[jitsi-dev] outgoing call fails once (initiating call)


#1

Hi,

I'd like to report an issue that other users on this mailing list already experienced.

An idle Jisti client (no active calls for several minutes) tried to dial an extension via LAN SIP server (Asterisk) but stayed in the "calling state" (ie. it was displaying the message defined by service.gui.INITIATING_CALL_STATUS). The user could not even hear the ring tone. There was complete silence for several seconds until it hung up with an error (I don't remember the exact error message).
After this type of error happens the Jitsi user usually can successfully dial the destination immediately.

Maybe it can be related to sip registration timeout values or something asterisk-specific, I don't know yet. However, maybe the Jitsi log could be more verbose when initiating calls so we can see at what point of the code it's waiting so long to establish the call.

I'm pasting the last lines of the log but there's nothing regarding the "initiating outgoing call" steps. I'm supposing it happened around 30 seconds before 12:23 when impl.protocol.sip.CallPeerSipImpl.hangup() was called.
The stuff at 12:10 has nothing to do with this issue but I'm pasting it so you can see when it last logged something.

By the way, it's not a DNS name resolution timeout issue because I'm specifying a LAN IP address for the Asterisk SIP server.

12:10:36.458 INFO: [5191] service.protocol.media.MediaHandler.registerDynamicPTsWithStream().966 Dynamic PT map: 96=rtpmap:-1 opus/48000 fmtp:usedtx=1; 97=rtpmap:-1 iLBC/8000; 99=rtpmap:-1 H264/90000 fmtp:profile-level-id=4DE01f; 98=rtpmap:-1 H264/90000 fmtp:profile-level-id=4DE01f;packetization-mode=1;
12:10:36.458 INFO: [5191] service.protocol.media.MediaHandler.registerDynamicPTsWithStream().983 PT overrides []
12:10:36.626 INFO: [5215] net.sf.fmj.media.Log.info() Starting RTPSourceStream.
12:10:36.658 INFO: [5191] org.jitsi.impl.neomedia.MediaStreamImpl.info() audio codec/freq: GSM/8000 Hz
12:10:36.658 INFO: [5191] org.jitsi.impl.neomedia.MediaStreamImpl.info() audio remote IP/port: 10.215.147.112/19696
12:10:36.658 INFO: [5191] net.sf.fmj.media.Log.info() Starting RTPSourceStream.
12:10:37.669 INFO: [5249] org.jitsi.impl.neomedia.MediaStreamImpl.info()
Receive stream stats: discarded RTP packets: 7
Receive stream stats: decoded with FEC: 0
12:10:37.670 INFO: [5249] net.sf.fmj.media.Log.info() Stopping RTPSourceStream.
12:10:37.672 INFO: [5249] org.jitsi.impl.neomedia.MediaStreamImpl.info() rtpstat:call stats for outgoing audio stream SSRC:-2021763546
rtpstat:bytes sent: 1650
rtpstat:RTP sent: 50
rtpstat:remote reported min interarrival jitter : -1
rtpstat:remote reported max interarrival jitter : 0
rtpstat:local collisions: 0
rtpstat:remote collisions: 0
rtpstat:RTCP sent: 0
rtpstat:transmit failed: 0
12:10:37.673 INFO: [5249] org.jitsi.impl.neomedia.MediaStreamImpl.info() rtpstat:call stats for incoming rtpmap:3 GSM/8000 stream SSRC:1809848788
rtpstat:packets received: 61
rtpstat:bytes received: 2745
rtpstat:packets lost: 0
rtpstat:min interarrival jitter : -1
rtpstat:max interarrival jitter : 0
rtpstat:RTCPs received: 0
rtpstat:bad RTCP packets: 0
rtpstat:bad RTP packets: 0
rtpstat:local collisions: 0
rtpstat:malformed BYEs: 0
rtpstat:malformed RRs: 0
rtpstat:malformed SDESs: 0
rtpstat:malformed SRs: 0
rtpstat:packets looped: 0
rtpstat:remote collisions: 0
rtpstat:SRRs received: 0
rtpstat:transmit failed: 0
rtpstat:unknown types: 0
12:10:37.673 INFO: [5249] net.sf.fmj.media.Log.info() net.sf.fmj.media.rtp.RTPSourceStream Total packets added: 61
12:10:37.674 INFO: [5249] net.sf.fmj.media.Log.info() net.sf.fmj.media.rtp.RTPSourceStream Times reset() called: 0
12:10:37.674 INFO: [5249] net.sf.fmj.media.Log.info() net.sf.fmj.media.rtp.RTPSourceStream Times grow() called: 0
12:10:37.674 INFO: [5249] net.sf.fmj.media.Log.info() net.sf.fmj.media.rtp.RTPSourceStream Times shrink() called: 0
12:10:37.674 INFO: [5249] net.sf.fmj.media.Log.info() net.sf.fmj.media.rtp.RTPSourceStream Times read() while empty:0
12:10:37.675 INFO: [5249] net.sf.fmj.media.Log.info() net.sf.fmj.media.rtp.RTPSourceStream Packets dropped because full: 7
12:10:37.675 INFO: [5249] net.sf.fmj.media.Log.info() net.sf.fmj.media.rtp.RTPSourceStream Packets dropped while shrinking: 0
12:10:37.675 INFO: [5249] net.sf.fmj.media.Log.info() net.sf.fmj.media.rtp.RTPSourceStream Packets dropped because they were late: 0
12:10:37.675 INFO: [5249] net.sf.fmj.media.Log.info() net.sf.fmj.media.rtp.RTPSourceStream Packets dropped because they were late by more than MAX_SIZE: 0
12:10:37.675 INFO: [5249] net.sf.fmj.media.Log.info() net.sf.fmj.media.rtp.RTPSourceStream Packets dropped in reset(): 0
12:10:37.676 INFO: [5249] net.sf.fmj.media.Log.info() net.sf.fmj.media.rtp.RTPSourceStream Max size reached: 4
12:10:37.676 INFO: [5249] net.sf.fmj.media.Log.info() net.sf.fmj.media.rtp.RTPSourceStream Adaptive jitter buffer mode was enabled
12:10:37.676 INFO: [5249] net.sf.fmj.media.Log.info() Stopping RTPSourceStream.
12:10:37.696 INFO: [5249] net.sf.fmj.media.Log.info() Stopping RTPSourceStream.
12:10:37.698 INFO: [5238] net.sf.fmj.media.Log.info() Stopping RTPSourceStream.
12:10:37.699 INFO: [5249] net.sf.fmj.media.Log.info() Stopping RTPSourceStream.
12:23:30.723 SEVERE: [36] impl.protocol.sip.CallPeerSipImpl.hangup().1049 Could not determine call peer state!

Thanks,

Vieri