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:
Brian, isn't this solved in 1.6.16?

  • FS-10037 [core] Update OpenSSL to version 1.0.2k for Windows

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