Hi, I would like to use FS (multiple) as a ZRTP client register on a the main FS. I have already managed to setup a Gateway with user/pass on the FS client and register it on the main FS for regular incoming/outgoing calls (without ZRTP). To enable ZRTP calls I setup the FS client/main as follow: None ZRTP SIP phone -> FS client -> NAT Internet -> FS main -> CSipSimple ZRTP enabled). FS Client (ver 1.4) : ZRTP enabled globally in VARS and in dialplan. Media proxy disabled on both internal and external profiles and in dialplan. FS Main (ver 1.4): Media proxy enabled - including late negotiation. I forced the FS client and the CSipSimple to use only PCMU codec to avoid transcoding. The problem is that on a call from the SIP phone via the FS client there are always CRC errors on the ZRTP log. Only if the FS main is set to disable media proxy then there are no CRC errors - but then it becomes a MITM with incompatible SASs.
From reading other posts about FS and ZRTP CRC errors I assume that it happens because the FS client is not creating a zrtp hash in the invite SDP.
So my question is how to make the FS client to generate the zrtp hash in the invite SDP to act as real ZRTP enabled client? I've already tried to set the FS client internal/external/dialplan with several zrtp configs with no success. I would appreciate any tip to resolve this issue. Regards Assaf The client FS
did you try and see if this is the case with a current release instead of the old one?
On Mar 28, 2016, at 8:14 AM, Assaf Dahary <adahary@gmail.com> wrote:
Hi,
I would like to use FS (multiple) as a ZRTP client register on a the main FS.
I have already managed to setup a Gateway with user/pass on the FS client and register it on the main FS for regular incoming/outgoing calls (without ZRTP).
To enable ZRTP calls I setup the FS client/main as follow: None ZRTP SIP phone -> FS client -> NAT Internet -> FS main -> CSipSimple ZRTP enabled).
FS Client (ver 1.4) : ZRTP enabled globally in VARS and in dialplan. Media proxy disabled on both internal and external profiles and in dialplan.
FS Main (ver 1.4): Media proxy enabled – including late negotiation.
I forced the FS client and the CSipSimple to use only PCMU codec to avoid transcoding.
The problem is that on a call from the SIP phone via the FS client there are always CRC errors on the ZRTP log. Only if the FS main is set to disable media proxy then there are no CRC errors – but then it becomes a MITM with incompatible SASs.
From reading other posts about FS and ZRTP CRC errors I assume that it happens because the FS client is not creating a zrtp hash in the invite SDP.
So my question is how to make the FS client to generate the zrtp hash in the invite SDP to act as real ZRTP enabled client? I've already tried to set the FS client internal/external/dialplan with several zrtp configs with no success.
I would appreciate any tip to resolve this issue.
Regards
Assaf
The client FS
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OK. I'll test it with the latest release 1.6. But do you know if this issue was resolved in 1.6? Assaf On Mar 28, 2016, 18:13, at 18:13, Michael Jerris <mike@jerris.com> wrote:
did you try and see if this is the case with a current release instead of the old one?
On Mar 28, 2016, at 8:14 AM, Assaf Dahary <adahary@gmail.com> wrote:
Hi,
I would like to use FS (multiple) as a ZRTP client register on a the main FS.
I have already managed to setup a Gateway with user/pass on the FS client and register it on the main FS for regular incoming/outgoing calls (without ZRTP).
To enable ZRTP calls I setup the FS client/main as follow: None ZRTP SIP phone -> FS client -> NAT Internet -> FS main -> CSipSimple ZRTP enabled).
FS Client (ver 1.4) : ZRTP enabled globally in VARS and in dialplan. Media proxy disabled on both internal and external profiles and in dialplan.
FS Main (ver 1.4): Media proxy enabled – including late negotiation.
I forced the FS client and the CSipSimple to use only PCMU codec to avoid transcoding.
The problem is that on a call from the SIP phone via the FS client there are always CRC errors on the ZRTP log. Only if the FS main is set to disable media proxy then there are no CRC errors – but then it becomes a MITM with incompatible SASs.
From reading other posts about FS and ZRTP CRC errors I assume that it happens because the FS client is not creating a zrtp hash in the invite SDP.
So my question is how to make the FS client to generate the zrtp hash in the invite SDP to act as real ZRTP enabled client? I've already tried to set the FS client internal/external/dialplan with several zrtp configs with no success.
I would appreciate any tip to resolve this issue.
Regards
Assaf
The client FS
_________________________________________________________________________
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Few notes regarding ZRTP: A. It's an encryption, it's am encryption traversal mechanism that allow supported servers and client to choose to use it. B. The encryption will be SRTP, and it is also good to have SIPS and everything else encrypted. I hope it will help better to understand it. Ido On Mar 28, 2016 6:58 PM, "Assaf Dahary" <adahary@gmail.com> wrote:
OK. I'll test it with the latest release 1.6.
But do you know if this issue was resolved in 1.6?
Assaf On Mar 28, 2016, at 18:13, Michael Jerris <mike@jerris.com> wrote:
did you try and see if this is the case with a current release instead of the old one?
On Mar 28, 2016, at 8:14 AM, Assaf Dahary <adahary@gmail.com> wrote:
Hi,
I would like to use FS (multiple) as a ZRTP client register on a the main FS.
I have already managed to setup a Gateway with user/pass on the FS client and register it on the main FS for regular incoming/outgoing calls (without ZRTP).
To enable ZRTP calls I setup the FS client/main as follow:
None ZRTP SIP phone -> FS client -> NAT Internet -> FS main -> CSipSimple ZRTP enabled).
*FS Client (ver 1.4) :*
ZRTP enabled globally in VARS and in dialplan.
Media proxy disabled on both internal and external profiles and in dialplan.
*FS Main (ver 1.4):*
Media proxy enabled – including late negotiation.
I forced the FS client and the CSipSimple to use only PCMU codec to avoid transcoding.
The problem is that on a call from the SIP phone via the FS client there are always CRC errors on the ZRTP log.
Only if the FS main is set to disable media proxy then there are no CRC errors – but then it becomes a MITM with incompatible SASs.
From reading other posts about FS and ZRTP CRC errors I assume that it happens because the FS client is not creating a zrtp hash in the invite SDP.
So my question is how to make the FS client to generate the zrtp hash in the invite SDP to act as real ZRTP enabled client?
I've already tried to set the FS client internal/external/dialplan with several zrtp configs with no success.
I would appreciate any tip to resolve this issue.
Regards
Assaf
The client FS
_________________________________________________________________________ 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
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Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
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Following Mike's advise, I've compiled the latest FS version 1.6.7 on a Raspi-2 device with the zrtp libs and setup: ./configure --enable-zrtp. I've enabled zrtp in vars.XML and in dialplan and tried to make a regular zrtp call with several zrtp clients but failed to make it happen. The fs_CLI log shows the zrtp hash and there is a statement that zrtp is detected BUT no zrtp session is starting with HELLO or anything else. It looks like the ZRTP module is not active.
From fs_CLI I get zrtp_enabled=false.
I've also recompiled with ver 1.7 with same results. When I'm recompiling back with ver 1.4 then zrtp is working. Is there any issue with fs 1.6.7 on raspi-2 using zrtp? Regards Assaf On Mar 28, 2016, 18:13, at 18:13, Michael Jerris <mike@jerris.com> wrote:
did you try and see if this is the case with a current release instead of the old one?
On Mar 28, 2016, at 8:14 AM, Assaf Dahary <adahary@gmail.com> wrote:
Hi,
I would like to use FS (multiple) as a ZRTP client register on a the main FS.
I have already managed to setup a Gateway with user/pass on the FS client and register it on the main FS for regular incoming/outgoing calls (without ZRTP).
To enable ZRTP calls I setup the FS client/main as follow: None ZRTP SIP phone -> FS client -> NAT Internet -> FS main -> CSipSimple ZRTP enabled).
FS Client (ver 1.4) : ZRTP enabled globally in VARS and in dialplan. Media proxy disabled on both internal and external profiles and in dialplan.
FS Main (ver 1.4): Media proxy enabled – including late negotiation.
I forced the FS client and the CSipSimple to use only PCMU codec to avoid transcoding.
The problem is that on a call from the SIP phone via the FS client there are always CRC errors on the ZRTP log. Only if the FS main is set to disable media proxy then there are no CRC errors – but then it becomes a MITM with incompatible SASs.
From reading other posts about FS and ZRTP CRC errors I assume that it happens because the FS client is not creating a zrtp hash in the invite SDP.
So my question is how to make the FS client to generate the zrtp hash in the invite SDP to act as real ZRTP enabled client? I've already tried to set the FS client internal/external/dialplan with several zrtp configs with no success.
I would appreciate any tip to resolve this issue.
Regards
Assaf
The client FS
_________________________________________________________________________
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I've reinstalled both FS's, client and main, with the latest version 1.6.7. None ZRTP SIP phone -> FS client -> NAT Internet -> FS main -> CSipSimple ZRTP enabled). All set to PCMU codec (no transcoding). All connect on same network (no NAT). The FS client is setup with media-proxy=false (dialplan and profiles). And The FS main is setup with media-proxy=true. I still get the same problem, I get zrtp CRC warning on the FS client - and no end-to-end encryption. As before (with version 1.4) the FS client is NOT sending any zrtp hash, as I suppose it should do. When FS main is set to media-proxy=false then there are no waning as it now acting as MITM. Any tip/advise on the right direction will be very much appreciated. Regards Assaf On Mar 28, 2016, 18:13, at 18:13, Michael Jerris <mike@jerris.com> wrote:
did you try and see if this is the case with a current release instead of the old one?
On Mar 28, 2016, at 8:14 AM, Assaf Dahary <adahary@gmail.com> wrote:
Hi,
I would like to use FS (multiple) as a ZRTP client register on a the main FS.
I have already managed to setup a Gateway with user/pass on the FS client and register it on the main FS for regular incoming/outgoing calls (without ZRTP).
To enable ZRTP calls I setup the FS client/main as follow: None ZRTP SIP phone -> FS client -> NAT Internet -> FS main -> CSipSimple ZRTP enabled).
FS Client (ver 1.4) : ZRTP enabled globally in VARS and in dialplan. Media proxy disabled on both internal and external profiles and in dialplan.
FS Main (ver 1.4): Media proxy enabled – including late negotiation.
I forced the FS client and the CSipSimple to use only PCMU codec to avoid transcoding.
The problem is that on a call from the SIP phone via the FS client there are always CRC errors on the ZRTP log. Only if the FS main is set to disable media proxy then there are no CRC errors – but then it becomes a MITM with incompatible SASs.
From reading other posts about FS and ZRTP CRC errors I assume that it happens because the FS client is not creating a zrtp hash in the invite SDP.
So my question is how to make the FS client to generate the zrtp hash in the invite SDP to act as real ZRTP enabled client? I've already tried to set the FS client internal/external/dialplan with several zrtp configs with no success.
I would appreciate any tip to resolve this issue.
Regards
Assaf
The client FS
_________________________________________________________________________
Professional FreeSWITCH Consulting Services: consulting@freeswitch.org <mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
Official FreeSWITCH Sites http://www.freeswitch.org <http://www.freeswitch.org/> http://confluence.freeswitch.org <http://confluence.freeswitch.org/> http://www.cluecon.com <http://www.cluecon.com/>
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participants (3)
-
Assaf Dahary
-
ik
-
Michael Jerris