[Freeswitch-users] Changing audio DTLS state from HANDSHAKE to FAIL issue

Thu May 18 16:19:03 UTC 2017

I can’t specifically… I know i’ve had luck with sip.js in the past.

> On May 17, 2017, at 12:56 PM, sachin <snehsach at rediffmail.com> wrote:
> 
> Hello Michael, 
> 
> Thanks for the information. Can you please let me know what was the client used? Also I am using self signed certificates on my local LAN. 
> 
> 
> Thanks and Regards
> SD
> 
> 
> 
> 
> 
> From: Michael Jerris <mike at jerris.com>
> Sent: Fri, 12 May 2017 03:58:08 GMT+0530
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Subject: Re: [Freeswitch-users] Changing audio DTLS state from HANDSHAKE to FAIL issue
> 
> we just had confirmed on Jira that 1.6.17 windows is working fine with latest chrome.  If you are getting errors still with something that is other than sipml5, please confirm details of what exactly the issue is.
> 
>> On May 8, 2017, at 8:08 AM, sachin <snehsach at rediffmail.com> wrote:
>> 
>> Hello Michael,
>> 
>> I tried using Sip.js instead of sipml5 and with Linux as the server. I have installed the FreeSWITCH Version 1.9.0+git~20170501T171230Z~e3ef041517~64bit (git e3ef041 2017-05-01 17:12:30Z 64bit)
>> 
>> I am getting INCOMPATIBLE  DESTINATION error. Also tried sipml5.. with bot the client I am getting the same error.
>> 
>> Please let me know what I am missing.
>> 
>> Thanks and Regards,
>> SD
>> 
>> 
>> From: Michael Jerris <mike at jerris.com>
>> Sent: Fri, 05 May 2017 21:22:20
>> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>> Subject: Re: [Freeswitch-users] Changing audio DTLS state from HANDSHAKE to FAIL issue
>> 
>> can you try this using sip.js or something thats confirmed to work when using linux as a server please.  This may just be an issue with sipml5.
>>  
>>> On May 5, 2017, at 9:53 AM, Brian West <brian at freeswitch.org> wrote:
>>>  
>>> Clearly that isn't the case, I suspect a define is missing to actually enable ECDSA at this point.
>>>  
>>> On Fri, May 5, 2017 at 8:47 AM, Gregor Nanger <gregor at infomedia.si> wrote:
>>> Brian, isn't this solved in 1.6.16?
>>>  
>>> FS-10037 <http://t.sidekickopen68.com/e1t/c/5/f18dQhb0S7lC8dDMPbW2n0x6l2B9nMJN7t5XZsfmT10W5wLWvl4WYKY2W3MhGXn56dLgFf1Tdg-W02?t=https%3A%2F%2Ffreeswitch.org%2Fjira%2Fbrowse%2FFS-10037&si=5036397111279616&pi=a3abf16e-d54d-4338-ab43-13050b016003====> [core] Update OpenSSL to version 1.0.2k for Windows
>>>  
>>> 2017-05-05 15:26 GMT+02:00 Brian West <brian at freeswitch.org>:
>>> OpenSSL on the windows build needs to be updated.
>>>  
>>> https://freeswitch.org/jira/browse/FS-9510 <https://freeswitch.org/jira/browse/FS-9510====>
>>>  
>>> On Thu, May 4, 2017 at 2:31 PM, Gregor Nanger <gregor at infomedia.si> wrote:
>>> This error is familiar to me, I think so, if I remembered correctly. 
>>>  
>>> I am using windows. I had this error when client was Chrome. I just updated openssl dlls. I think that openssl library is updated on windows build now with version 1.6.17. And this version is also compiled on FS FTP to just download and install it.
>>>  
>>> 2017-05-04 19:15 GMT+02:00 Anthony Minessale <anthony.minessale at gmail.com>:
>>> FS 1.5 sounds like a bad plan.
>>> Try latest FS 1.6 or master.
>>>  
>>>  
>>> On Thu, May 4, 2017 at 12:04 PM, Ítalo Rossi <italo at freeswitch.org> wrote:
>>> Old openssl version maybe?
>>> Em qui, 4 de mai de 2017 às 11:19, sachin <snehsach at rediffmail.com> escreveu:
>>> Hello All,
>>> 
>>> I am trying to setup the Webrtc call between 2 SIPML5 client. I am using FreeSWITCH Version 1.6.14 on Windows. I am using internal LAN.
>>> 
>>> The clients are getting registered over wss. I have created self signed certificates.  In var.xml I have set the codecs setting as follows
>>> 
>>>   <X-PRE-PROCESS cmd="set" data="global_codec_prefs=PCMU,PCMA">
>>>   <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU,PCMA">
>>> 
>>> I am able to establish the call and there is 2 way voice when I call from Sipmpl5 to Linphone and it works. But when I call from Linphone to Firefox (Simpl5) then the call is not getting established and I am getting following error
>>> 
>>> 2017-05-04 11:29:04.543434 [ERR] switch_rtp.c:2907 audio Handshake failure 1
>>> 2017-05-04 11:29:04.543434 [INFO] switch_rtp.c:2908 Changing audio DTLS state from HANDSHAKE to FAIL
>>> 2017-05-04 11:29:04.543434 [NOTICE] switch_rtp.c:2889 Hangup sofia/internal/sip:1001 at df7jal <sip:1001 at df7jal====>23ls0d.invalid [CS_EXCHANGE_MEDIA] [DESTINATION_OUT_OF_ORDER]
>>> 
>>> My setup is as follows
>>> 
>>> SIPML5 (FireFox browser 53.0)<-----> FS (v1.5)(Windows) <----> Linphone
>>> 
>>> I am attaching the logs for the reference.
>>> fs-logleve9.txt : Debug trace with loglevel =9
>>> fs-sip-trace.txt : Sip trace
>>> Please let me know what could the issue and pointers to resolve the same.
>>> 
>>> Thanks and Regards
>>> SD
> 
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