Hi Werner, I've done some more testing, and the strange behaviour continues. This time I tried from a Jitsi client calling to an Android 4.2.2 running on 3G Indeed it seems like CSip crashes, although it may just be dropping the registration and re-registering. It all happens very fast. As you can see, the logging stops suddenly when the call drops/csip crashes. I've uploaded the PJSIP logs here <http://pastebin.com/0xfVN2QN>and also the corresponding logcat logs here <http://pastebin.com/WRzrF1sg>. Earlier today (not in the logs I'm posting) I managed to have a 6 minute conversation before the inevitable happened (and the logs do show the SRTP replay error messages). But usually the conversation never even reaches 1 minute. I hope you can make some sense of this. Thanks On Fri, Jun 7, 2013 at 6:54 AM, Werner Dittmann <Werner.Dittmann at t-online.de > wrote: > Hmmm - this looks suspicious :-) . An automatic re-registration could mean > that CSipSimple automatically restarts. I know of another VoIP client that > performs an auto-restart in case it crashes. Such a restart happens quite > fast and is often barely noticeable. > > I cannot reproduce the problem here, thus can you check/see if CSipSimple > just crashes and performs a automatic restart? If this is the case than we > can try to get the Android logfies from the devices and analyse them. > > Werner > > Am 06.06.2013 19:43, schrieb Privus 007: > > *The available log does not show the "call drop", the log suddenly stops > in > > the midst > > of another warning message. Any info about this? What does it mean "drops > > the connection?" > > RTP/SRTP does not have a connection per se, it's UDP thus CS cannot "drop > > it". > > Does it drop the SIP connection (you use TLS and this CS must hold a TCP > > connection)? A re-registration is only necessary if either the register > > timer triggers or if the SIP TLS(TCP) connection is lost. > > > > * > > Correct, I use TLS (TCP) for the SIP signalling and RTP/SRTP for the > media. > > What I mean by the "connection drops" is that the call suddenly ends in > mid > > call and I then see CSip immediately re-registering on the server. > > Since I don't see any TLS errors in the logs, and only the SRTP replay > > error, I think the issue lies with ZRTP. Plus, as I said, SDES SRTP calls > > (also using TLS) never suddenly "drop" nor does CSip suddenly > re-register, > > which I think tends to rule out TLS as the problem. > > Still, I'm going to make a few more calls with the fullOpenSSL build and > > I'll upload the logs to pastebin.* > > * > > > > > > On Thu, Jun 6, 2013 at 4:33 PM, Werner Dittmann < > Werner.Dittmann at t-online.de > >> wrote: > > > >> Am 06.06.2013 16:46, schrieb Privus 007: > >>> *So some questions: > >>> - is the audio completely disturbed during the call? Or what's the > > <SNIP ---- SNAP> > > -- > ---------------------------------------------- > Werner Dittmann Werner.Dittmann at t-online.de > Tel +49 173 44 37 659 > PGP key: 82EF5E8B > > _______________________________________________ > Visit our blog: http://blog.pjsip.org > > pjsip mailing list > pjsip at lists.pjsip.org > http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/attachments/20130607/a8b7bb56/attachment-0001.html>
