[Freeswitch-users] SRTP issue with latest master + (possible) DTMF change

[Privus 007]

Ah, I see. Thank you both


On Fri, Nov 15, 2013 at 1:02 AM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> The var names are rtp_ for those instead of sip_ now.
> On Nov 14, 2013 4:46 PM, "Privus 007" <privus007 at gmail.com> wrote:
>
>> Hi,
>>
>> I've been using FS successfully for some years now. Recently I decided to
>> update from 1.2.12 to latest master via git (running FS on bare metal
>> Debian 7.0 64bit)
>>
>> Obviously I saved my conf directory and tried to apply it to the
>> 1.5.7b+git~20131114 version I just installed and have up and running.
>>
>> I realize that master is not yet stable but I notice that there seems to
>> be some incompatibility issues, namely with SRTP.
>>
>> All my SRTP calls are now failing with "incompatible destination"
>> messages in the logs, and looking through them more closely I see this:
>>
>>
>> parsing [features->is_secure] continue=true
>> Dialplan: sofia/external/1000 at mydomain Regex (PASS) [is_secure]
>> ${sip_via_protocol}(tls) =~ /tls/ break=on-false
>> Dialplan: sofia/external/1000 at mydomain Regex (FAIL) [is_secure]
>> ${sip_secure_media_confirmed}() =~ /^true$/ break=on-false
>> Dialplan: sofia/external/1000 at mydomain ANTI-Action eval(not_secure)
>> 2013-11-14 22:02:22.006273 [NOTICE] switch_core_session.c:2940 Execute
>> eval(not_secure)
>>
>>
>> Notice the FAIL for sip_secure_media_confirmed. This is very strange
>> since I'm sure that SRTP is enabled (both CSipSimple Android client and
>> Groundwire iOS client confirm that indeed the signalling is secured via TLS
>> and the media via SDES SRTP).
>> A few seconds earlier in the logs, FS also sees the crypto taking place
>> and there doesn't seem to be any problem
>>
>> 2013-11-14 22:02:21.986279 [INFO] switch_rtp.c:2830 Activating Audio
>> Secure RTP SEND
>> 2013-11-14 22:02:21.986279 [DEBUG] switch_core_sqldb.c:2354 Secure Type:
>> srtp:sdes:AES_CM_128_HMAC_SHA1_80
>>
>> So the problem seems to be with the sip_secure_media_confirmed variable.
>> This same setup worked fine yesterday with 1.2.12, so I'm at a loss as to
>> what changed.
>> Any ideas? To further add some confusion, since my clients are configured
>> for mandatory SRTP, all calls to them fail, and FS routes to VM. So far,
>> pretty normal. Except that the VM message we usually hear sounds super slow
>> like the voice is drunk. This is definetely not normal, but I'm not sure if
>> it's related to the crypto issue. I don't think it's a flite issue since
>> calling into the IVR sounds as normal as ever.
>>
>> Also, I notice a change in FS handling DTMF. My CSipSimple client which
>> worked flawlessly with DTMF before now just doesn't work at all, but my
>> Groundwire client continues to send DTMF without a problem.
>> Has something changed in the latest master regarding DTMF?
>>
>> I'd appreciate any help in debugging these issues. Perhaps the new conf
>> in latest master has different variables or options and by simply copying
>> my old conf directory over the new one wasn't too smart after all.
>>
>> Thanks
>>
>> _________________________________________________________________________
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>> http://www.freeswitchsolutions.com
>>
>> 
>> 
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> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
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> http://www.freeswitchsolutions.com
>
> 
> 
>
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>
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