Werner, Please let me know if there are any other tests I can undertake to help pinpoint the issue. Thanks On Sat, Jun 8, 2013 at 12:04 PM, Privus 007 <privus007 at gmail.com> wrote: > I just realised my previous logcat logs are pretty sparse. > Here's a better capture from another ZRTP call that quickly > dropped/crashed CSip with the relevant info from right before the crash. > Notice that PJSIP's log was cutoff in mid logging activity. The relevant > Android log (full debug this time) is here <http://pastebin.com/0C5LpzKX>. > Notice the "D/SIP SRV (1694): Stop sip stack" message. Something is > definitely wrong here. > > Also, I had a SDES/SRTP call running with the exact same endpoints and > network last night, with no problems whatsoever for over 15 minutes. > This, to me, seems to strongly indicate there is some bug or integration > issue between PJSIP/CSipSimple and Werner's ZRTP implementation. > I'm pretty certain by now there is a real issue here. I hope others can > test and replicate this. You need to let the ZRTP call run for some time to > observe this behaviour, because it doesn't always crash immediately. Just > yesterday I managed to hold a successful ZRTP call for almost 6 minutes > before it inevitably crashed. > > > 22:12:24.114 zrtp_android.c !ZRTP warning message: Dropping packet because > SRTP replay check failed! > 22:12:24.683 zrtp_android.c ZRTP warning message: Dropping packet because > SRTP replay check failed! > 22:12:25.144 zrtp_android.c !ZRTP warning message: Dropping packet because > SRTP replay check failed! > 22:12:27.844 zrtp_android.c !ZRTP warning message: Dropping packet because > SRTP replay check failed! > 22:12:28.876 ec0x738859f0 !Underflow, buf_cnt=197, will generate 1 frame > 22:12:35.045 zrtp_android.c !ZRTP warning message: Dropping packet because > SRTP replay check failed! > 22 > > > On Fri, Jun 7, 2013 at 9:48 PM, Privus 007 <privus007 at gmail.com> wrote: > >> 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/20130610/3c6b57c8/attachment-0001.html>
