#262 Add support for ZRTP

[Anonymous]

Originally created by: wheresau...@lavabit.com
Originally owned by: r3gis...@gmail.com

Not a bug - enhancement

please add ZRTP support this allows for a more secure handling of handset to handset key exchanges.   Two linux softphones I know of built on pjsip allow SRTP and ZRTP, allow using them at the same time.

other pjsip implementations:
http://www.sflphone.org/
http://twinklephone.com/

[Anonymous]

Originally posted by: r3gis...@gmail.com

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...

[Anonymous]

Originally posted by: hvtaifwk...@gmail.com

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?

[Anonymous]

Originally posted by: r3gis...@gmail.com

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).

[Anonymous]

Originally posted by: hvtaifwk...@gmail.com

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.

[Anonymous]

Originally posted by: r3gis...@gmail.com

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).

[Anonymous]

Originally posted by: felix.kn...@googlemail.com

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?

[Anonymous]

Originally posted by: r3gis...@gmail.com

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.