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.
-Eric
On 06/23/2016 01:06 PM, Patrick Wakano wrote:
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...
better be safe than sorry....
On Thu, Jun 23, 2016 at 3:31 PM, Eric Tamme <[email protected]
<mailto:[email protected]>> wrote:
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 :)
https://gist.github.com/etamme/b864010448a29007b7e0457682e81d52
-Eric
On 06/23/2016 12:23 PM, John Nash wrote:
Ok i am ready with logs. About gist may I use private option as
traces have our IPs, user
On Thu, Jun 23, 2016 at 10:32 PM, Eric Tamme <[email protected]
<mailto:[email protected]>> wrote:
Hey John,
Please paste a full UNALTERED sip trace into a gist
(gist.github.com <http://gist.github.com>) from the proxy
servers perspective and provide a link so that we can see
what comes in, and what goes out from both sides.
EG: ngrep -qtd any -W byline port 5060
This will show us the traffic that is leaving the proxy
destined for the Freeswitch box, and what the freeswitch box
sends back.
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.
-Eric
Am I using correct sip.js example? I copied it to my server
and accessing it using https: (used letsencrypt)
On Thu, Jun 23, 2016 at 7:58 PM, Eric Tamme
<[email protected] <mailto:[email protected]>> wrote:
1. I would suggest using SIP.js -
https://github.com/onsip/SIP.js it is a much more active
project that sipml5.
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.
the offer to freeswitch would be:
$var(rtpengine_flags) = "RTP/AVP replace-session-connection
replace-origin ICE=remove";
and the answer back up to the browswer would be:
$var(rtpengine_flags) = "UDP/TLS/RTP/SAVPF ICE=force";
-Eric
On 06/23/2016 08:20 AM, John Nash wrote:
I am following
http://www.opensips.org/Documentation/Tutorials-WebSocket-2-2
and trying to test a call
sipml5 ----------->Opensips + rtpengine --------> SIP
end point (Freeswitch)
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"
Anyone tested this scenario positive?
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