[OpenSIPS-Users] Opensips + rtpengine + Sipml5 webrtc

[Eric Tamme]

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 <eric at uphreak.com 
> <mailto:eric at 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 <eric at uphreak.com
>>     <mailto:eric at 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
>>>         <eric at uphreak.com <mailto:eric at 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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>>>>             http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>
>>>
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>>>
>>>
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>>
>>
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>>
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>
>
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