[Freeswitch-users] How can I trigger a ZRTP connection on an existing call?

Peter Steinbach lists at telefaks.de
Wed Feb 11 16:19:28 MSK 2015


concerning:

>your client or FS will have to trigger the start by doing the hello,
which isn't what you want.

Do you see a way for re-establishing the call in a way that ZRTP may be
triggered on the client's side? Is there a way to re-establish a call
beyond a SIP reinvite? Or maybe force some new codecs to be negociated?

Best regards
Peter



On 02/10/15 20:59, Brian West wrote:
> ZRTP happens in the media stream, what you're seeing is a zid in the
> sdp, which doesn't mean that it will start the ZRTP exchange, your
> client or FS will have to trigger the start by doing the hello, which
> isn't what you want.
>
> On Tue, Feb 10, 2015 at 11:58 AM, Peter Steinbach <lists at telefaks.de
> <mailto:lists at telefaks.de>> wrote:
>
>     Hello Brian,
>
>     thanks for the hint. In the meantime, I tried both, setting zrtp
>     globally in Freeswitch and having zrtp in the original dialplan
>     and in the {}-Variables for dialing the B party. This did not help.
>
>     Then I totally disabled ZRTP inside FSs and in the dialplans and
>     grepped on the network. Then I issued a "uuid_media off" for
>     starting a reinvite between the 2 endpoints. On the Freeswitch
>     console, I could see, that ZRTP is exchanged in the SDP, although
>     Freeswitch does a reinvite with ZRTP.
>     Having a look at the channel variables, I could see that
>
>       * variable_rtp_local_sdp_str does _not_ contain ZRTP
>       * variable_switch_m_sdp _does_ contain ZRTP
>       * variable_switch_r_sdp _does_ contain ZRTP
>
>     But ZRTP however is not started on both sides. Then I killed
>     Freeswitch during a call and started again, the calls were
>     automatically recovered, and ZRTP was there. But this was surely
>     done with a normal invite, right?
>
>     I tried with CsipSimple on Android and Jitsi nightly on Ubuntu.
>     Both clients behave the same. I assume, that they will not start
>     ZRTP on a reinvite.
>
>     So unless there is now way to reistablish a call (I tried
>     uuid_transfer to the same number with autoanswer, but target
>     number was still considered as busy), I think it will not work.
>
>     Any other hints?
>
>     Best regards
>     Peter
>
>
>
>
>     On 02/10/15 16:05, Brian West wrote:
>>     You can't set the zrtp_secure_media=true after a session is
>>     created, it needs to be a global or you set it inside the {} on
>>     the bridge or originate line.    Then you're outlining uuid_media
>>     off, which would make the media go between the two endpoints,
>>     FreeSWITCH isn't even involved in the zrtp exchange at that
>>     point.  Technically your endpoints should be the ones doing this
>>     and ZRTP should be disabled in FreeSWITCH, You really only need
>>     ZRTP with MiTM if you MUSt transcode or you're terminating
>>     directly on the FreeSWITCH instance.   I should really just rip
>>     out MiTM support.
>>
>>     On Tue, Feb 10, 2015 at 5:53 AM, Peter Steinbach
>>     <lists at telefaks.de <mailto:lists at telefaks.de>> wrote:
>>
>>         Hello,
>>
>>         I want to establish ZRTP, after I am bridging 2 calls, which
>>         have been active already before.
>>
>>         My scenario is as follows:
>>
>>           * Caller A is calling Freeswitch, the call is answered and
>>             an announcement is played
>>           * In the meantime Freeswitch calls another Party B and
>>             connects it to an internal IVR, where also announcements
>>             are played.
>>           * Now, after a while, I decide to connect the calls
>>             together, then send an "uuid_media off <uuid>" to the
>>             calling party A leg, reinvites occur and bypass media works.
>>           * However the 2 SIP clients (Csipsimple) do not start a
>>             ZRTP session
>>           * If caller A calls B directly via bypass media, the ZRTP
>>             is active, by the way.
>>
>>         I suspect that ZRTP somehow should be forced or triggered. Is
>>         there a way to do that? I already invited the B leg with
>>         "zrtp_secure_media=true" and issued
>>
>>           uuid_setvar 1bd508af-19c7-4e38-8621-731baaf33920
>>         zrtp_secure_media true
>>           uuid_media off 1bd508af-19c7-4e38-8621-731baaf33920
>>         to the A leg. But this did not work.
>>
>>         The main question should be: How can I trigger a ZRTP
>>         connection on an existing call?
>>
>>
>>         -- 
>>         With kind regards
>>         Peter Steinbach 
>>
>>         Telefaks Services GmbH
>>         mailto:lists (att) telefaks.de <http://telefaks.de>
>>         Internet: www.telefaks.de <http://www.telefaks.de>
>>
>>
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>>
>>
>>     -- 
>>
>>     */Brian West/*
>>     brian at freeswitch.org <mailto:brian at freeswitch.org>
>>
>>
>>     */Twitter: @FreeSWITCH , @briankwest/*
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>>
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>>
>>
>>     _________________________________________________________________________
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>>     http://www.freeswitchsolutions.com
>>
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>
>
>     -- 
>     With kind regards
>     Peter Steinbach 
>
>     Telefaks Services GmbH
>     mailto:lists (att) telefaks.de <http://telefaks.de>
>     Internet: www.telefaks.de <http://www.telefaks.de>
>
>
>     _________________________________________________________________________
>     Professional FreeSWITCH Consulting Services:
>     consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>     http://www.freeswitchsolutions.com
>
>     Official FreeSWITCH Sites
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>     http://confluence.freeswitch.org
>     http://www.cluecon.com
>
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>
>
>
> -- 
>
> */Brian West/*
> brian at freeswitch.org <mailto:brian at freeswitch.org>
>
>
> */Twitter: @FreeSWITCH , @briankwest/*
> http://www.freeswitchbook.com
> http://www.freeswitchcookbook.com
>
> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
>
>
>
> _________________________________________________________________________
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>
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-- 
With kind regards
Peter Steinbach 

Telefaks Services GmbH
mailto:lists (att) telefaks.de
Internet: www.telefaks.de

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