[Freeswitch-users] FreeSwith does not setup secure call with leg B.

[Daniel Ivanov]

First of all, srtp is not end-to-end, if you are looking for end to end,
you should look over the zrtp_* varset.
And also look over whether you clients are configured to use SRTP or ZRTP.
10 сент. 2014 г. 15:50 пользователь "Michael Jerris" <mike at jerris.com>
написал:

> Does it work on the latest 1.4 release with the config change he mentioned?
>
> On Sep 10, 2014, at 1:48 AM, Chandrakant Marathe <cwmarathe at gmail.com>
> wrote:
>
> Sorry Brian, I should have told the version. My bad.
> I am using 1.2 stable release. When I hit "git branch -av", it outputs -
>
> * v1.2.stable      2b62885 fs_cli: fix compiler error on CentOS 6 caused
> by recent short uuid logging change
>
> And from FreeSwitch console, when I hit "version" command, it shows -
>
> FreeSWITCH Version 1.2.24+git~20140630T213113Z~2b62885f21~32bit (git
> 2b62885 2014-06-30 21:31:13Z 32bit)
>
> I went through "vars.xml" and found only one config related with
> "rtp_secure_media" and that to related with zrtp
>
> <X-PRE-PROCESS cmd="set" data="zrtp_secure_media=true"/>
>
> Is there any more configuration to do in v1.2 so as to enable end-to-end
> secure call or shall I move to v1.4 beta?
>
> Thanks.
> --
> CWM
>
>
> On Tue, Sep 9, 2014 at 9:04 PM, Brian West <brian at freeswitch.org> wrote:
>
>> You didn't mention what rev you're using, If you're in 1.4 then its
>> rtp_secure_media, please see latest configs and extensive docs in vars.xml
>> about srtp and all the nice knobs you can use to tweak it.
>>
>>
>> On Tue, Sep 9, 2014 at 12:52 AM, Chandrakant Marathe <cwmarathe at gmail.com
>> > wrote:
>>
>>> Hi All,
>>> I have setup FreeSwith PBX. I am facing an issue of not having end to
>>> end secure call. Caller sends INVITE request with SIPS in request URI and
>>> RTP/SAVP in SDP. But when FreeSwith forwards the request to caller, it is
>>> not using RTP/SAVP in SDP.
>>>
>>> I have followed the steps mentioned in WIKI (
>>> https://wiki.freeswitch.org/wiki/SIP_TLS#Configuration) for TLS
>>> configuration and marked "exports sip_secure_media" true in
>>> "conf/dialplan/default.xml" file (as per
>>> https://wiki.freeswitch.org/wiki/SRTP). But still FreeSwitch does not
>>> use RTP/SAVP for leg-B.
>>>
>>> Also, I am bit confused with following condition in "default.xml" when
>>> wiki page suggests that
>>> late coded negotiation must NOT to be turned on.
>>>
>>>     <!--
>>>          Since we have inbound-late-negotation on by default now the
>>>          above behavior isn't the same so you have to do one extra step.
>>>         -->
>>>
>>>       <condition field="${endpoint_disposition}" expression="^(DELAYED
>>> NEGOTIATION)"/>
>>>       <condition field="${switch_r_sdp}"
>>> expression="(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)"
>>> break="never">
>>>         <action application="set" data="sip_secure_media=true"/>
>>>         <!-- Offer SRTP on outbound legs if we have it on inbound. -->
>>>         <action application="export" data="sip_secure_media=true"/>
>>>       </condition>
>>>
>>> By commenting/un-commenting "sip_secure_media=true" above, it did not
>>> worked.
>>>
>>> Any help with this would be greatly appreciated.
>>>
>>> --
>>> Thanks
>>> CWM
>>>
>>> _________________________________________________________________________
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>>
>>
>>
>> --
>>
>> *Brian West*
>> brian at freeswitch.org
>>
>>
>> *Twitter: @FreeSWITCH , @briankwest*
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>>
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>>
>> _________________________________________________________________________
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>>
>> Official FreeSWITCH Sites
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>>
>> 
>> 
>>
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>
> _________________________________________________________________________
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>
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>
> 
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>
>
> _________________________________________________________________________
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>
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>
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