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?)
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.
-Eric
On 06/23/2016 01:24 PM, John Nash wrote:
OK here is the log
https://gist.github.com/johnnash13/0d2cb5238f3551cd3a8c6b4e638dd744
Sorry took me a while to convert wireshark trace to text file.
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.
On Fri, Jun 24, 2016 at 12:43 AM, Eric Tamme <e...@uphreak.com
<mailto:e...@uphreak.com>> wrote:
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 <e...@uphreak.com
<mailto:e...@uphreak.com>> 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
<e...@uphreak.com <mailto:e...@uphreak.com>> 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
<e...@uphreak.com <mailto:e...@uphreak.com>> 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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