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.
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.
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).
I think you should ask in #freeswitch on freenode at this point.
-Eric
On 06/23/2016 01:42 PM, John Nash wrote:
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 "SRTP output wanted, but no crypto suite was
negotiated"
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?
On Fri, Jun 24, 2016 at 1:05 AM, Eric Tamme <[email protected]
<mailto:[email protected]>> wrote:
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 <[email protected]
<mailto:[email protected]>> 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
<[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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