <-> 90.1.2.3:54438 <-> 90.1.2.1:10262 (RTP: 90.1.2.1:10262, RTCP: 90.1.2.1:10263) # After the announcement the call is accepted, message registration # starts: 13:51:03 OpenSIPS[757]: Reply - S=200 D=OK F=sip:[email protected] T=sip:[email protected] SRC=90.1.2.1:5060 [email protected] 13:51:03 media-dispatcher[1697]: debug: Issuing "update" command to relay at 90.1.2.3 13:51:03 OpenSIPS[776]: Loose routing detected - M=ACK RURI=sip:[email protected]:5060 F=sip:[email protected] T=sip:[email protected] SRC=80.2.3.4:5060 [email protected] 13:51:03 OpenSIPS[776]: Request leaving server, D-URI='<null>' - M=ACK RURI=sip:[email protected]:5060 F=sip:[email protected] T=sip:[email protected] SRC=80.2.3.4:5060 [email protected] # Update reaches the relay - but RTP information for the caller is still # unknown (please note that SDP is fine, correctly parsed by QOS modul, # available via MI - and also fine for Mediaproxy if timeout is below 60 # secs): 13:51:03 media-relay[690]: debug: updating existing session [email protected]: [email protected] (69D9B578-1DC8) --> [email protected] 13:51:03 media-relay[690]: debug: Received updated SDP answer 13:51:03 media-relay[690]: debug: Unchanged stream: (audio) 100.3.4.5:60846 (RTP: Unknown, RTCP: Unknown) <-> 90.1.2.3:54436 <-> 90.1.2.3:54438 <-> 90.1.2.1:10262 (RTP: 90.1.2.1:10262, RTCP: 90.1.2.1:10263) # If I do not hang up and wait (hearing nothing) - after some seconds # the call has been teared down: 13:51:20 media-relay[690]: debug: expired session [email protected]: [email protected] (69D9B578-1DC8) --> [email protected] 13:51:20 media-relay[690]: (Port 54436 Closed) 13:51:20 media-relay[690]: (Port 54437 Closed) 13:51:20 media-relay[690]: (Port 54438 Closed) 13:51:20 media-relay[690]: (Port 54439 Closed) # Dispatcher receives the timeout and tells OpenSIPS to stop the call: 13:51:20 media-dispatcher[1697]: session with call_id [email protected] from relay 90.1.2.3 did timeout 13:51:20 media-dispatcher[1697]: debug: Got statistics: {'from_tag': '69D9B578-1DC8', 'dialog_id': '2475:922919439', 'start_time': 1245412252.01, 'timed_out': True, 'call_id': '[email protected]', 'to_tag': 'as4a00af3c', 'streams': [{'status': 'no-traffic timeout', 'caller_codec': 'Unknown', 'post_dial_delay': 61.350157022499999, 'callee_codec': 'G711a', 'start_time': 0, 'caller_bytes': 0, 'callee_bytes': 0, 'caller_packets': 0, 'end_time': 28, 'callee_remote': '90.1.2.1:10262', 'caller_remote': 'Unknown', 'media_type': 'audio', 'callee_local': '90.1.2.3:54438', 'timeout_wait': 90, 'caller_local': '90.1.2.3:54436', 'callee_packets': 0}], 'duration': 28, 'to_uri': '[email protected]', 'from_uri': '[email protected]', 'callee_ua': 'ROL B2BUA', 'caller_ua': 'Cisco-SIPGateway/IOS-12.x'} 13:51:20 testproxy media-dispatcher[1697]: mediaproxy.interfaces.opensips.UNIXSocketProtocol starting on '/var/run/mediaproxy/opensips_01.sock' Any idea where to look for this timeout? Kind regards, Thomas Gelf _______________________________________________ Users mailing list [email protected] http://lists.opensips.org/cgi-bin/mailman/listinfo/users
