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What is the verison of the app you are using?
Recently latest ZRTP version has been added to support in zrtpcpp/zrtp4pj and
integrated into csipsimple.
Original comment by r3gis...@gmail.com
on 30 Mar 2013 at 7:16
r2180, r2174, r2170, ...
I tried also with phoner lite on Win8, ZRTP did not work.
Original comment by hvtaifwk...@gmail.com
on 31 Mar 2013 at 11:06
The change I mentioned was introduced in r2169
(http://code.google.com/p/csipsimple/source/detail?r=2169)
I'm not sure to understand from your post if you tried one before r2169.
Original comment by r3gis...@gmail.com
on 31 Mar 2013 at 12:06
I have used also the Google Play version and IIRC r2165 or r2163
Original comment by hvtaifwk...@gmail.com
on 31 Mar 2013 at 12:17
Ok thanks, so it's not a problem with this last change.
So, I've no idea what could be the root cause. :/
The problem is only with linphone? It seems you already tried with other
softphones, so I guess yes (maybe you can try with twinkle that was one of the
first to support zrtp).
In logs what seems to be the root problem is : "RTP decode error: Invalid RTP
version (PJMEDIA_RTP_EINVER) [err:220122]". It could come if csipsimple is
configured to enable SRTP(with key exch on sdp) and ZRTP at the same time. But
I guess you take care to check that already. So there is maybe something on
linphone side that put pjsip in a mode that bypass zrtp transport adapter...
Original comment by r3gis...@gmail.com
on 31 Mar 2013 at 12:33
I try with any softphone I can get to compile/work on Linux.
Now I tried with Twinkle, when it answered call from CSipSimple, ZRTP did not
get enabled and it crashed with segmentation fault when I hanged up. And it
did not manage to make a call to CSipSimple (Send Internal: 404).
SRTP is disabled in CS.
With what software have you managed to get ZRTP working (calling both ways)
with CS?
Original comment by hvtaifwk...@gmail.com
on 31 Mar 2013 at 1:35
I do my unit tests with twinkle and no problem so far (I just did a new one
just now with css calling and with twinkle calling and no problems).
So I guess there is something else.
Are you trying peer to peer calls? Or using a sip server in the middle? Maybe I
can have a look to your full logs (see HowToCollectLogs). There is maybe
something that will ring a bell. (For example if SIP ACK never arrives maybe
there is some network topology vs configuration problem... for example if stun
is enabled but trying to call on local network only).
Original comment by r3gis...@gmail.com
on 31 Mar 2013 at 2:16
With sflphone ZRTP works both ways, but nothing is heard from the CS speaker.
Then CS crashed with segfault
17:35:56.425 strm0x134ddb4 RTP status: badpt=0, badssrc=0, dup=0,
outorder=-1, probation=-1, restart=0
17:35:56.445 Master/sound !Underflow, buf_cnt=0, will generate 1 frame
17:35:56.446 Master/sound Underflow, buf_cnt=0, will generate 1 frame
[dmesg]
<3>[03-31 14:35:56.460] [audio_pcm_in.c:audpcm_in_get_dsp_frames] Error! not
able to keep up the read
17:35:56.468 Master/sound Underflow, buf_cnt=0, will generate 1 frame
17:35:56.474 strm0x134ddb4 !RTP status: badpt=0, badssrc=0, dup=0,
outorder=-1, probation=-1, restart=0
17:35:56.474 strm0x134ddb4 RTP status: badpt=0, badssrc=0, dup=0,
outorder=-1, probation=-1, restart=0
17:35:56.474 strm0x134ddb4 RTP status: badpt=0, badssrc=0, dup=0,
outorder=-1, probation=-1, restart=0
17:35:56.478 Master/sound !456 samples reduced, buf_cnt=2104
17:35:56.478 Master/sound 326 samples reduced, buf_cnt=2098
17:35:56.478 Master/sound 354 samples reduced, buf_cnt=2064
...
17:35:56.654 strm0x134ddb4 RTP status: badpt=0, badssrc=0, dup=0,
outorder=-1, probation=-1, restart=0
17:35:56.664 strm0x134ddb4 RTP status: badpt=0, badssrc=0, dup=0,
outorder=-1, probation=-1, restart=0
17:35:
log ended here and CS started new log file..
There's always some SIP server in the way, yeah. I'd prefer not to.
I have tried quite a many, iptel.org seems to work most reliably.
Original comment by hvtaifwk...@gmail.com
on 31 Mar 2013 at 3:36
I would be very very interested by the logs of the crash (*Using the
HowToCollectLogs instructions* : it captures more than the pjsip log file). If
you can send me the crash logs by mail it will be very valuable for me because
the crash logs indicates exactly where in the native stack it crashes.
If you want to test without sip server in the middle it's pretty easy if your
pc and your smartphone are on same network. (That's how I do many of my tests
when I want to sort out all NAT problems)
So to test that :
In csipsimple create a "Local" account.
There is probably the equivalent on other desktop softphone (on twinkle just
create an account with as domain the lan ip of your pc).
Then you can call from one device to the other by dialing sip:user@192.168.x.x
(user doesn't matter really here, it can also be omitted depending on the sip
softphone and 192.168.x.x is the ip of the other side).
Original comment by r3gis...@gmail.com
on 31 Mar 2013 at 3:57
I was wondering if the experimental video is also encrypted with zrtp? I guess
not cause only one SAS is shown. Are there any plans to encrypt and show both?
Original comment by felix.kn...@googlemail.com
on 3 Nov 2013 at 11:12
No experimental video feature is not yet encrypted with ZRTP. There is some
work to do in order to use multistream feature of lib ZRTPCpp.
Original comment by r3gis...@gmail.com
on 4 Nov 2013 at 8:34
Original comment by r3gis...@gmail.com
on 22 Jun 2015 at 11:30
Original issue reported on code.google.com by
wheresau...@lavabit.com
on 5 Oct 2010 at 10:52