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@jerris.com>
Sent: Fri, 05 May 2017 21:22:20
To: FreeSWITCH Users Help <freeswitch-users@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@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@infomedia.si> wrote:2017-05-05 15:26 GMT+02:00 Brian West <brian@freeswitch.org>:OpenSSL on the windows build needs to be updated.On Thu, May 4, 2017 at 2:31 PM, Gregor Nanger <gregor@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@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@freeswitch.org> wrote:Old openssl version maybe?Em qui, 4 de mai de 2017 às 11:19, sachin <snehsach@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@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