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And are you forcing RTPengine to act as an ice light client? It
looks like you are gettin a single ICE candidate in the answer back
from freeswitch which would indicate that you are.<br>
<br>
I'd check your chrome webrtc statistics to see if tis failed to do
do ice/stun negotiation on the 183. In general the signalling looks
good.<br>
<br>
I think you may have an error on your Freeswitch side - some thing
that is trying to force it to use SRTP all the time, even though the
signalling has requested plain RTP (to freeswitch).<br>
<br>
I think you should ask in #freeswitch on freenode at this point.<br>
<br>
-Eric<br>
<br>
<div class="moz-cite-prefix">On 06/23/2016 01:42 PM, John Nash
wrote:<br>
</div>
<blockquote
cite="mid:CAA4rGPyFpKbhbLsRwKuP28HJD0rAFjZnHFOpAvtf09JP5QRHYg@mail.gmail.com"
type="cite">
<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 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"> 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
moz-do-not-send="true"
href="https://gist.github.com/johnnash13/0d2cb5238f3551cd3a8c6b4e638dd744"
target="_blank"><a class="moz-txt-link-freetext" href="https://gist.github.com/johnnash13/0d2cb5238f3551cd3a8c6b4e638dd744">https://gist.github.com/johnnash13/0d2cb5238f3551cd3a8c6b4e638dd744</a></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"><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"> 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
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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