<div dir="ltr">It's already there, unless your version of OpenSSL doesn't have DTLS v1.2, Its wrapped in an ifdef HAVE_OPENSSL_DTLSv1_2_method<div><br></div><div>/b</div></div><br><div class="gmail_quote"><div dir="ltr" class="gmail_attr">On Tue, May 19, 2020 at 4:56 AM Valli A. Vallimamod <<a href="mailto:vma@vallimamod.org">vma@vallimamod.org</a>> wrote:<br></div><blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex">Hi,<br>
<br>
As you look familiar with the source code, you may add <br>
<br>
        SSL_CTX_set_min_proto_version(dtls->ssl_ctx, DTLS1_2_VERSION);<br>
<br>
as a quick hack in switch_rtp.c around where DTLS_server_method() / DTLS_client_method() are called.<br>
<br>
But it looks like a bug, you should create an issue on github.<br>
<br>
<br>
<br>
Best Regards,<br>
-- <br>
Valli A. Vallimamod<br>
SIP Solutions<br>
vma@sip.solutions<br>
<a href="http://linkedin.com/in/vallimamod" rel="noreferrer" target="_blank">linkedin.com/in/vallimamod</a><br>
.<br>
<br>
<br>
> On 12 May 2020, at 19:43, François-Xavier Geneste <<a href="mailto:fx.geneste@telemaque.fr" target="_blank">fx.geneste@telemaque.fr</a>> wrote:<br>
> <br>
> Hello guys,<br>
> <br>
>     I'm facing a big trouble for several hours ago and need help.... I'm using Freeswitch v1.10.2 with webRTC successfully installed and running. On the user/webphone side, I'm using Chrome 81.0.4044.138. Incoming and outgoing calls works fine with my webphone stack on my browsers (Firefox, Chrome). No warnings or errors at both sides.<br>
> <br>
>     But when I do the following scenario with a webphone that can manage several calls at the same time (multi-line feature), it does not work :<br>
> <br>
>       • make a first call routed to a webrtc extension, answer it and keep it connected<br>
>       • make a second call routed to the same extension, do not answer and keep the first call connected<br>
>       • make a third call routed to the same extension and hold the first line to accept this new call=> when I try to answer this 3rd call, the call is always dropped<br>
>     After digging into logs, and packets captured with wireshark, I found that when the freeswitch try to exchange with the browser to negociate SRTP flow for the 3rd call, it use DTLS v1.0 protocol (instead of v1.2) :<br>
> <br>
> <lnancehjiedpjici.png><br>
> <br>
>     Unfortunately, support for DTLS v1.0 seems to have been dropped on my webphone/browser side and the freeswitch fail on last DTLS exchange with this logs :<br>
> <br>
> [INFO] switch_rtp.c:3736 Activate RTP/RTCP audio DTLS client<br>
> [INFO] switch_rtp.c:3903 Changing audio DTLS state from OFF to HANDSHAKE<br>
> [...]<br>
> [ERR] switch_rtp.c:3266 audio Handshake failure 1. This may happen when you use legacy DTLS v1.0 (legacyDTLS channel var is set) but endpoint requires DTLS v1.2.<br>
> <br>
> <br>
>     On freeswitch side, I found only one option linked to the DTLS version (legacyDTLS, as written in logs) which I never set in my config. I checked my open ssl version on the freeswitch server (1.1.1d).<br>
> <br>
>     The thing that is disturbing to me is that if I hold the first call and answer the second call, it works well. The issue occurs only for the third call and after a missed/refused call while still connected with first call in parallel.<br>
> <br>
>     Digging into freeswitch source, I found that it seems to use version-flexible DTLS methods of openssl (DTLS_server_method() and DTLS_client_method()) and I cannot see how to quicly and simply always force DTLS v1.2 ?<br>
> <br>
>     Have any of you ever had this kind of problem or know how to solve it ?<br>
> <br>
> Regards,<br>
> <br>
> FX<br>
> <br>
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