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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.<br>
<br>
-Eric<br>
<br>
<div class="moz-cite-prefix">On 06/23/2016 01:24 PM, John Nash
wrote:<br>
</div>
<blockquote
cite="mid:CAA4rGPz3DAooeHVWH75k2R9TF=cSuZAZ3M-=VZbk1QL3_Xoe5g@mail.gmail.com"
type="cite">
<div dir="ltr">OK here is the log <a moz-do-not-send="true"
href="https://gist.github.com/johnnash13/0d2cb5238f3551cd3a8c6b4e638dd744">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 moz-do-not-send="true"
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 class="HOEnZb"><font color="#888888"><br>
<br>
-Eric</font></span>
<div>
<div class="h5"><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
moz-do-not-send="true"
href="mailto:eric@uphreak.com"
target="_blank"><a class="moz-txt-link-abbreviated" href="mailto:eric@uphreak.com">eric@uphreak.com</a></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 moz-do-not-send="true"
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
moz-do-not-send="true"
href="mailto:eric@uphreak.com"
target="_blank"><a class="moz-txt-link-abbreviated" href="mailto:eric@uphreak.com">eric@uphreak.com</a></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
moz-do-not-send="true"
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
moz-do-not-send="true" href="mailto:eric@uphreak.com" target="_blank"><a class="moz-txt-link-abbreviated" href="mailto:eric@uphreak.com">eric@uphreak.com</a></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
moz-do-not-send="true"
href="https://github.com/onsip/SIP.js" target="_blank"><a class="moz-txt-link-freetext" href="https://github.com/onsip/SIP.js">https://github.com/onsip/SIP.js</a></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
moz-do-not-send="true"
href="http://www.opensips.org/Documentation/Tutorials-WebSocket-2-2"
target="_blank"><a class="moz-txt-link-freetext" href="http://www.opensips.org/Documentation/Tutorials-WebSocket-2-2">http://www.opensips.org/Documentation/Tutorials-WebSocket-2-2</a></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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