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@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 Internet: www.telefaks.de
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-- With kind regards Peter Steinbach Telefaks Services GmbH mailto:lists (att) telefaks.de Internet: www.telefaks.de
_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting@freeswitch.org
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Brian
West
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Twitter:
@FreeSWITCH , @briankwest
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T:+19184209001
| F:+19184209002 | M:+1918424WEST (9378)
iNUM:+883 5100 1420 9001 | ISN:410*543 | Skype:briankwest
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org