[Freeswitch-users] Random calls failing with WRONG_CALL_STATe when using TLS

Brian West brian at freeswitch.org
Mon Feb 16 16:44:42 MSK 2015


Via: SIP/2.0/TLS 1.2.3.4:443;branch=z9hG4bK6Kv171Q3U5rrD

Your issue is the contact has no port 443 or transport=tls right?  What
sits between FS and the endpoint?

On Sun, Feb 15, 2015 at 5:38 AM, Emrah <lists at kavun.ch> wrote:

> Thanks Ken. Is there a way to filter the SIP trace? It's a busy box.
>
> On Feb 14, 2015, at 3:35 AM, Ken Rice <krice at freeswitch.org> wrote:
>
> Open a jire with a full debug login including sip tracing on
>
> Sent from my iPhone
>
> On Feb 13, 2015, at 7:57 PM, Emrah <lists at kavun.ch> wrote:
>
> Hi,
> The issue is persistent. I am curious to know if anyone else on the list
> is experiencing this. It doesn't seem to have been reported before.
> Should I dedicate a profile to TLS use only?
> I also posted a message on the list about receiving options packet with
> the wrong transport. Are these 2 issues connected? Here is a copy paste of
> my message:
>
> My experience with FS and TLS has been rather mixed so far. It's been a
> little inconsistent in keeping NAT sessions up and users discoverable.
> One thing I've noticed is that FS advertises the wrong information in
> option packets. The following is what I receive over my TLS session which
> is working on port 443.
> 1.2.3.4:443 -(SIP over TLS)-> 10.0.0.99:51132
> OPTIONS sip:53178246 at 10.0.0.99:56494;transport=tls;received=5.6.7.8:51132
>  SIP/2.0
> Via: SIP/2.0/TLS 1.2.3.4:443;branch=z9hG4bK6Kv171Q3U5rrD
> Route: <sip:53178246 at 5.6.7.8:51132>;transport=tls
> Max-Forwards: 70
> From: <sip:mod_sofia at 1.2.3.4:5060>;tag=Q6XDFHeUUrcHD
> To: <sip:user at domain.com>
> Call-ID:
> 0a052f23-34a8-4158-8c88-fd2a70ffb561_c2RhaSoOYBR6jfJe4ndLoTTKJMrO2gMv
> CSeq: 71498568 OPTIONS
> Contact: <sip:mod_sofia at 1.2.3.4:5060>
> User-Agent: FreeSWITCH
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER,
> REFER, NOTIFY, PUBLISH, SUBSCRIBE
> Supported: timer, path, replaces
> Allow-Events: talk, hold, conference, presence, as-feature-event, dialog,
> line-seize, call-info, sla, include-session-description, presence.winfo,
> message-summary, refer
> Content-Length: 0
>
> As you can see FS stamps the packet with a port 5060... No reference to
> port 443 with a transport=tls.
>
> What shall be done?
>
> On Feb 5, 2015, at 3:18 PM, Emrah <lists at kavun.ch> wrote:
>
> Hi there,
> This issue is happening all around with devices using TLS. It's not very
> frequent with softphones, but not inexistant.
> Any pointers would be greatly appreciated. Do you have  best practice
> configs you'd like to share?
>
> Thanks
>
> On Jan 30, 2015, at 6:10 PM, Emrah <lists at kavun.ch> wrote:
>
> Hi all,
> I am facing a very frustrating issue. I often have to dial twice when
> using my Yealink phone with TLS because the first attempt times out.
> The logs on the Yealink indicate that the first invite is successfully
> received, to which my FS sends a 100 trying and 407 proxy auth required. It
> is subsequently when my phone sends back the invite that the connection
> crashes with the following error:
> SSL ERROR SYSCALL
>
> Is this something common? Why does the SSL connection crashes when the
> phone attempts to send the second invite? My phone is behind NAT.
>
> It is going to be a crazy expedition to collect the logs and Pastebin
> them, so I am tempting my luck on the list first to see if you have any
> pointers.
>
> As a last piece, my Bria on my iPHone, among other clients, never had this
> issue. I did experience it from time to time with Blink on Mac OS X.
>
> Any help appreciated.
>
> Emrah
>
>
>
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>
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-- 

*Brian West*
brian at freeswitch.org


*Twitter: @FreeSWITCH , @briankwest*
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