<div dir="ltr">Actually the issue is i hear no audio on either side and just after session progress (I guess when media starts coming from remote media server) i see error <span style="font-size:12.8px"> "SRTP output wanted, but no crypto suite was negotiated"</span><div><span style="font-size:12.8px"><br></span></div><div><span style="font-size:12.8px">I had also checked media logs i could see RTP packets being sent from freeswitch to RTPengine IP but there was no packet at all just after that. Ideally after RTP packet from freeswitch to rtpengine, Rtpengine should send that packet to browser using wss?</span></div></div><div class="gmail_extra"><br><div class="gmail_quote">On Fri, Jun 24, 2016 at 1:05 AM, Eric Tamme <span dir="ltr"><<a href="mailto:eric@uphreak.com" target="_blank">eric@uphreak.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div text="#000000" bgcolor="#FFFFFF">
So - i dont see a problem here - Chrome is getting UDP/TLS/RTP/SAVPF
and Freeswitch is getting RTP/AVP. Freeswitch responded to the
offer in the invite with an answer in the 183, and in the 200. What
is the failure you are seeing, and where is it happening (in
freeswitch? in the browser?)<br>
<br>
The only thing that looks bad is that you are retransmitting the ACK
which FS either ... doesnt like, or is never getting, because it
keeps retransmitting the 200, which is why you get a 481 when you
send BYE.<span class="HOEnZb"><font color="#888888"><br>
<br>
-Eric</font></span><div><div class="h5"><br>
<br>
<div>On 06/23/2016 01:24 PM, John Nash
wrote:<br>
</div>
<blockquote type="cite">
<div dir="ltr">OK here is the log <a href="https://gist.github.com/johnnash13/0d2cb5238f3551cd3a8c6b4e638dd744" target="_blank">https://gist.github.com/johnnash13/0d2cb5238f3551cd3a8c6b4e638dd744</a>
<div><br>
</div>
<div>Sorry took me a while to convert wireshark trace to text
file.</div>
<div><br>
</div>
<div>My freeswitch is running on private IP (127.0.0.1) and
opensips I run on both public and private so that for outside
world opensips is the only public IP they see. In proxy log I
pasted Opensips ===> Freeswitch logs and back.</div>
<div><br>
</div>
<div><br>
</div>
<div><br>
</div>
<div><br>
</div>
<div><br>
</div>
</div>
<div class="gmail_extra"><br>
<div class="gmail_quote">On Fri, Jun 24, 2016 at 12:43 AM, Eric
Tamme <span dir="ltr"><<a href="mailto:eric@uphreak.com" target="_blank">eric@uphreak.com</a>></span>
wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div text="#000000" bgcolor="#FFFFFF"> No - it's annoying to
look at a trace that's had information removed and try and
piece together whats happening. Your paranoid side is
wrong, sorry.<span><font color="#888888"><br>
<br>
-Eric</font></span>
<div>
<div><br>
<br>
<div>On 06/23/2016 01:06 PM, Patrick Wakano wrote:<br>
</div>
<blockquote type="cite">
<div dir="ltr">
<div>my paranoic side would recommend to
hide/change private informations, specially any
authentication line that might appear... this is
certainly a sort of social engineering threat we
should worry...<br>
</div>
<div>better be safe than sorry....<br>
</div>
<div><br>
</div>
</div>
<div class="gmail_extra"><br>
<div class="gmail_quote">On Thu, Jun 23, 2016 at
3:31 PM, Eric Tamme <span dir="ltr"><<a href="mailto:eric@uphreak.com" target="_blank"></a><a href="mailto:eric@uphreak.com" target="_blank">eric@uphreak.com</a>></span>
wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div text="#000000" bgcolor="#FFFFFF"> I mean
you can use a private gist, but you will be
publishing the link in a public email list.
In general I personally dont believe
revealing ip addresses etc. is any problem -
to put my money where my mouth is here is a
gist link to an unaltered SIP trace on my
server :)<br>
<br>
<a href="https://gist.github.com/etamme/b864010448a29007b7e0457682e81d52" target="_blank">https://gist.github.com/etamme/b864010448a29007b7e0457682e81d52</a><span><font color="#888888"><br>
<br>
-Eric</font></span>
<div>
<div><br>
<br>
<div>On 06/23/2016 12:23 PM, John Nash
wrote:<br>
</div>
<blockquote type="cite">
<div dir="ltr">Ok i am ready with
logs. About gist may I use private
option as traces have our IPs, user</div>
<div class="gmail_extra"><br>
<div class="gmail_quote">On Thu, Jun
23, 2016 at 10:32 PM, Eric Tamme <span dir="ltr"><<a href="mailto:eric@uphreak.com" target="_blank"></a><a href="mailto:eric@uphreak.com" target="_blank">eric@uphreak.com</a>></span>
wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div text="#000000" bgcolor="#FFFFFF"> Hey John,<br>
<br>
Please paste a full UNALTERED
sip trace into a gist (<a href="http://gist.github.com" target="_blank">gist.github.com</a>)
from the proxy servers
perspective and provide a link
so that we can see what comes
in, and what goes out from
both sides.<br>
<br>
EG: ngrep -qtd any -W byline
port 5060<br>
<br>
This will show us the traffic
that is leaving the proxy
destined for the Freeswitch
box, and what the freeswitch
box sends back.<br>
<br>
Also - you can look in your
browsers console log and
provide the SIP trace from
there in a seperate gist, so
that we can see what opensips
sends back up to your browser.<span><font color="#888888"><br>
<br>
-Eric</font></span>
<div>
<div><br>
<br>
<blockquote type="cite">
<div dir="ltr">
<div><span style="font-size:12.8px">Am
I using correct
sip.js example? I
copied it to my
server and
accessing it using
https: (used
letsencrypt)</span></div>
</div>
<div class="gmail_extra"><br>
<div class="gmail_quote">On
Thu, Jun 23, 2016 at
7:58 PM, Eric Tamme
<span dir="ltr"><<a href="mailto:eric@uphreak.com" target="_blank"></a><a href="mailto:eric@uphreak.com" target="_blank">eric@uphreak.com</a>></span>
wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div text="#000000" bgcolor="#FFFFFF">
1. I would
suggest using
SIP.js - <a href="https://github.com/onsip/SIP.js" target="_blank"></a><a href="https://github.com/onsip/SIP.js" target="_blank">https://github.com/onsip/SIP.js</a>
it is a much
more active
project that
sipml5.<br>
<br>
2. Im guessing
that you are not
properly passing
flags to
RTPEngine. If
you want to have
DTLS-SRTP
between the
browser, and
plain RTP/AVP
between
RTPEngine and
freeswitch, you
need to "offer"
rtp/avp to
freeswitch, and
"answer"
dtls-srtp back
up to the
browser.<br>
<br>
the offer to
freeswitch would
be: <br>
<pre> $var(rtpengine_flags) = "RTP/AVP replace-session-connection replace-origin ICE=remove";
</pre>
and the answer
back up to the
browswer would
be:<br>
<br>
<pre> $var(rtpengine_flags) = "UDP/TLS/RTP/SAVPF ICE=force";</pre>
<br>
-Eric
<div>
<div><br>
<br>
<br>
<div>On
06/23/2016
08:20 AM, John
Nash wrote:<br>
</div>
</div>
</div>
<blockquote type="cite">
<div>
<div>
<div dir="ltr">I
am following <a href="http://www.opensips.org/Documentation/Tutorials-WebSocket-2-2" target="_blank"></a><a href="http://www.opensips.org/Documentation/Tutorials-WebSocket-2-2" target="_blank">http://www.opensips.org/Documentation/Tutorials-WebSocket-2-2</a>
and trying to
test a call
<div><br>
</div>
<div>sipml5
----------->Opensips
+ rtpengine
-------->
SIP end point
(Freeswitch)<br>
<div><br>
</div>
<div>But I do
not have any
audio on both
sides. I see
this error at
rtpengine log
"SRTP output
wanted, but no
crypto suite
was
negotiated"<br>
</div>
</div>
<div><br>
</div>
<div>Anyone
tested this
scenario
positive?</div>
</div>
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