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

Fri Nov 15 19:08:19 MSK 2013

that var is now split into

rtp_secure_audio_confirmed
rtp_secure_video_confirmed

so in your case, change media to audio.

On Fri, Nov 15, 2013 at 7:44 AM, Privus 007 <privus007 at gmail.com> wrote:

> Ok, so I applied the changes in the dialplan and still there's a crypto
> problem.
> I actually decided to put aside my previous diaplan and am testing with
> the default one provided by master.
>
> Now indeed I see that crypto gets properly detected in the default
> dialplan, and it gets exported to b leg:
>
> Dialplan: sofia/external/1010 at mydomain Regex (PASS) [global] ${rtp_has_crypto}(AES_CM_128_HMAC_SHA1_80) =~ /^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$/ break=never
> Dialplan: sofia/external/1010 at mydomain Action set(rtp_secure_media=true)
> Dialplan: sofia/external/1010 at mydomain Action export(rtp_secure_media=true)
>
>
> But then I see it still doesn't pass the rtp_secure_media_confirmed check.
>
> Dialplan: sofia/external/1010 at mydomain parsing [features->is_secure] continue=true
>
> Dialplan: sofia/external/1010 at mydomain Regex (PASS) [is_secure] ${sip_via_protocol}(tls) =~ /tls/ break=on-false
> Dialplan: sofia/external/1010 at mydomain Regex (FAIL) [is_secure] ${rtp_secure_media_confirmed}() =~ /^true$/ break=on-false
>
> Dialplan: sofia/external/1010 at mydomain ANTI-Action eval(not_secure)
> 2013-11-15 13:33:38.386273 [NOTICE] switch_core_session.c:2940 Execute eval(not_secure)
> EXECUTE sofia/external/1010 at mydomain eval(not_secure)
>
>
> So what am I doing wrong? Why is it failing rtp_secure_media_confirmed? Should I change it to plain rtp_secure_media and leave out the "confirmed" bit, or do I have to change var names somewhere else besides the default and features dialplan?
>
>
> Thanks
>
>
>
> On Fri, Nov 15, 2013 at 1:36 AM, Privus 007 <privus007 at gmail.com> wrote:
>
>> 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
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
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>>> _________________________________________________________________________
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>>>
>>> 
>>> 
>>>
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>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
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>
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-- 
Anthony Minessale II

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