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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