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

[Anthony Minessale]

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 df7jal23ls0d.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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>
>
> _________________________________________________________________________
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> http://www.freeswitchsolutions.com
>
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>
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-- 
Anthony Minessale II       ♬ @anthmfs  ♬ @FreeSWITCH  ♬

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