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

[Emrah]

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 <mailto: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: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 <sip:53178246 at 5.6.7.8:51132>>;transport=tls
>> Max-Forwards: 70
>> From: <sip:mod_sofia at 1.2.3.4:5060 <sip:mod_sofia at 1.2.3.4:5060>>;tag=Q6XDFHeUUrcHD
>> To: <sip:user at domain.com <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 <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 <mailto: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 <mailto: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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